Sonifex Catalogue 2016/17 (large file 17.5MB)

Sonifex Catalogue 2016/17 (large file 17.5MB)
2016-17
Catalogue
Welcome
Welcome To The New Sonifex Catalogue For 2016-17
Welcome to the new Sonifex catalogue
for 2016-17. We have a number of new
products, (see pages iv-v) which have been
added to our extensive brand lines.
Made in the UK by
Silver Member
We’ve got a heritage of building products
of the highest quality in the UK and we’re
proud to be a British manufacturer. Most
British radio & TV broadcast studios have
used Sonifex equipment and we currently
export on average 70% of our products,
supplying equipment to over 60 countries
world-wide. We’re recognised for the high
quality and reliability of our designs, in
recognition of both the excellent technical
audio/video quality and also the practical
design and pleasant aesthetic appearance of
our products.
2 Year Warranty
Key to Symbols:
When the symbol is not in use it is coloured
grey. When active, the symbol is gold.
By keeping as much of the design and
manufacturing processes in house as
possible, we can control the quality needed
to service the broadcast industry, where
24/7/365 operation is a requirement.
We have our own R&D team to design
hardware and software, two surface mount
PCB assembly lines with in-line product
testing and a full engineering workshop.
The production and stocking areas use a
product bar-coding system, so that we can
more tightly control and guarantee the high
quality of the products that you’re buying.
The constant need to innovate is now an
integral part of our culture, combining
healthy and outward-looking ideas with
sound and efficient design practices. We are
expanding our position in the broadcasting
industry by increasing our R&D efforts in
order to offer new designs of equipment
reflecting the quality and reliability that is
expected by you, our customers.
We hope that this catalogue contains the
products that you need, but if it doesn’t then
give us a call, we often manufacture bespoke
products for customers.
Thank you for using our products.
Marcus Brooke
Managing Director, Sonifex Ltd
You can go online to extend your standard 1 year warranty to 2 years at:
http://www.sonifex.co.uk/technical/register/index.asp
Radio Broadcasting
Studio/Engineering
Radio Broadcasting
Transmission
TV Broadcasting
Studio/Engineering
OB Vans/Trucks
Pro Audio
Post Production
Live Sound/PA
Commercial
Infrastructure
Security
Education
Controlled via RS232 or Ethernet using
the free of charge Sonifex Control
Interface (SCi) software.
Built-in web server for
control via a web browser.
i
ii
Contents
Contents
What’s New?
iv
Audio Monitors, Meters & Monitor Controllers
Reference Monitors
Confidence Monitors
Reference Monitor Meters
Reference Monitor Controllers
1
2
14
18
23
Redbox Audio & Video Interfaces
Redbox Introduction
Rackmount Kits
3G/HD/SD-SDI Video Distribution Amplifiers
3G/HD/SD-SDI Video Embedders & De-Embedders
3G/HD/SD-SDI Video De-Embedders & Re-Embedders
With Dolby Encode/Decode
Dolby Encoders & Decoders
Digital Audio Converters
Synchronisation Add-On Boards
Synchronisers, Delays & Silence Detectors
Matching Converters
Audio Distribution Amplifiers
Phantom Power Supply
Headphone Distribution Amplifiers
Mixers, Source Selectors & Switchers
Microphone Amplifiers & Limiters
Stereo to Mono Converters
Power Controllers
Talkback & Communications
Tone Generators
29
30
31
32
35
44
48
52
62
63
86
95
116
117
122
135
142
144
145
150
Portable Distribution Amps
153
Professional PCIe Sound Cards & Radio
Capture Cards
Professional PCIe Sound Cards
Radio Capture Cards
157
158
163
Audio Logging
Net-Log Network Audio Logger
Net-Log-Win Software
Mentor Time-Server
PC-FLS8 Flashlog 8 Audio & Radio Capture
Logging Software
167
168
170
173
Radio Broadcast Mixing Consoles
S0 Radio Broadcast On-Air Mixer
S1 Radio Broadcast On-Air Mixer
S2 Digital I/O Analogue Radio Broadcast Mixer
179
180
184
190
Radio Studio Packages
S2 Solutions Single and Split Mixer Furniture
Portable Radio Studio Package
Monitor Speaker Stands
215
218
220
221
Talkback & Commentary
Talkback Controller Units
Gooseneck Condenser Microphone
Commentator Units
Belt-Packs
Talkback Intercoms
223
224
229
230
238
240
Telephony & Communications
Line Powered Hybrids
Analogue Telephone Hybrids
245
248
250
Digital Telephone Hybrids
Digital GSM Hybrids
Digital HD Voice Hybrids
Pro Audio Streamers
252
258
260
265
Studio Illuminated Signs
SignalLED Range of LED Signs
Custom Signs
273
274
276
Index
279
174
iii
Whats New?
New Products
In this catalogue we’ve a host of new innovative products for
radio and TV broadcasters.
PC-AD2 2 Stereo Analogue I/O & 2 Stereo AES-3 I/O PCIe Half Height
Sound Card
The PC-AD2 is a dual stereo analogue input/
output and dual stereo AES-3 digital input/
output sound card in the PCIe half height
format. One of the analogue inputs can be
used as a mic input, there is a dedicated
PC-FLS8LT1/2 Flashlog 8 Single/Dual
Stereo Channel Logger
New licences of the Flashlog 8 software are
now available to allow logging of either one
or two stereo channels (plus IP streams).
Aimed at the community radio market,
these new licences offer full Flashlog
functionality and excellent value for money.
Page 176
iv
RB-FS42 Audio Failover Switcher, 4
Main I/O, 2 Standby I/O
The RB-FS42 4 + 2 audio failover switcher is
a smaller channel count version of the RBFS82, offering a lower number of inputs and
outputs. Page 82
Redbox RB-MSP6 6 Way Phantom
Power Supply
The RB-MSP6 is a 1U rack-mount phantom
power supply, designed to be used in
conjunction with up to six RB-MS4X3s,
providing mains input redundancy to ensure
the target microphones remain powered in
the event of a single mains failure. Page 116
RB-DDA6W-2P 6 Way Word Clock
Distribution Amplifier with Dual
Power Supplies
The RB-DDA6W-2P word clock distribution
amplifier distributes a word clock BNC
input signal to 6 word clock BNC outputs
re-conditioned. It is used to distribute
reference clocks for digital audio systems.
Page 106
RB-MA2G Dual Microphone Amplifier
Gain
The RB-MA2G consists of two independent
low-noise microphone pre-amplifiers with
front panel level controls & metering for
converting microphone level signals to
line level, or for driving long lines from
microphones to mixing equipment. Page 137
AES-11 sync input and also 2 GPIOs. It
is fully compatible with the Windows™
Wave, DirectSound, DirectShow, MCI and
Core Audio APIs, supporting audio up to
24 bit, 192kHz. Page 163
Sonifex Switchboard Software
The switchboard software is available free of
charge and can be used to remotely control
the DHY-04/G/HD range of telephone
hybrids over
an Ethernet
connection.
Page 263
What’s New?
New AoIP (Audio Over IP) Products
Please refer to our AoIP RAVENNA/AES67 catalogue for
more information on our new network based audio products.
S2-CDAS Dual Stereo
Line Channel &
S2-CDASE Dual Stereo
Line Channel with EQ
The S2-CDAS dual stereo line
channel is a new channel for
the S2 mixer and has one
24-bit 96kHz AES/EBU digital
input and one balanced
analogue stereo input.
Page 196
Sonifex joined the Ravenna group in
2012 and the AVN product range is the
result of our R&D in the area of audio
over IP. We have the following AoIP
products:
- The AVN-GMCS is a GPS receiver &
PTPv2 grandmaster clock, designed
to synchronise devices on an AoIP
network.
- The AVN-TB range is a new series of
talkback/intercom products which use
AoIP for communication.
AVN-GMCS IEEE1588 PTP
Grandmaster Clock with GPS
Receiver
The AVN-GMCS is a PTPv2 grandmaster clock
for use with AoIP applications. IEEE15882008 PTPv2 (precision time protocol) is
used to synchronise all the nodes within
a network. The AVN-GMCS supports the
Default (RAVENNA) & Media (AES67) Profiles
and in normal operation, the unit has PTPv2
time stamping resolution to 8nsec.
It can lock to GPS signals and sets the
onboard PTP & RTC clocks, providing
low jitter clock signals to be distributed
across the network. Options for precision
onboard crystals provides accurate
holdover sync if the GPS signal is lost. The
unit also has physical wordclock and PPS
outputs for studio use.
AVN-TB Range of AoIP RAVENNA/AES67 Talkback Intercoms
DHY-04HD Single Automatic HD Voice
& GSM Hybrid, AES/EBU & Analogue
I/O With Ethernet
The DHY-04HD HD Voice Hybrid is used on a
3G or GSM cellular (mobile) phone network
and can receive and make high quality
broadcast calls over the cellular network.
Page 260
AVN-TB10AR 10 Button Advanced
Talkback Intercom, AoIP Portal
The AVN-TB10AR is an advanced 10 button
intercom/talkback/paging intercom to
enable voice/audio communication between
different areas in a facility or building
complex. Comms can be made as a Talk
action, a Listen action or a duplex Talk/Listen
action to/from each station.
AVN-TB20AR 20 Button Advanced
Talkback Intercom, AoIP Portal
This is a 20 button talkback intercom in a 2U
rack chassis. Typical applications include OB
truck comms, theatre comms & inter-studio
comms in a TV or radio station.
AVN-TB20AD 20 Button Advanced
Talkback Intercom, AoIP Desktop
Portal
This is a desktop version providing access to
20+ other intercom stations on the network.
With a host of advanced features , 4-wire
analogue I/O and network audio, it’s the
perfect talkback system.
v
vi
Audio Monitors, Meters & Monitor Controllers
The Reference Monitor range offers a number of solutions for monitoring:
Reference Monitor Rackmount Audio Monitors
A range of 1U rack-mount monitors offering loudspeaker monitoring
& bar-graph LED metering of multiple audio inputs.
2
Confidence Monitors
Using the same outstanding speaker system as the Reference Monitor
range, the confidence monitors offer a superb sound with simple controls.
14
Reference Monitor Meters
Using the same accurate precision LED bar-graph meters, these products
are for visual metering of many analogue or digital audio inputs.
18
Reference Monitor Controllers
For use in controlling the level to a pair of speakers, or 5.1 surround sound
system, together with transmission light-sign control.
23
Image of Sonifex Reference Monitors in an HD-OB truck.
Photo supplied courtesy of Broadcast Solutions GmbH.
1
Audio Monitors, Meters & Monitor Controllers
Reference Monitor Rackmount Audio Monitors
The three monitors in the range are:
RM-2S4 Reference Monitor, 2 LED
meters, 4 stereo channel audio inputs.
RM-2S10 Reference Monitor, 2 LED
meters, 10 stereo channel audio inputs.
RM-4C8 Reference Monitor, 4 LED
meters, 8 channel inputs, dual selectors.
have been designed to ensure that their
audio performance is not compromised.
Each speaker uses a separate, highly
efficient class-D switching amplifier.
Even cable lengths to and from the speaker
enclosures have been kept short to reduce
any potential microphonic induction.
Uniquely an embedded 5 band parametric
EQ allows you to configure the monitor for
your environment or to suit your listening
tastes.
Anti-Vibration
A welded and sealed stainless-steel case
with milled aluminium fascia provides
exceptional rigidity and has been used to
ensure that there are no extraneous metallic
rattles. The lid is sealed with extensive thin
foam cut-outs to provide damping to the lid
and multi-point screw fixings are used to
ensure lid rigidity.
Detail In The Design
In the design of the product, every care has
been taken to ensure the best and most
accurate reproduction of the audio sources.
The XLR and USB port connectors on the
rear panel are sealed with foam, and silicon
sealant is used on components which could
move, or vibrate, under high SPL conditions.
In a 1U rack, the propogation of high power
sound waves in such a small enclosure could
have a tendency to produce rattles or move
components, but the Reference Monitors
Accurate Sound System
The speaker system comprises a three-way
arrangement with two mid/high frequency
speakers providing excellent stereo imaging
and a separately driven, forward facing, dual
magnet, mono bass driver.
The Reference Monitor Range is a series
of rack-mount audio monitors, combining
the latest DSP technology with outstanding
audio enclosure design to produce
monitors of the highest standards with
exceptional sound quality, a comprehensive
feature set and good looks in the rack.
Custom-moulded, profiled, HF enclosures
are used to minimise standing waves and
eliminate response peaks, and acoustic
damping in the HF enclosures is used to
reduce colouration, effectively creating a
separate, sealed, infinite-baffle enclosure
for each driver.
2
Audio Modifiers
Six illuminated soft-touch pushbuttons allow
front panel muting and dimming of the
loudspeakers, stereo-to-mono conversion,
phase inversion and Middle+Side encoding/
decoding with all front panel settings stored
in non-volatile memory which is recalled
at power-up. A universal power supply
ensures global voltage operation without
adjustment.
Six illuminated soft-touch pushbuttons.
Audio Monitors, Meters & Monitor Controllers
DSP Based Design
The use of a modern electronic architecture
allows a much better audio performance
to be realised. The DSP-based, 3rd-order
active crossover provides perfect separation
between mid-range and bass sounds.
A DSP-based electronic equalisation is
used to flatten the frequency response and
also enables the 5 band parametric EQ.
Additionally, the fast-attack DSP loudspeaker
limiter protects the drivers from overload
damage.
Optional HD Expansion Cards
A number of plug-in expansion cards are
available allowing multiple AES groups
embedded within an HD-SDI or SD-SDI
signal to be de-embedded and monitored,
or allowing Dolby® E or Dolby® Digital
encoded signals to be monitored directly:
RM-HD1 HD-SDI expansion card.
RM-HDE1 HD-SDI & Dolby® E expansion
card.
RM-E1X
Dolby® E expansion card, XLR.
RM-E1B
Dolby® E expansion card, BNC.
Inputs - RM-2S4 & RM-4C8
Analogue
Or Digital
Inputs x 8
A/D
Converters
A/D
Auto
Switch
Digital Signal Processor
Amp
AES/EBU
Receivers
x8
AES/EBU
Termination
A
Digital
Inputs
x 10
A/D
Converters
Power
Supply
D
E
F
D/A
Converters
G
Fixed/Variable
Line-Level
O/P
Phase
Metering
Level
Metering
Amp
Amp
D/A
Converters
AES/EBU
Transmitters
Headphones
Left
Speaker
Mid
Speaker
Right
Speaker
Line
Level
Outputs
A/D
Output
Alarms &
Logic I/O
Configuration
Switches
AES/EBU
Termination
Global
Mains
Power
I/O
Metering
(RM-2S4
Only)
A/D
Bank
Select
AES/EBU
Receivers
B
C
I/P Select &
Gain Modifiers
Inputs - RM-2S10
Ana
Inputs
x 10
Amp
Microcontroller
USB Remote
Control
Front Panel
+24V
+5V
-5V
Front
Panel
Indicators
RS232 Remote
Control
Display
Driver
Slave
Micro
Front Panel Switches
Front Panel Rotary Controls
A = 10 Way Selector
C = Fixed Speaker EQ
E = Volume & Balance Control
B = Signal Presence Detector
D = User Variable EQ
F = Speaker Protection Limiter
G = Crossover
Reference Monitor Block Diagram.
5 Band Parametric Equalisation
Each product in the Reference Monitor
range contains an embedded 5 band
parametric equaliser.
The Reference Monitor products are
often installed in OB trucks or engineering
rooms where the acoustics aren’t ideal.
The addition of a 5 band parametric EQ
allows you to alter the sound of your
Available Scales/Ballistics
For Reference Monitors
There are nine scales available with
accurately modelled ballistics. Each stereo
meter pair can have a different scale, set via
a DIPswitch on the underside.
A complete set of overlays are provided
per stereo meter so that you can define the
scale(s) that you need:
1.
2.
3.
4.
5.
6.
7.
8.
9.
Dual BBC PPM + standard VU
BBC PPM
IEC60268-10 11a
EBU PPM
IEC60268-10 11b
Nordic PPM
IEC60268-10 1
AES/EBU
IEC60268-18
digital PPM
DIN PPM
DIN45406
Standard VU
IEC60268-17
Extended VU
IEC60268-17
German PPM
monitor to account for the acoustic
conditions, or to suit your particular
listening tastes.
Using the free of charge SCi remote
control software, preset EQ settings can
be selected, or different EQ settings can
be created and stored.
3
Audio Monitors, Meters & Monitor Controllers
RM-2S4 Reference Monitor, 2 LED Meters & 4 Stereo Inputs
Category: Reference Monitors
Product Function: Audible and visual
metering of audio sources.
Typical Applications: Monitoring of
multiple audio sources in an OB truck or
engineering racks room.
Features: 3 speaker system, 5 band
parametric EQ, phase meter, audio
modifiers, multiple scales, analogue and
digital inputs, superb sound, option for
HD/SD-SDI de-embedding.
4
The RM-2S4 is a 1U rackmount unit offering
quality loudspeaker monitoring and
accurate, high-resolution metering of up to
four stereo audio sources and more with
the addition of optional expansion cards.
Sources may be in any mixture of analogue
and AES/EBU digital formats, with sample
rates up to 192kHz accepted.
Audio inputs can be analogue or digital
because they are autoswitching using the
left Neutrik™ XLR for AES/EBU, or both
Neutrik™ XLRs for analogue inputs and they
can be used in any combination.
Analogue inputs can be balanced or
unbalanced. The digital inputs have
switchable AES/EBU termination for (closerange) bridging operation and there is extra
global input gain available for both analogue
and digital low-level sources.
Sources (and additional banks of sources, if
fitted) are selected via a front panel rotary
encoder, with clear LED indication of the
current selection.
error conditions. On the rear panel, opencollector alarm outputs provide hardware
indication of sustained underlevel, overlevel,
phase errors and digital source lock.
A pair of line level audio outputs,
configurable as analogue or AES/EBU digital,
follow the selected source at either a fixed
level or one mirroring the loudspeaker
volume.
Six illuminated pushbuttons provide
access to a range of audio ‘modifiers’ –
instant dimming of the volume, individual
muting of each audio channel, stereo-tomono conversion, phase inversion and
Middle+Side transcoding. On the rear panel,
logic-level inputs allow direct remote access
to the DIM and MUTE functions.
The level of the chosen source is displayed
on a pair of bright, multicoloured
53-segment bargraph meters, with a number
of accurately modelled scales/responses
to suit different applications and local
preferences. Clear scale labels are provided
for you to chose the scale displayed and
the meter brightness can be adjusted from
the front panel. A separate phase meter
indicates channel correlation or phase
The three-way loudspeaker system is fed
via a DSP-based active crossover and a
trio of highly efficient Class-D amplifiers.
Careful attention to driver selection,
materials and case design, plus active DSP
equalisation, has ensured a flat response
and outstanding reproduction from such a
Audio Monitors, Meters & Monitor Controllers
The RM-2S4 and RM-2S10 are both monitor bridges which accept
stereo audio sources and have the same high specifications.
shallow unit. A protective limiter prevents
damage to the loudspeakers under overload
conditions and the front-panel headphone
socket automatically mutes the internal
loudspeakers when a plug is inserted.
A balance control allows you to alter
the stereo imaging of the left and right
channels.
A further five-band parametric equaliser
can be accessed for room-equalisation
purposes via Sonifex SCi Windows-based
remote control software. Source selection,
status monitoring and unit ID functions, plus
firmware updates to add extra functionality,
are all accessible remotely via both USB
and RS232 connections. The open control
protocol also allows operation with terminal
programs or customised applications.
An RM-HD1 HD/SD-SDI input expansion card
can be added, allowing multiple AES groups
embedded within an HD-SDI or SD-SDI signal
to be de-embedded and monitored.
The RM-2S4 operates from global mains
voltages (85-264V AC, 47-63Hz).
RM-2S10 Reference Monitor, 2 LED Meters,
10 Stereo Analogue & 10 Stereo Digital Inputs
The RM-2S10 has the same extended
feature set of the RM-2S4, monitoring
stereo channels, but has 10 dedicated
stereo analogue and 10 dedicated stereo
AES/EBU digital audio sources, with sample
rates up to 192kHz accepted.
The RM-2S10 uses 25 way D-type
connectors for the audio inputs on the rear
panel instead of Neutrik™ XLR connectors.
Sources are still selected via a front panel
rotary encoder, but on the RM-2S10 the
Source LEDs also act as signal present
indicators.
SCi Software Screen for RM-2S10.
In all other aspects the units have the same
feature set.
5
Audio Monitors, Meters & Monitor Controllers
Technical Specification For RM-2S4 & RM-2S10
Inputs
DIM:
Reduces speaker audio level by 10dB
Audio Inputs
(RM-2S4):
4 x stereo analogue or AES/
EBU digital (autoselecting)
CUT L & CUT R:
Mutes left/right speaker audio
MONO:
Combines left and right audio inputs
Audio Inputs
(RM-2S10):
10 x stereo analogue, plus
10 x stereo AES/EBU digital
PHASE INVERT:
Inverts phase of right audio input
M+S:
Converts stereo input to Middle (sum)
and Side (difference) signals
Max Level
+18dBu (analogue)/0dBFS (digital)
(0dB Input gain):
CMRR:
>60dB typical
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termination switchable)
AES/EBU
Sample Rate:
32 to 192kHz, converted internally
to 48kHz
Input Gain:
Selection:
Parametric
Bands:
Five
Centre
Frequency:
200Hz to 18kHz
Bandwidth:
0.25 to 2 octaves
0, +6, +12 or +18dB digital
gain (switchable)
Boost/Cut:
±12dB
Front panel rotary control with indicator
LEDs
Programming:
Via USB/serial control port
Amplifier/Loudspeakers
Audio Outputs:
1 x stereo analogue or AES/EBU
digital (switchable)
Gain re
Selected Input:
Unity or variable, following volume
control (switchable)
Maximum
Output Level:
+18dBu (analogue)/0dBFS (digital)
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AES/EBU
Sample Rate:
48kHz
Configuration:
Three-way with stereo mid/
high-frequency drivers & mono
low-frequency driver
Power Output:
2 x 5W (HF) + 20W (LF) with protective
limiter
Crossover:
500Hz (3rd order Butterworth)
Distortion
(HF Outputs):
< 0.05% (1kHz, 3W output)
Distortion
(LF Output):
< 0.05% (100Hz, 6W output)
Noise:
More than 80dB below full output
Mute to full volume via front
panel rotary control
Distortion:
<0.02% (1kHz, +8dBu output)
Volume:
Noise:
-84dB RMS, unity gain ref
+8dBu output
Balance Trim:
±6dB via front panel rotary control.
Frequency
Response:
20Hz-20kHz +0/-0.5dB
Peak Acoustic
Level:
102dB SPL @ 2ft
Crosstalk
1kHz input:
10kHz input:
Analogue I/O, ref 0dBu
<-90dB
<-85dB
Level Metering
Audio Modifiers
Modifier
Selection:
6
Selectable by switch from:
1. Dual BBC PPM + standard VU
2. BBC PPM
3. EBU PPM
4. Nordic PPM
5. AES/EBU digital PPM
6. DIN PPM
7. Standard VU
8. Extended VU
9. German PPM
Audio Inputs
(RM-2S10):
3 x D-type 25-pin female (balanced,
may be unbalanced)
Audio Outputs:
2 x XLR 3-pin male (balanced, may
be unbalanced)
Headphones:
1/4” (6.35mm) A-gauge 3-pole stereo
jack socket
Ballistics:
According to selected characteristic
Remote I/O:
D-sub 15-pin male
Line-Up Level:
According to selected characteristic
Mains Input:
Filtered 3-pin IEC male, continuously
rated 85 - 264VAC, 47 - 63Hz, fused,
60W peak, 30W average
5-segment, indication at 0, 45, 90, 135
and 180 degrees
Fuse Rating:
Anti-surge fuse 2A 20 x 5mm
User-Variable Equalisation
Type:
Line Level Outputs
Characteristics:
Number:
2 x 53-segment tri-colour
LED bargraphs
Type B socket
Serial:
D-sub 9-pin female
Phase Metering
Type:
Equipment Type
Remote Control
USB:
Slave device, 19200 baud
Serial:
RS232, 19200 baud, 3-wire connection
Alarm Outputs:
1. Audio underlevel/fail (latching)
2. Audio overlevel (latching)
3. Sustained phase error (latching)
4. AES/EBU input unlock (non-latching)
Open-collector outputs rated at 30V,
50mA maximum
Output low/conducting in normal
condition (no alarm)
Control Inputs:
1. Mute audio
2. Dim audio
3. Alarm reset
Pull-to-ground to activate inputs
Status Indicators
LIMIT:
Indicates loudspeaker protection
limiter is active.
CLIP:
Indicates internal digital clipping due
to overlevel.
LOCK:
Indicates lock achieved on selected
digital input(s).
OPT:
For future use.
Connectors
Illuminated front panel
pushbuttons
USB:
Audio Inputs
(RM-2S4):
8 x XLR 3-pin female (balanced, may
be unbalanced)
RM-2S4:
Reference Monitor, 2 LED meters,
4 stereo channel inputs
RM-2S10:
Reference Monitor, 2 LED meters,
10 stereo channel inputs
Physical Specification
Dimensions
(Raw):
48cm (W) x 30.5cm (D) x 4.4cm (H) (1U)
19” (W) x 12” (D) x 1.73” (H) (1U)
Dimensions
(Boxed):
58cm (W) x 52cm (D) x 14cm (H)
22.8” (W) x 20.5” (D) x 5.5” (H)
Weight:
Nett: 4.5kg Gross: 6kg
Nett: 10lb Gross: 13.2lb
Options
RM-HD1:
HD-SDI expansion card
Audio Monitors, Meters & Monitor Controllers
RM-4C8 Reference Monitor, 4 LED Meters, 8 Channel Inputs & Dual Source Selectors
Category: Reference Monitors.
Product Function: Audible and visual
metering of audio sources.
Typical Applications: Monitoring of
multiple audio sources in an OB truck or
engineering racks room.
Features: 3 speaker system, 5 band
parametric EQ, phase meter, audio
modifiers, multiple scales, analogue and
digital inputs, superb sound, options for
HD/SD-SDI de-embedding & Dolby® E
decoding.
With 4 x bright high-resolution 26-segment
meter displays and separate left and right
source selectors, the RM-4C8 is ideal
for monitoring audio channels in an SDI
group, or groups of de-embedded AES/EBU
channels.
The RM-4C8 offers the same functionality as
the RM-2S4 but with an additional source
selector so that any of the 4 channels in the
selected group, or bank, can be monitored
independently on left and right speakers.
Also, 4 meters are provided so that every
channel in the selected group, or bank, can
be visually monitored. All audio channels in
an HD-SDI signal can be monitored using the
optional RM-HD1 card and Dolby encoded
audio signals can be monitored using the
RM-HDE1, RM-E1B or RM-E1X expansion
cards.
input XLR, or both XLRs for analogue inputs.
The inputs can be used in any combination
of analogue or digital.
With full remote control via GPI, RS232 or
USB, a 5 band parametric equaliser, 6 front
panel modifier buttons and the ability to
take the optional HD-SDI expansion cards,
the RM-4C8 is a flexible and versatile
monitoring solution.
The same high level of care has been taken
in the design of the RM-4C8 as in the RM2S4 to ensure that it’s the best sounding 1U
rack-mount audio monitor that you’ll hear.
The audio inputs on the RM-4C8 are autosensing, for digital AES/EBU using the left
SCi Software Screen for RM-4C8.
7
Audio Monitors, Meters & Monitor Controllers
Technical Specification For RM-4C8
Audio Specification
Audio Modifiers
Inputs
Modifier
Selection:
Illuminated front panel
pushbuttons
Number:
4 x 26-segment tri-colour
LED bargraphs
Audio Inputs:
8 x XLR 3-pin female (balanced, may
be unbalanced)
Characteristics:
Selectable by switch from:
Audio Outputs:
2 x XLR 3-pin male (balanced, may
be unbalanced)
Headphones:
1/4” (6.35mm) A-gauge 3-pole stereo
jack socket
Audio Inputs
8 analogue or AES/ EBU digital
channels (autoselecting)
Level Metering
DIM:
Reduces speaker audio level by 10dB
Max Level
+18dBu (analogue)/0dBFS (digital)
(0dB Input Gain):
CUT L & CUT R:
Mutes left/right speaker audio
MONO:
Combines left and right audio inputs
CMRR:
>60dB typical
PHASE INVERT:
Inverts phase of right audio input
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switchable)
M+S:
Converts stereo input to Middle (sum)
and Side (difference) signals
AES/EBU
Sample Rate:
32 to 192kHz, converted internally
to 48kHz
Input Gain:
Selection:
Connectors
1. Dual BBC PPM + standard VU
2. BBC PPM
3. EBU PPM
4. Nordic PPM
5. AES/EBU digital PPM
6. DIN PPM
7. Standard VU
8. Extended VU
9. German PPM
User-Variable Equalisation
Type:
Parametric
0, +6, +12 or +18dB digital
gain (switchable)
Bands:
Five
Ballistics:
According to selected characteristic
Centre Frequency: 200Hz to 18kHz
Line-Up Level:
According to selected characteristic
2 x Front panel rotary controls with
indicator LEDs
Bandwidth:
0.25 to 2 octaves
Phase Metering
Boost/Cut:
±12dB
Type:
Programming:
Via USB/serial control port
Line Level Outputs
Audio Outputs:
1 x stereo analogue or AES/EBU
digital (switchable)
Gain re
Selected Input:
Unity or variable, following volume
control (switchable)
Amplifier/Loudspeakers
Configuration:
Three-way with stereo mid/
high-frequency drivers & mono
low-frequency driver
Remote I/O:
D-sub 15-pin male
Mains Input:
Filtered 3-pin IEC male, continuously
rated 85 - 264VAC, 47 - 63Hz, fused,
60W peak, 30W average
Fuse Rating:
Anti-surge fuse 2A 20 x 5mm
RM-4C8:
USB:
Slave device, 19200 baud
Physical Specification
Serial:
RS232, 19200 baud, 3-wire connection
Alarm Outputs:
1. Audio underlevel/fail (latching)
2. Audio overlevel (latching)
3. Sustained phase error (latching)
4. AES/EBU input unlock
(non-latching)
Open-collector outputs rated at 30V,
50mA maximum
Output low/conducting in normal
condition (no alarm)
Dimensions
(Raw):
48cm (W) x 30.5cm (D) x 4.4cm (H) (1U)
19” (W) x 12” (D) x 1.73” (H) (1U)
Dimensions
(Boxed):
58cm (W) x 52cm (D) x 14cm (H)
22.8” (W) x 20.5” (D) x 5.5” (H)
Weight:
Nett: 4.5kg Gross: 6kg
Nett: 10lb Gross: 13.2lb
Remote Control
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Impedance:
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(digital)
Crossover:
500Hz (3rd order Butterworth)
48kHz
Distortion
(HF Outputs):
< 0.05% (1kHz, 3W output)
Distortion:
<0.02% (1kHz, +8dBu output)
< 0.05% (100Hz, 6W output)
Noise:
-84dB RMS, unity gain ref
+8dBu output
Distortion
(LF Output):
Noise:
More than 80dB below full output
Frequency
Response:
20Hz-20kHz +0/-0.5dB
Volume:
Mute to full volume via front
panel rotary control
Crosstalk
1kHz input:
10kHz input:
Analogue I/O, ref 0dBu
<-90dB
<-85dB
Balance Trim:
±6dB via front panel rotary control
Status Indicators
Peak Acoustic
Level:
102dB SPL @ 2ft
LIMIT:
Indicates loudspeaker protection
limiter is active.
CLIP:
Indicates internal digital clipping due
to overlevel.
LOCK:
Indicates lock achieved on selected
digital input(s).
OPT:
For future use.
8
D-sub 9-pin female
5-segment, indication at 0, 45, 90, 135
and 180 degrees
+18dBu (analogue)/0dBFS (digital)
AES/EBU
Sample Rate:
Type B socket
Serial:
Equipment Type
Maximum
Output Level:
Power Output:
USB:
2 x 5W (HF) + 20W (LF) with
protective limiter
Control Inputs:
1. Mute audio
2. Dim audio
3. Alarm reset
Pull-to-ground to activate inputs
Reference Monitor, 4 LED meters,
8 channel inputs & dual source
selectors
Options
RM-HD1:
HD-SDI expansion card
RM-HDE1:
HD-SDI & Dolby® E expansion card
RM-E1X:
HD-SDI Dolby® E decoder XLR
expansion card
RM-E1B:
HD-SDI & Dolby® E decoder BNC
expansion card
Audio Monitors, Meters & Monitor Controllers
RM-4C8-HD1 Reference Monitor, 4 LED Meters, 8 Channel Inputs & Dual Source Selectors
With RM-HD1 3G/HD/SD-SDI De-Embedder Card
Category: Reference Monitors.
Product Function: Allows the deembedding of audio channels in a 3G/
HD/SD-SDI stream.
Typical Applications: OB truck
monitoring of incoming video feed
from a sports ground. Post Production
monitoring of HD-SD/SDI audio.
Features: Automatic 3G/HD/SD
input detection, visual monitoring of
all channels in a group, delay of an
embedded group with PC configuration.
This is a version of the RM-4C8 with
the RM-HD1 expansion card fitted. The
expansion card allows the monitoring of
embedded non-encoded linear PCM audio
channels within a single link 3G/HD/SD-SDI
video signal.
The RM-4C8 has individual channel selection
for each speaker allowing individual
channels in a 5.1 or multichannel mix to
be selected for monitoring, e.g. the Centre
or LFE channel can be isolated, or Left and
Right Rear surround channels.
The expansion board can extract any
selected AES/EBU audio group (4 channels)
from the video signal and pass them to the
main Reference Monitor unit for monitoring.
Any group can be selected and can also be
optionally delayed by up to ~300ms by using
the Sonifex SCi remote control software.
All channels in the selected group can be
monitored visually and 2 channels can be
selected to be monitored on the speakers.
The input is auto-sensing for either SD,
HD or single link 3G input formats and the
extraction of embedded audio complies with
SMPTE-272 (SD) and SMPTE-299 (HD/3G).
The SDI input is equalized, internally
re-clocked and re-transmitted to provide
a re-clocked output to pass to external
equipment such as another reference
monitor.
SCi Software Screen for RM-4C8-HD1.
For the technical specification please refer
to the RM-HD1 card itself.
9
Audio Monitors, Meters & Monitor Controllers
RM-4C8-HDE1 Reference Monitor, 4 LED Meters, 8 Channel Inputs & Dual Source Selectors
With De-Embedder & Dolby Decoder Card
Category: Reference Monitors.
Product Function: Allows the deembedding of audio channels in a 3G/HD/
SD-SDI stream and subsequent Dolby E
decoding for audible & visual monitoring.
Typical Applications: For taking a satellite
feed and monitoring HD-SDI and decoded
Dolby E sources.
Features: Dolby metadata display on
connected PC, 8 input metering on 4
meters, 3G/HD/SD-SDI de-embedding.
10
This version of the Reference Monitor is the
RM-4C8 with the RM-HDE1 expansion card
already fitted. So, it has all of the features
of the RM-4C8 as standard together with
the ability to de-embed audio from an HDSDI stream and then decode a Dolby digital
audio signal from that de-embedded audio.
The RM-HDE1 adds to the unit a 75ohm
HD-SDI input and output on BNC connectors
with the output acting as a loop-through of
the input.
The monitor can take an HD-SDI input and
de-embed audio from any selected audio
group, with left and right speakers being
able to monitor any 2 channels within that
group. The 4 channels in a group are shown
on the 4 bar-graph meters. If a Dolby E
or Dolby Digital input is detected on the
selected input, the Channels (CHANS) button
flashes to indicate that decoding is possible.
Pressing the CHANS button displays the
Dolby Digital 5.1 or Dolby E 8 channel signal,
using the 4 bar-graph meters.
Each meter is split into 2 sections allowing
you to view L, R, C, LFE, Ls, Rs for Dolby
Digital and an additional Lt, Rt for Dolby E.
The Programmes (PROGS) button allows
any Dolby programmes to be selected
and monitored.
Full Dolby Metadata is available using the
SCi remote control software. For a more
detailed description of Dolby Decoding
operation please refer to the Reference
Monitor handbook.
With full remote control via GPI, RS232 or
USB, a 5 band parametric equaliser, 6 front
panel modifier buttons and the ability to
take the optional HD-SDI expansion cards,
the RM-4C8 is a flexible and versatile
monitoring solution.
Audio Monitors, Meters & Monitor Controllers
RM-HD1 Reference Monitor 3G/HD/SD-SDI Expansion Card
Reference Monitor
RM-2S4 RM-2S10 RM-4C8
RM-HD1
RM-HDE1
RM-E1X
RM-E1B
This expansion board allows the monitoring
of embedded non-encoded linear PCM
audio channels within a single link 3G/HD/
SD-SDI video signal.
The RM-HD1 card can be used with the
RM-2S4, RM-2S10 and RM-4C8 monitors.
The expansion board can extract any
selected AES/EBU audio group (4 channels)
from the video signal and pass them to the
The SDI input is equalized, internally
re-clocked and re-transmitted to provide
a re-clocked output to pass to external
equipment such as another reference
monitor.
Technical Specification For RM-HD1
SDI Specification
SDI Input:
1 x BNC, 3G/HD/SD-SDI
SDI Outputs:
1 x BNC, 3G/HD/SD-SDI, Re-clocked
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Output Alignment
Jitter:
<0.2UI
Output Level:
800mV +/- 10%
Return Loss:
<15dB at 1500MHz
SDI Supported
Standards:
270Mbps, SMPTE-259M-C (SD)
1.485 or 1.4835Gbps, SMPTE-292M
(HD)
2.97 or 2.967Gbps, SMPTE-424M
(3G)
Supported Video
Formats:
525/59.94 (SMPTE-125M)
625/50 (ITU-R BT.656)
720p/23.98, 24, 25, 29.97, 30, 50,
59.94, 60 (SMPTE-296M)
1035i/59.94, 60 (SMPTE-260M)
1080i/50, 59.94, 60 (SMPTE-274M)
1080p/23.98, 24, 25, 50, 59.94, 60
(SMPTE-274M)
1080pSF/23.98, 24, 25, 29.97, 30
(RP-211)
1080i/50 (SMPTE-295M)
1080p/50 (SMPTE-295M)
Embedded Audio:
48kHz, synchronous (HD
asynchronous)
SMPTE-272M-ABC
SMPTE-299M
Equipment Type
SDI
nput
Reclocker
EQ
SDI
Receiver
Audio
PLL
SDI
utput
CD
Audio
De-embedder
1 Group
RM-HD1:
Audio
Delay
4 Channels
To Reference
Monitor
HVF
Expansion
Card
Category: Reference Monitor Expansion
Cards.
Product Function: Allows the deembedding of audio channels in a 3G/
HD/SD-SDI stream.
Typical Applications: OB truck
monitoring of incoming video feed
from a sports ground. Post Production
monitoring of HD-SD/SDI audio.
Features: Automatic 3G/HD/SD
input detection, visual monitoring of
all channels in a group, delay of an
embedded group with PC configuration.
main Reference Monitor unit for monitoring.
Any group can be selected and can also be
optionally delayed by up to ~300ms by using
the Sonifex SCi remote control software.
All channels in the selected group can be
monitored visually and 2 channels can be
selected to be monitored on the speakers
(for the RM-2S4 and RM-2S10, stereo
pairs are selected; the RM-4C8 allows any
2 channels to be monitored). The input is
auto-sensing for either SD, HD or single
link 3G input formats and the extraction of
embedded audio complies with SMPTE-272
(SD) and SMPTE-299 (HD/3G).
Bank/Source
Select
3G/HD/SD-SDI expansion card
Physical Specification
Dimensions (Raw):
15cm (W) x 11.5cm (D) x 3.1cm (H)
5.9” (W) x 4.5” (D*) x 1.2” (H) (1U)
Dimensions
(Boxed):
22.9cm (W) x 12.7cm (D) x 7.6cm (H)
9.0” (W) x 5.0” (D) x 3.0” (H)
Weight (RM-HD1):
Nett: 0.13kg Gross: 0.4kg
Nett: 0.3lb Gross: 0.9lb
11
Audio Monitors, Meters & Monitor Controllers
RM-HDE1 Reference Monitor HD-SDI & Dolby® E & Dolby® Digital Expansion Card
Category: Reference Monitor Expansion
Cards.
Product Function: Allows the deembedding of audio channels in a 3G/
HD/SD-SDI stream and subsequent Dolby
E decoding.
Typical Applications: For taking a
satellite feed and monitoring HD-SDI
and decoded Dolby E sources. Post
Production monitoring of HD-SD/SDI
audio.
Features: Dolby metadata display on
connected PC, 8 input metering on 4
meters, 3G/HD/SD-SDI de-embedding
and monitoring
This expansion board allows the monitoring
of embedded audio channels within a
single link 3G/HD/SD-SDI video signal.
The embedded audio can be either nonencoded linear PCM, Dolby® E or Dolby®
Digital. Please note that the RM-HDE1 card
is for use with the RM-4C8 unit only.
The expansion board can extract any
selected AES/EBU audio group (4 channels)
from the video signal and pass them to the
main Reference Monitor unit for monitoring.
Any group can be selected. All channels
in the selected group can be monitored
SDI
Input
Reclocker
Audio
De-embedder
Full Dolby Metadata is available using the
SCi remote control software. For a more
detailed description of Dolby Decoding
operation please refer to the handbook.
The input is auto-sensing for either SD,
HD or single link 3G input formats and the
extraction of embedded audio complies
with SMPTE-272 (SD) and SMPTE-299
(HD/3G). The SDI input is equalized,
1 Group
CD
SDI Specification
SDI Input:
1 x BNC, 3G/HD/SD-SDI
SDI Output:
1 x BNC, 3G/HD/SD-SDI, Re-clocked
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Output Alignment
Jitter:
<0.2UI
Output Level:
800mV +/- 10%
Return Loss:
<15dB at 1500MHz
SDI Supported
Standards:
270Mbps, SMPTE-259M-C (SD)
1.485 or 1.4835Gbps, SMPTE-292M
(HD)
2.97 or 2.967Gbps, SMPTE-424M
(3G)
Supported Video
Formats:
525/59.94 (SMPTE-125M)
625/50 (ITU-R BT.656)
720p/23.98, 24, 25, 29.97, 30, 50,
59.94, 60 (SMPTE-296M)
1035i/59.94, 60 (SMPTE-260M)
1080i/50, 59.94, 60 (SMPTE-274M)
1080p/23.98, 24, 25, 50, 59.94, 60
(SMPTE-274M)
1080pSF/23.98, 24, 25, 29.97, 30
(RP-211)
1080i/50 (SMPTE-295M)
1080p/50 (SMPTE-295M)
Embedded Audio:
48kHz, synchronous (HD
asynchronous)
SMPTE-272M-ABC & SMPTE-299M
RM-HDE1:
2
8
Dolby E &
Dolby Digital Downmix
Decoder
Metadata
12
Technical Specification For RM-HDE1
Equipment Type
Audio
PLL
SDI
Output
internally re-clocked and re-transmitted
to provide a re-clocked output to pass
to external equipment such as another
reference monitor.
SMPTE 337
Detector
HVF
EQ
SDI
Receiver
visually and any 2 channels can be selected
to be monitored on the speakers. If a Dolby
E or Dolby Digital input is detected on the
selected channel, the Channels (CHANS)
button flashes to indicate that decoding
is possible. Pressing the CHANS button
displays the Dolby Digital 5.1 or Dolby E
8 channel signal, using the 4 bar-graph
meters. Each meter is split into 2 sections,
top and bottom, allowing you to monitor all
8 channels coming from the Decoder.
8 Channels
To Reference
Monitor
Bank/Source
Select
RS232 Output
HD-SDI & Dolby® E expansion card
Physical Specification
Dimensions (Raw):
15cm (W) x 11.5cm (D) x 3.1cm (H)
5.9” (W) x 4.5” (D*) x 1.2” (H) (1U)
Dimensions
(Boxed):
22.9cm (W) x 12.7cm (D) x 7.6cm (H)
9.0” (W) x 5.0” (D) x 3.0” (H)
Weight (RM-HDE1):
Nett: 0.15kg Gross: 0.4kg
Nett: 0.3lb Gross: 0.9lb
Audio Monitors, Meters & Monitor Controllers
RM-E1X Reference Monitor Dolby E & Dolby® Digital Decoder XLR AES Expansion Card
RM-E1B Reference Monitor Dolby E & Dolby® Digital Decoder BNC Expansion Card
Category: Reference Monitor Expansion
Cards.
Product Function: Allow Dolby E
decoding from an AES/EBU audio input
Typical Applications: For monitoring a
Dolby E or D audio source.
Features: Dolby metadata display on
connected PC, 8 input metering on 4
meters, Dolby E decoding & monitoring.
RM-E1X Expansion Card.
These expansion boards for the Reference
Monitor RM-4C8 allow the monitoring of
a digital audio stream containing either
Linear PCM or Dolby* encoded audio.
The RM-E1X has 2 x XLR inputs to accept
AES/EBU level inputs and the RM-E1B has
2 x BNC connectors for S/PDIF level inputs.
RM-E1B Expansion Card.
PCM, then this audio data is simply passed
straight through.
Technical Specification For
RM-E1X and RM-E1B
Audio Specification
The input is re-transmitted with minimal
delay, to allow connection to other
equipment.
* Dolby and the double-D symbol are registered
trademarks of Dolby Laboratories.
Dolby data is decoded and passed to the
Reference Monitor as either a stereo down
mix or the full complement of individual
channels. If the signal is standard Linear
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Output:
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balanced I/O (RM-E1X)
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Output
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unbalanced I/O (RM-E1B)
Sample Rates:
32 - 48 kHz
Audio Formats:
Linear PCM
Dolby E (16, 20 and 24 bit)
Dolby Digital (16 and 32 bit)
Connectors
AES Input:
AES Output:
1 x XLR 3 pin socket (RM-E1X)
1 x XLR 3 pin plug (RM-E1X)
S/PDIF Input:
1 x BNC (RM-E1B)
S/PDIF Output:
1 x BNC (RM-E1B)
Equipment Type
AES
or S/PDIF
Input
AES
or S/PDIF
Output
AES / S/PDIF
Receiver
AES / S/PDIF
Transmitter
Dolby
Decoder
8 Channels To
Reference Monitor
RM-E1X:
RM-E1B:
Dolby® E Decoder XLR expansion card
Dolby® E Decoder BNC expansion card
Physical Specification
Dimensions
(Raw):
15cm (W) x 11.5cm (D) x 3.1cm (H)
5.9” (W) x 4.5” (D) x 1.2” (H) (1U)
Dimensions
(Boxed):
22.9cm (W) x 12.7cm (D) x 7.6cm (H)
9.0” (W) x 5.0” (D) x 3.0” (H)
Weight
(RM-E1X/B):
Nett: 0.13kg Gross: 0.4kg
Nett: 0.3lb Gross: 0.9lb
* Dolby and the double-D symbol are registered trademarks of Dolby Laboratories.
13
Audio Monitors, Meters & Monitor Controllers
RM-CA2 Confidence Monitor, 2 LED Meters & 2 Analogue Stereo Inputs
Category: Confidence Monitors.
Product Function: Audible & visual
monitoring of analogue audio inputs.
Typical Applications: Radio station
engineering racks-room for monitoring
news feeds.
Features:
• Anti-vibration steel case.
• Sealed lid with foam cut-outs to
dampen lid.
14
• Multi-point screw fixings ensure lid
rigidity.
• Rear connector ports sealed with foam.
• Glue used on components which could
move, or vibrate.
• Accurate 3-way speaker system.
• Two mid/high frequency speakers provide
excellent stereo imaging.
• Separately driven, forward facing, dual
magnet, mono bass driver.
• Custom-moulded, profiled, HF enclosures
minimise standing waves.
• Acoustic damping in the HF enclosures
reduces colouration.
• Separate, sealed, infinite-baffle
enclosure for each driver.
• Separate, highly efficient class-D
switching amplifier for each speaker.
• Short, even cable lengths to and from
the speaker enclosures to reduce any
potential microphonic induction.
• DSP based design allows better audio
performance to be realised.
• Active crossover provides perfect
separation between mid-range and bass
sounds.
• A universal power supply ensures global
voltage operation without adjustment.
• Optional DC power input.
Audio Monitors, Meters & Monitor Controllers
Using the same outstanding speaker
system as the Reference Monitor
range, the confidence monitors offer
a superb sound with simpler controls.
The confidence monitors have been
designed to give the best possible
performance at a reduced price.
The RM-CA2 is a 1U rack-mount unit
offering quality loudspeaker monitoring
and 2 channel metering of two stereo
analogue audio sources. Input 1 has stereo
balanced Neutrik™ XLRs and Input 2 has
both stereo balanced Neutrik™ XLRs and
stereo unbalanced RCA phono connectors.
The balanced analogue inputs can be wired
unbalanced if required.
Sources are selected via a front panel
push-button switch, with clear LED
indication of the current source.
A rear panel DIPswitch setting allows the
unit to monitor either:
• Stereo signals, with the two front panel
control knobs acting as stereo volume
and balance control, to alter the stereo
imaging of the left and right channels, or
• Dual mono signals, with the two front
panel control knobs acting as left and
right volume controls.
There is a front panel headphone socket
which responds to the volume controls
and the headphone socket automatically
mutes the internal loudspeakers when a
plug is inserted.
A pair of line-level analogue audio outputs
follow the selected source at the selected
level, or optionally at 10dB lower (if using
the unbalanced input), set via rear-panel
DIPswitch.
case design, plus active DSP equalisation,
has ensured a flat response and outstanding
reproduction from such a shallow unit. A
protective limiter prevents damage to the
loudspeakers under overload conditions
The level of the chosen source is shown on
an 8 segment LED bar-graph display with
PPM and VU scales indicated. The bar-graph
can optionally be lowered by 10dB (if using
the unbalanced input), set via rear-panel
DIPswitch.
The RM-CA2 operates from global mains
voltages (85-264V AC, 47-63Hz) without
adjustment and can optionally be ordered
with a DC 9V to 36V input instead of the
AC input.
A single phase meter LED indicates channel
correlation or phase error conditions.
The three-way loudspeaker system is fed via
a DSP-based active crossover and a trio of
highly efficient Class-D amplifiers. Careful
attention to driver selection, materials and
15
Audio Monitors, Meters & Monitor Controllers
RM-CAD8 Confidence Monitor, 2 LED Meters, 2 Analogue & 6 Digital Stereo Inputs
Category: Confidence Monitors.
Product Function: Audible & visual
monitoring of analogue audio inputs.
Typical Applications: Radio station
engineering racks-room for monitoring
news feeds.
Features: The RB-CAD8 has the same
feature set as the RB-CA2 with the
addition of AES/EBU, S/PDIF and TOSlink
digital audio inputs.
The RM-CAD8 has all the features of
the RM-CA2 together with the ability
to select from an additional 6 stereo
digital inputs.
16
As well as the 2 stereo analogue inputs
there are also:
• 4 x stereo AES/EBU balanced inputs on
XLR 3 pin female.
• 1 x stereo S/PDIF unbalanced input on
RCA phono female.
• 1 x stereo TOSlink unbalanced input on an
optical connector.
The source select button illuminates
to indicate synchronisation lock to the
incoming digital source.
Sample rate converters on the digital inputs
allow sources of different sample rates to be
connected and monitored, between 32kHz
and 96kHz.
All other features of the unit are identical to
the RM-CA2.
Audio Monitors, Meters & Monitor Controllers
Technical Specification For RM-CA2 & RM-CAD8
Audio Specification
Audio Outputs:
1 x stereo analogue
Inputs
Gain re
Selected Input:
Unity or -10dB (switchable)
Maximum
Output Level:
+18dB
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Impedance:
Audio Inputs
(RM-CA2):
2 x stereo analogue (1 x XLR balanced,
1 x XLR balanced or RCA phono unbal)
Audio Inputs
(RM-CAD8):
2 x stereo analogue (1 x XLR balanced,
1 x XLR balanced or RCA phono unbal)
4 x stereo XLR balanced AES/EBU digital
1 x stereo RCA phono S/PDIF digital
1 x stereo optical TOSLink digital
Max Level
+18dBu (analogue)/0dBFS (digital)
(0dB Input Gain):
CMRR:
>60dB typical
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/ŵƉĞĚĂŶĐĞƐ͗
Z͗хϭϬŬɏƵŶďĂůĂŶĐĞĚ
Digital Input
/ŵƉĞĚĂŶĐĞƐ͗
(RM-CAD8 only):
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AES/EBU
Sample Rate:
32 to 96kHz, converted internally
to 48kHz
(RM-CAD8 only):
Input Gain:
+10dB on unbalanced input
Selection:
Front panel push button with
indicator LEDs
Line Level Outputs
< 0.1% (100Hz, 6W output)
Headphones:
1/4” (6.35mm) A-gauge 3-pole stereo
jack socket
Noise:
More than 102dB below full output
Mains Input:
Volume:
Mute to full volume via front
panel rotary control
Filtered 3-pin IEC male, continuously
rated 85 - 264VAC, 47 - 63Hz, fused,
60W peak, 30W average
фϱϬɏ
Balance Trim:
±6dB via front panel rotary control.
Fuse Rating:
Anti-surge fuse 2A 20 x 5mm
98dB SPL @ 2ft
Equipment Type
Distortion:
<0.02% (1kHz, +8dBu output)
Peak Acoustic
Level:
Noise:
-95dB RMS, unity gain ref
+8dBu output
Level & Phase Metering
Number:
Frequency
Response:
20Hz-20kHz +0/-0.5dB
Crosstalk
1kHz input:
10kHz input:
Analogue I/O, ref 0dBu
<-90dB
<-85dB
Amplifier/Loudspeakers
Configuration:
Power Output:
Three-way with stereo mid/
high-frequency drivers & mono
low-frequency driver
Distortion
(LF Output):
RM-CA2:
Confidence Monitor, 2 LED meters,
2 analogue stereo inputs
2 x 8-segment LED bargraphs
RM-CA2-DC:
Line-Up Level:
0dB on scale can be set to 0dB or -10dB
via rear panel DIPswitch
Confidence Monitor, 2 LED meters,
2 analogue stereo inputs, DC supply
RM-CAD8:
Phase Meter:
Single LED indication showing average
Confidence Monitor, 2 LED meters,
2 analogue & 6 digital stereo inputs
RM-CAD8-DC:
Confidence Monitor, 2 LED meters,
2 analogue & 6 digital stereo inputs,
DC supply
Connectors
Audio Inputs
(RM-CA2):
4 x XLR 3-pin female balanced
2 x RCA female phono unbalanced
Audio Inputs
(RM-CAD8):
4 x XLR 3-pin female analogue
2 x RCA phono analogue
4 x XLR 3-pin female AES/EBU digital
1 x RCA phono female S/PDIF digital
1 x optical TOSLink digital
Dimensions
(Raw):
48cm (W) x 27cm (D) x 4.4cm (H) (1U)
19” (W) x 12” (D) x 1.73” (H) (1U)
Dimensions
(Boxed):
57cm (W) x 52cm (D) x15cm (H)
22.4” (W) x 20.5” (D) x 5.9” (H)
2 x XLR 3-pin male (balanced, may
be unbalanced)
Weight:
Nett: 4.5kg Gross: 6kg
Nett: 10lb Gross: 13.2lb
2 x 7W (HF) + 15W (LF) with
protective limiter
Crossover:
250Hz (24dB/octave, Linkwitz-Riley)
Distortion
(HF Outputs):
< 0.1% (1kHz, 3W output)
Audio Outputs:
Physical Specification
17
Audio Monitors, Meters & Monitor Controllers
Reference Monitor Meters
The Reference Monitor Meters are a range
of freestanding and 1U rack-mount precision
meters offering accurate, high resolution
metering of between 1 and 4 stereo audio
sources.
Each stereo source is auto-switching between
either analogue or digital AES/EBU format
with sample rates up to 192kHz accepted.
The level of each stereo source is displayed
on a pair of bright, multi-coloured bargraph
meters, with a large choice of accurately
modelled scales/responses to suit different
applications and local preferences. Separate
5 LED phase meters indicate channel
correlation or phase error conditions, and
additional LEDs show digital input lock and
audio level alarm status.
On the rear panel, open-collector alarm
outputs provide hardware indication of audio
under-level or silence, audio over-level,
sustained phase errors above 90 degrees and
digital source lock.
Status monitoring and unit ID functions, plus
firmware updates to add extra functionality,
are all accessible remotely via a RS232
connection in conjunction with Sonifex SCi
software.
18
All Reference Monitor Meters operate
from global mains voltages (85-264V AC,
47-63Hz) without adjustment.
Input 1
Analogue
or Digital
Input
There are eight Reference Monitor
Meters:
• RM-M1F53 1 Stereo 53-Segment
Meter, Free-Standing
• RM-M1R53 1 Stereo 53-Segment
Meter, Rack-Mount
• RM-M2F53 2 Stereo 53-Segment
Meters, Free-Standing
• RM-M2R53 2 Stereo 53-Segment
Meters, Rack-Mount
• RM-M4R53 4 Stereo 53-Segment
Meters, Rack-Mount
• RM-M1F106 1 Stereo 106-Segment
Meter, Free-Standing
• RM-M1R106 1 Stereo 106-Segment
Meter, Rack-Mount
• RM-M2R106 2 Stereo 106-Segment
Meters, Rack-Mount
Micro Controller
A/D
Converter
A/D Auto
Switch
AES/EBU
Receiver
I/P Gain
AES/EBUTermination
Input 2 (Input 2 not on RM-M1F53 or RM-M1R53)
Analogue
or Digital
Input
A/D
Converter
Level
Metering
A/D Auto
Switch
AES/EBU
Receiver
Configuration
Switches
Phase
Metering
AES/EBUTermination
RS232
Remote Control
Alarms &
Logic I/O
Front Panel
Global
Mains
Power
Global
Mains
Power
(Select Switch
on RM-M1F06,
RM-M1R106 &
RM-M2R106 Only)
Reference Monitor Meters Block Diagram.
Display
Driver
Indicators
Metering Display
Audio Monitors, Meters & Monitor Controllers
RM-M1F53 1 Stereo 53-Segment Meter, Free-Standing
Category: Reference Monitor Meters.
Product Function: Visual monitoring of
analogue and digital audio inputs.
Typical Applications: Radio station
engineering racks-room for monitoring
news feeds and audio sources.
Features:
• Bright, accurate precision LED meters.
• Auto-switching analogue or digital
audio inputs.
• 9 meter scales and ballistics types
• Phase meter.
• Alarm GPOs for under/over level.
• RS232 connection for PC updates and
control.
• Desktop or rack-mount formats.
The RM-M1F53 is a free-standing,
single stereo channel meter with
bright, multi-coloured
53-segment bargraph
meters and all the
features listed in the
summary.
Available Scales/Ballistics
For All Meters
There are nine scales available with
accurately modelled ballistics. Each stereo
meter pair can have a different scale, set via
a DIPswitch on the rear panel. A complete
set of overlays are provided per stereo
meter so that you can define the scale(s)
that you need:
1.
2.
3.
4.
5.
RM-M1R53 1 Stereo 53-Segment Meter,
Rack-Mount
6.
7.
8.
9.
Dual BBC PPM + standard VU
BBC PPM
IEC60268-10 11a
EBU PPM
IEC60268-10 11b
Nordic PPM
IEC60268-10 1
AES/EBU
IEC60268-18
digital PPM
DIN PPM
DIN45406
Standard VU
IEC60268-17
Extended VU
IEC60268-17
German PPM
The RM-M1R53 is a 1U rack-mount
version of the RM-M1F53.
SCi Software Screen for RM-M1F53 .
19
Audio Monitors, Meters & Monitor Controllers
RM-M2F53 2 Stereo 53-Segment Meters, Free-Standing
The RM-M2F53 is a free-standing, twin
stereo channel meter with bright, multicoloured 53-segment bargraph meters and
all the features listed in the summary.
RM-M2R53 2 Stereo 53-Segment
Meters, Rack-Mount
The RM-M2R53 is a 1U
rack-mount version of
the RM-M2F53.
RM-M4R53 4 Stereo 53-Segment Meters,
Rack-Mount
The RM-M4R53 is a 1U rack-mount which
combines 2 x RM-M2R53 units, providing
4 stereo channel monitoring.
20
Audio Monitors, Meters & Monitor Controllers
RM-M1F106 1 Stereo 106-Segment Meter, Free-Standing
The RM-M1F106 is a single stereo channel
free-standing meter but with 2 stereo inputs
and a front panel source selector.
Category: Reference Monitor Meters.
Product Function: Visual monitoring of
analogue and digital audio inputs.
Typical Applications: Radio station
engineering racks-room for monitoring
news feeds and audio sources.
Features: As per 53 segment meters
with the addition of a stereo source
select button.
This unit has a pair of multi-coloured
106-segment bargraph meters for greater
display accuracy. A front panel mounted
push-button selects the input source to
be monitored, from either of the 2 autoswitching analogue or digital AES/EBU
stereo inputs, or from a mono mix of each
channel, with channel 1 shown on the left
meter and channel 2 shown on the right
meter. All other functionality is listed in the
summary.
RM-M1R106 1 Stereo 106-Segment Meter, Rack-Mount
The RM-M1R106 is a 1U rack-mount version
of the RM-M1F106.
SCi Software Screen for RM-M1F106.
21
Audio Monitors, Meters & Monitor Controllers
RM-M2R106 2 Stereo 106-Segment Meters, Rack-Mount
A dual rack mount version, the RM-M2R106,
combines 2 x RM-M1R106 units, giving 4
stereo channel monitoring.
Connectors
Audio Inputs:
(All balanced, but may be wired unbal)
2 x XLR 3-pin female (RM-M1F53 &
RM-M1R53)
4 x XLR 3-pin female (RM-M2F53,
RM-M2R53, RM-M1F106 & RM-M1R106)
8 x XLR 3-pin female (RM-M4R53 &
RM-M2R106)
Serial:
D-sub 9-pin female
Remote I/O:
D-sub 15-pin female
Mains Input:
Filtered 3-pin IEC male, continuously
rated 85 – 264VAC, 47 – 63Hz, fused 1A,
30W peak, 15W average
Equipment Type
Technical Specification For Reference Monitor Meters
Audio Specification
Level Metering
Inputs
Number:
Audio Inputs:
1 x stereo analogue or AES/EBU digital
(auto-selecting) (RM-M1F53 & RM-M1R53)
2 x stereo analogue or AES/EBU digital
(auto-selecting) (RM-M2F53, RM-M2R53,
RM-M1F106 & RM-M1R106)
4 x stereo analogue or AES/EBU digital
(auto-selecting) (RM-M4R53 &
RM-M2R106)
Max Level
+18dBu (analogue)/0dBFS digital
(0dB Input Gain):
CMRR:
>60dB typical
/ŶƉƵƚ
/ŵƉĞĚĂŶĐĞ͗
ϮϬŬё;ĂŶĂůŽŐƵĞͿ
ϭϭϬё;ĚŝŐŝƚĂůǁŝƚŚƚĞƌŵŝŶĂƚŝŽŶƐǁŝƚĐŚĂďůĞ
via DIPswitch)
AES/EBU
Sample Rate:
32 to 192kHz, converted internally to
48kHz
Input Gain:
0, +6, +12 or +18dB digital gain
(switchable via DIPswitch)
Selection:
Front panel illuminated push switch
button (RM-M1F106, RM-M1R106 &
RM-M2R106)
22
Remote Control
1 x 53-segment tri-colour LED bargraphs
(RM-M1F53 & RM-M1R53)
2 x 53-segment tri-colour LED bargraphs
(RM-M2F53, RM-M2R53)
4 x 53-segment tri-colour LED bargraphs
(RM-M4R53)
1 x 106-segment tri-colour LED bargraphs
(RM-M1F106, RM-M1R106)
2 x 106-segment tri-colour LED bargraphs
(RM-M2R106)
Characteristics:
1.
2.
3.
4.
5.
6.
7.
8.
9.
Ballistics:
Line-Up Level:
Selectable by DIPswitch from:
Dual BBC PPM + standard VU
BBC PPM
IEC60268-10 11a
EBU PPM
IEC60268-10 11b
Nordic PPM
IEC60268-10 1
AES/EBU digital PPM IEC60268-18
DIN PPM
DIN45406
Standard VU
IEC60268-17
Extended VU
IEC60268-17
German
According to selected characteristic
According to selected characteristic
Phase Metering
Type:
5-segment, indication at 0, 45, 90,
135 and 180 degrees, 1 per stereo meter
RM-M1F53
1 stereo 53-segment meter, free-standing
RM-M1R53
1 stereo 53-segment meter, rack-mount
RM-M2F53
2 stereo 53-segment meters, free-standing
RM-M2R53
2 stereo 53-segment meters, rack-mount
RM-M4R53
4 stereo 53-segment meters, rack-mount
RM-M1F106
1 stereo 106-segment meter, free-standing
RM-M1R106
1 stereo 106-segment meter,rack-mount
RM-M2R106
2 stereo 106-segment meters, rack-mount
Serial:
RS232, 19200 baud, 3-wire connection
Physical Specification
Alarm Outputs:
1.
2.
3.
(1-4 single input units, 1-8 twin units)
Audio input 1 under-level/fail (latching)
Audio input 1 over-level (latching)
Audio input 1 sustained phase error
above 90 degrees (latching)
AES/EBU input 1 unlock (non-latching)
Audio input 2 under-level/fail (latching)
Audio input 2 over-level (latching)
Audio input 2 sustained phase error
above 90 degrees (latching)
AES/EBU input 2 unlock (non-latching)
Open-collector outputs rated at 30V,
50mA maximum
Output low/conducting in normal
condition (no alarm)
Dimensions
(Raw):
4.
5.
6.
7.
8.
Control Inputs:
1.
2.
(1 single input units, 2 twin units)
Input 1 Alarm reset
Input 2 Alarm reset
Pull-to-ground to activate inputs
Dimensions
(Boxed):
Dimensions
(Raw):
(Boxed):
Indicates lock achieved on digital input
Level Alarm:
Indicates audio over-level
RM-M1R53, RM-M2R53, RM-M4R53 &
RM-M1R106 & RM-M2R106
48cm (W) x 14.8cm (D) x 4.4cm (H)
19” (W) x 5.8” (D) x 1.73” (H)
59cm (W) x 27cm (D) x 7cm (H)
21” (W) x 10.6” (D) x 2.75” (H)
Weight:
RM-M1F53 & RM-M1R53
Nett: 1.3kg Gross: 2.0kg
Nett: 2.9lb Gross: 4.4lb
Weight:
RM-M2F53 & RM-M1F106
Nett: 1.4kg Gross: 2.1kg
Nett: 3.1lb Gross: 4.6lb
Weight:
RM-M2R53 & RM-M1R106
Nett: 1.45kg Gross: 2.15kg
Nett: 3.2lb Gross: 4.7lb
Weight:
RM-M4R53 & RM-M2R106
Nett: 2.8kg Gross: 3.5kg
Nett: 6.2lb Gross: 7.7lb
Status Indicators
AES Lock:
RM-M1F53, RM-M2F53 & RM-M1F106
21.8cm (W) x 14.8cm (D) x 4.4cm (H)
8.6” (W) x 5.8” (D) x 1.73” (H)
34.5cm (W) x 27cm (D) x 7cm (H)
13.6” (W) x 10.6” (D) x 2.75” (H)
Audio Monitors, Meters & Monitor Controllers
Reference Monitor Controllers
The Reference Monitor Controllers are a
range of 1U rack-mount audio production
control units, for providing source
selection, volume, DIM and CUT controls
for external analogue monitors, together
with light controls for the Sonifex SignalLED
range of studio signs, or similar.
Each of the products has a light control
section. This section has 3 buttons: Transmit
(TX), STANDBY and Rehearse (REH). The
TX and REH buttons control opto-isolated
outputs on the rear panel remote connector,
and these provide a direct control interface
to the Sonifex SignalLED illuminated studio
sign, allowing twin signs to be controlled
individually.
The Standby button initiates a 2 minute
countdown where the TX and REH buttons
and their associated remote outputs flash
alternately. After the 2 minutes has elapsed,
the TX output remains on.
The TX, STANDBY and REH buttons can
optionally be replaced if you want to use
different buttons for other applications.
There are four Monitor Controller products:
• RM-MC1L Monitor Controller, Single
Stereo Input & Output With Light Control
• RM-MC4L Monitor Controller, 4 Stereo
Inputs, 1 Output With Light Control
• RM-MC51L Monitor Controller,
5.1 Stereo Inputs & Outputs With Light
Control
• RM-MCL Monitor Light Controller, 3
Button, TX-STANDBY-REH
The remote port also has open-collector
tally outputs for Transmit (TX), Rehearse
(REH), as well as remote input control for
the audio controls DIM and CUT, and the
light controls TX, Standby and REH. There is
also a +12 Volt source at 200mA maximum.
23
Audio Monitors, Meters & Monitor Controllers
RM-MC1L Monitor Controller, Single Stereo Input & Output With Light Control
Category: Reference Monitor
Controllers.
Product Function: Volume control, dim
and mute control of powered speakers
with Rehearse/Transmit light control.
Typical Applications: Production control
room, live studio broadcasts, recording
studios.
Features: Separate remoteable DIM &
CUT buttons, tally outputs for REH and
TX, DIM level control.
24
The RM-MC1L is a stereo unit with a
single stereo balanced analogue input and
output on XLRs. Useful in a production
environment, the unit allows you to control
the audio level to external monitors and
to DIM the audio, or CUT it completely
with simple front panel controls, or remote
external inputs.
It has a master volume control that adjusts
the attenuation on the output channels
from 0dB down to -86dB. A CUT control
mutes the audio outputs when enabled,
whilst a rear panel DIM control attenuates
the audio output channels by a pre-set
amount from -2dB to -23dB.
All of the buttons on the front panel are
illuminated by LEDs and the brightness can
be changed to better suit the environment
in which the unit is used.
The remote port allows control of the DIM
and CUT buttons and has tally outputs for TX
(transmit) and REH (rehearse).
Audio Monitors, Meters & Monitor Controllers
RM-MC4L Monitor Controller, 4 Stereo Inputs, 1 Stereo Output With Light Control
Category: Reference Monitor
Controllers.
Product Function: Volume control, dim
and mute control of powered speakers
with a 4 source selector and Rehearse/
Transmit light control.
Typical Applications: Production control
room, live studio broadcasts, recording
studios.
Features: 4 input source selector,
separate remoteable DIM & CUT
buttons, tally outputs for REH and TX,
DIM level control.
The RM-MC4L has four stereo balanced
analogue inputs with a single stereo
balanced output, all using XLR connectors.
The RM-MC4L has the same master volume
control, CUT, DIM and light control functions
and adjustments as the RM-MC1L unit.
Four illuminated front panel buttons allow
you to simply select the source required at
the output.
25
Audio Monitors, Meters & Monitor Controllers
RM-MC51L Monitor Controller, 5.1 Stereo Inputs & Outputs With Light Control
Category: Reference Monitor
Controllers.
Product Function: Volume control, dim
and mute control of surround sound
powered speakers with Rehearse/
Transmit light control.
Typical Applications: 5.1 speaker system
level control, production control rooms,
live studio broadcasts, recording studios.
Features: Separate remoteable DIM &
CUT buttons, tally outputs for REH and
TX, DIM level control.
26
The RM-MC51L is a controller for surround
sound applications with 6 balanced
analogue inputs and 6 balanced analogue
outputs all on XLR connectors.
It is used in situations where you need
to adjust the volume, or CUT, or DIM all
of the speakers in a 5.1 surround system.
Alternatively, the unit can be used as
a global volume control for 6 analogue
outputs.
The RM-MC51L has the same master volume
control, CUT, DIM and light control functions
and adjustments as the RM-MC1L unit.
Audio Monitors, Meters & Monitor Controllers
RM-MCL Monitor Light Controller, 3 Button, TX-STANDBY-REH
The RM-MCL is a simple light controller
for controlling a TX - STANDBY - REH light
system in a production studio, as detailed in
the introduction. It has no audio inputs or
outputs.
Specification For RM-MC1L, RM-MC4L, RM-MC51L & RM-MCL Reference Monitor Controllers
Audio Specification
Front Panel Operation Controls
/ŶƉƵƚ
Impedance:
хϮϬŬё
Light Control
(All units):
Transmit (TX), Standby & Rehearse
(REH) illuminated buttons
Maximum
Input Level:
+25dBu
Channel Select
(RM-MC4L):
1, 2, 3 & 4 illuminated buttons
KƵƚƉƵƚ
Impedance:
фϱϬё
Audio Controls:
DIM & CUT illuminated buttons
(not on RM-MCL)
Maximum
Output Level:
+25dBu
Attenuation
Range:
>86dB
DIM
Attenuation
Range:
Adjustable -3dB to -23dB via rear panel
potentiometer
Volume Control: Rotary potentiometer for output
attenuation on front panel
(not on RM-MCL)
Rear Panel Connections
Common
>66dB typical
Mode Rejection:
Distortion:
<0.04% ref +8dBu
Crosstalk:
<-80dB @ 1kHz input, ref 0dBu
Audio Inputs
(RM-MC1L):
2 x XLR 3-pin female (balanced)
Audio Inputs
(RM-MC4L):
8 x XLR 3-pin female (balanced)
Audio Inputs
(RM-MC51L):
6 x XLR 3-pin female (balanced)
Audio Outputs
2 x XLR 3-pin male (balanced)
(RM-MC1L & RM-MC4L):
Audio Outputs
(RM-MC51L):
6 x XLR 3-pin male (balanced)
Remote
I/O Port:
15 way D-type socket
Mains Input:
Filtered 3-pin IEC male, continuously
rated 85-264VAC, 47-63Hz, fused
Power Consumption:
RM-MC1L
Peak 10W, average 4W
RM-MC4L
Peak 10W, average 6W
RM-MC51L
Peak 10W, average 7W
RM-MCL
Peak 10W, average 2W
Fuse Rating:
Anti-surge fuse 1A, 20 x 5mm
Equipment Type
RM-MC1L:
Monitor controller, single stereo input
& output with light control
RM-MC4L:
Monitor controller, 4 stereo inputs, 1
stereo output with light control
RM-MC51L:
Monitor controller, 5.1 stereo inputs &
outputs with light control
RM-MCL:
Monitor light controller, 3 button,
TX-STANDBY-REH
Physical Specification
Dimensions
(Raw):
48cm (W) x 10.8cm (D) x 4.3cm (H)
19” (W) x 4.3” (D) x 1.7” (H) (1U)
Dimensions
(Boxed):
58.5cm (W) x 22.5cm (D) x 7cm (H)
23” (W) x 8.9” (D) x 2.8” (H)
Weight :
Nett: 1.3kg Gross: 1.9kg
Nett: 2.9lbs Gross: 4.2lbs
27
28
Redbox Audio & Video Interfaces
The Redbox audio & video interfaces are problem solvers for use in radio, TV & recording
studios and video & post production suites. For broadcast applications and where products
are being used 24/7, Redbox’s outstanding product quality & reliability give you all the
assurance you need.
Redbox Introduction
30
Rackmount Kits
31
3G/HD/SD-SDI Video Distribution Amplifiers
32
3G/HD/SD-SDI Video Embedders & De-Embedders
35
3G/HD/SD-SDI Video De-Embedders & Re-Embedders With Dolby Encode/Decode
44
Dolby Encoders & Decoders
48
Digital Audio Converters
52
Synchronisation Add-On Boards
62
Synchronisers, Delays & Silence Detectors
63
Matching Converters
86
Audio Distribution Amplifiers
95
Phantom Power Supply
116
Headphone Distribution Amplifiers
117
Mixers, Source Selectors & Switchers
122
Microphone Amplifiers & Limiters
135
Stereo to Mono Converters
142
Power Controllers
144
Talkback & Communications
145
Tone Generators
150
Image of Sonifex Redboxes and Telephone Hybrids in a bus.
Photo supplied by BTS 98FM, Dublin.
29
Audio & Video Interfaces
Redbox Audio & Video Interfaces
Designed and manufactured to the
highest specifications, Redbox is a
range of analogue and digital audio &
video interfaces for use in a multitude
of professional audio applications,
including installations at radio
stations, TV studios, home studios,
video suites, production & postproduction houses and recording
studios.
Started in 1999, the Redbox range has
expanded to include over 75 high quality,
versatile and reliable units.
All of the ideas for new products have
come from you, the customers, so if you
have a requirement for a new interface, or
modifications to an existing one, then let us
know by sending an email to [email protected]
co.uk.
Manufactured to the highest standards
in our UK offices, utilisation of the finest
components and critical quality control
techniques ensure that your Redboxes will
work every time for years to come. Designed
for 24/7/365 operation for broadcast
applications, each Redbox is tested twice
by skilled audio engineers, before being
carefully assembled and packed.
The Redbox range of products are “fit and
forget” because you can set them up, fit
them in your installation and then forget
about them - they won’t trouble you.
Features
• All are equipped with IEC mains lead and
instruction manual.
• In-house design and manufacture
ensures high quality control standards.
• All units are 115V 60Hz or 230V 50Hz
switchable and all have a front panel LED
power indicator.
• Manufactured within ISO9000 standards
and guaranteed CE compliant.
• Housed in eye catching red anodised
aluminium cases.
All the Redboxes are screw mountable as
standard and are either rack-mounted or
have the option to be rack-mounted. The
RB-RK rack mount kits can be attached to
the front or the rear of the Redbox products
so that they can be rack mounted into a
standard 19 inch rack frame in 1U of space.
Rack Mounting
For rack mounting smaller (28cm) (11”)
units the optional RB-RK1 (Red) or RB-RK1B
(Black) kit can be used (which include 4 off
M6 panel fixing screws).
Rear Panel Mounting
For rear panel mounting you can use either
the RB-RK2, or RB-RK3, depending on the
size of your Redbox.
30
RB-RK2 1U rear panel rack kit for small
Redbox range, e.g., RB-BL2
RB-RK3 1U rear panel rack kit for large
Redbox range, e.g., RB-DA6.
The RB-RK1 can be used with the following
products:RB-SC1
RB-UL1
Sample rate converter
Unbalanced to balanced
single converter
RB-UL2
Dual unbalanced to
balanced converter
RB-BL2
Unbalanced to balanced
bi-directional converter
RB-PA2
Dual stereo RIAA phono
amplifier
RB-LI2
Stereo line isolation unit
RB-DDA6A 6 way AES/EBU digital
distribution amplifier
RB-DDA6A3 6 way stereo AES3ID digital
distribution amplifier
RB-DDA6S 6 way S/PDIF digital
distribution amplifier
RB-DDA6W 6 way word clock
distribution amplifier
RB-MA1
Single microphone amplifier
RB-MA2
Dual microphone amplifier
RB-ML2
Stereo microphone & line
level limiter
RB-SL2
Twin mono, or stereo, limiter
Audio & Video Interfaces - Rackmount Kits
Wherever you see this symbol an RB-RK1 rackmount kit can be used.
RB-SM1
RB-SM2
RB-LC3
RB-MM1
Single stereo to mono
converter
Dual stereo to mono converter
Light/Power controller
Mix-minus generator
Front Panel Kits for Redboxes
A number of customers have asked
for non-rackable fronts for the RB-HD1,
RB-HD2, RB-ADDA, RB-UL4, RB-BL4
and RB-LU4.
Wherever you see this symbol an RB-RK1B rackmount kit can be used.
Wherever you see this symbol an RB-RK2 small Redbox rear panel rack kit ears can be used.
These are available without rack-ears as:
RB-FR1, RB-FR2 and RB-FR3.
Wherever you see this symbol an RB-RK3 large Redbox rear panel rack kit ears can be used.
When Redboxes are shipped to different
countries, the correct IEC mains lead is
chosen at time of despatch:
RB-FR1 - Front panel, no 19” mount, for RB-ADDA, RB-UL4, RB-BL4 & RB-LU4.
UK - 230V, UK 3 pin to IEC
lead
EC - 230V, EEC Schuko 2 pin
to IEC lead
RB-FR2 - RB-HD1 or RB-DAC1 front panel, no 19” rack mount.
US - 115V, US 3 pin to IEC
lead
AU - 230V, Australasian 3 pin
to IEC lead
RB-FR3 - RB-HD2 front panel, no 19” rack mount.
31
Audio & Video Interfaces - Video Distribution Amplifiers
RB-VHDA2x4 3G/HD/SD-SDI 2 Input, 8 Output Video Distribution Amplifier
Category: 3G/HD/SD-SDI Video
Distribution Amplifiers.
Product Function: Distributes 2 x SDI
32
inputs to 4 outputs, or 1 SDI input to 8
outputs.
Typical Applications: Distribution of football
HD live feed in OB truck, distribution of video
from HD video player.
Features: 2 x SDI inputs, 8 outputs, fully
reclocked outputs, AC and DC power supplies
provide redundancy, web server for setup.
The RB-VHDA2x4 is a high performance,
reliable dual input and eight output digital
video distribution amplifier for re-clocking
and distributing up to two 3G, HD (high
definition) or SD (standard definition) SDI
sources.
By default, the RB-VHDA2x4 acts as two
independent video distribution amplifiers,
with input 1 routed to outputs 1-4 and
input 2 routed to outputs 5-8. Either input
can also be routed to all outputs if desired
which can be selected by the Input Select
switch on the front panel or via the built-in
webserver. The webserver also provides
status information for the unit.
The unit provides automatic input
detection, re-clocking and cable equalisation
of the input signals to 100/350 meters (HD/
SD) of coax cable.
Audio & Video Interfaces - Video Distribution Amplifiers
The full range of 3G, SD and HD standards
are supported by the unit allowing PAL
and NTSC signals up to 1080p 60Hz to be
distributed.
Each output is individually buffered
providing separately driven outputs for
greater reliability.
As well as an AC power input, the unit also
has a 12V DC input for use, for example,
in outside broadcast vehicles. Both power
inputs can be used to give a dual redundant
power supply; whichever voltage is higher
will be used.
RB-VHDA2x4 Home Page.
SDI Output 1
SDI Output 2
SDI Output 3
Reclock
RB-VHDA2x4 Network Page.
RB-VHDA2x4 Configuration Page.
Specification For RB-VHDA2X4
Rear Panel Connections
Audio Specification
3G/HD/SD-SDI
Video Input:
2 x BNC (Unbalanced)
3G/HD/SD-SDI
Video Outputs:
8 x BNC (Unbalanced)
Ethernet Port:
RJ45 with status LEDs
Compatible
Standards:
SMPTE-424M (3G)
SMPTE-292M (HD-SDI)
SMPTE-259M-AC1 (SD-SDI)
SMPTE 344M
DVB-ASI at 270Mb/s
1.
Other rates will be passed without reclocking.
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Equalisation:
Up to 100m of Belden 1694A
at 1.485Gbps
Up to 350m of Belden 1694A at
270 Mbps
Fuse Rating:
DC:
Data Rates:
143, 270, 1483.5,
Equipment Type
Fully supported
Passed but not
reclocked:
1485, 2967 & 2970Mbps
177, 360 & 540Mbps
RB-VHDA2x4:
Reclocking &
Equalisation:
Automatic
Resolution:
Automatic
SDI Output 4
SDI Input 1
Reclock
SDI Output 5
SDI Input 2
Crosspoint
SDI Output 6
SDI Output 7
SDI Output 8
RB-VHDA2x4 Block Diagram.
Power Supply:
AC:
Filtered IEC, continuously rated
85-264VAC, 47-63Hz, fused, max
10W
Anti-surge fuse 1A 20 x 5mm
12V 300mA max, DC supply,
2.5mm socket fused
3G/HD/SD-SDI 2 inputs, 8 output
video distribution amplifier
Physical Specification
Dimensions (Raw):
48cm (W) x 10.8cm (D) x 4.3cm (H)
19” (W) x 4.3” (D) x 1.7” (H) (1U)
Return Loss:
>15dB at 1500MHz
Dimensions
(Boxed):
58.5cm (W) x 22.5cm (D) x 7cm (H)
23” (W) x 8.9” (D) x 2.8” (H)
Output Jitter:
<0.2UI
Weight:
Signal Level:
800mV ±10%
Nett: 1.3kg Gross: 1.9kg
Nett: 2.9lbs Gross: 4.2lbs
Accessories
Front Panel Indicators
Power LED:
Red indicates power is present
Input Status
1 & 2 LEDs:
Green indicates signal is present,
flashing red indicates signal is
missing.
RB-RK3:
1U Rear panel rack kit for large
Redboxes
33
Audio & Video Interfaces - Video Distribution Amplifiers
RB-VHDA8 3G/HD/SD-SDI 1 Input, 8 Output Video Distribution Amplifier
The RB-VHDA8 is a high performance,
reliable eight way digital video distribution
amplifier for re-clocking and distributing
a 3G, HD (high definition) or SD (standard
definition) SDI source to eight outputs. The
unit provides automatic input detection,
re-clocking and cable equalisation of the
input signal to 100/350 meters (HD/SD) of
coax cable.
The full range of 3G, HD and SD standards
are supported by the unit allowing PAL
and NTSC signals up to 1080p 60Hz to be
distributed.
Each output is individually buffered
providing separately driven outputs for
greater reliability.
As well as an AC power input, the unit also
has a 12V DC input for use, for example,
in outside broadcast vehicles. Both power
inputs can be used to give a dual redundant
power supply; whichever voltage is higher
will be used.
Specification For RB-VHDA8
Audio Specification
Compatible
Standards:
SMPTE-424M (3G)
SMPTE-292M (HD-SDI)
SMPTE-259M-AC1 (SD-SDI)
SMPTE 344M1
DVB-ASI2 at 270Mb/s
1
2
Other rates will be passed without reclocking.
Non inverted outputs only.
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Equalisation:
Up to 100m of Belden 1694A
at 1.485Gbps
Up to 350m of Belden 1694A
at 270 Mbps
Data Rates:
Fully supported: 143, 270, 1483.5,
1485, 2967 & 2970Mbps
Passed but not reclocked: 177, 360 &
540Mbps
Reclocking &
Equalisation:
Automatic
Resolution:
Automatic
Return Loss:
>15dB at 1500MHz
Output Jitter:
<0.2UI
Signal Level:
800mV ±10%
Front Panel Indicators
Power LED:
Amber indicates power is present,
Red indicates detection of valid
incoming SDI signal
Rear Panel Connections
HD/SD-SDI Video
Input:
1 x BNC (Unbalanced)
HD/SD-SDI Video
Outputs:
8 x BNC (Unbalanced)
Power Supply:
AC:
Filtered IEC, continuously rated
85-264VAC, 47-63Hz, fused, max 10W
Fuse Rating:
Anti-surge fuse 2A 20 x 5mm
DC:
12V 300mA max, DC supply, 2.5mm
socket fused
Equipment Type
RB-VHDA8:
3G/HD/SD-SDI 1 input, 8 output video
distribution amplifier
Physical Specification
Dimensions
(Raw):
48cm (W) x 10.8cm (D) x 4.3cm (H)
19” (W) x 4.3” (D) x 1.7” (H) (1U)
Dimensions
(Boxed):
58.5cm (W) x 22.5cm (D) x 7cm (H)
23” (W) x 8.9” (D) x 2.8” (H)
Weight :
Nett: 1.3kg Gross: 1.9kg
Nett: 2.9lbs Gross: 4.2lbs
Accessories
RB-RK3:
34
1U Rear panel rack kit for large
Redboxes
Audio & Video Interfaces - Video Embedders & De-Embedders
3G, HD & SD-SDI Embedders & De-Embedders
Still in the familiar Redbox chassis
offering rackmounting as standard and
a universal AC power supply, these new
video Redboxes use latest technology
components to offer embedding and deembedding for analogue and digital audio
signals into and out of single link 3G, HD
and SD-SDI video signals, respectively.
With simple front panel controls, standard
BNC connectivity and remote operation on
serial and ethernet ports, there are currently
6 products in the range:
• RB-VHDMA8 De-Embedder, 8 Channel
Analogue Outputs
• RB-VHEMA8 Embedder, 8 Channel
Analogue Inputs
• RB-VHCMA4 Embedder & De-Embedder,
4 Channel Analogue I/O
• RB-VHCMD16 Embedder & De-Embedder,
16 Channel Digital I/O
• RB-VHEDD8 3G/HD/SD-SDI De-Embedder,
Dolby® E Encoder & Re-Embedder
• RB-VHDDD8 3G/HD/SD-SDI De-Embedder
Dolby® E & Dolby Digital Decoder &
Re-Embedder
Features of the 3G, HD & SD-SDI
Embedders & De-Embedders
• Support all standards of single link 3G, HD
and SD-SDI.
• Front panel control and remote Ethernet
control.
• Two SDI outputs which can be configured
independently.
• Input/output audio gain control,
depending on unit.
• Analogue and digital audio I/O versions.
• AC power supply.
• 1U rackmount.
Common Specifications
SDI Specification
SDI Input:
1 x BNC, 3G/HD/SD-SDI
SDI Outputs:
2 x BNC, 3G/HD/SD-SDI, Re-clocked
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Output
Alignment Jitter:
<0.2UI (3G <0.3UI)
Output Level:
800mV +/- 10%
Return Loss:
<15dB at 1500MHz
SDI Supported
Standards:
270Mbps, SMPTE-259M-C (SD)
1.485 or 1.4835Gbps, SMPTE-292M (HD)
2.97 or 2.967Gbps, SMPTE-424M (3G)
Supported
Video Formats:
525/59.94 (SMPTE-125M)
625/50 (ITU-R BT.656)
720p/23.98, 24, 25, 29.97, 30, 50,
59.94, 60 (SMPTE-296M)
1035i/59.94, 60 (SMPTE-260M)
1080i/50, 59.94, 60 (SMPTE-274M)
1080p/23.98, 24, 25, 50, 59.94, 60
(SMPTE-274M)
1080pSF/23.98, 24, 25,
29.97, 30 (RP-211) 1080i/50
(SMPTE-295M)1080p/50 (SMPTE-295M)
Video Delay:
SD: 290 pixels / 22 us
HD: 570 pixels / 8 us
3G: 570 pixels / 4 us
Embedded
Audio:
48kHz, synchronous
SMPTE-272M-ABC
SMPTE-299M
Serial Port:
RS232, 9 way D-type
Ethernet Port:
10/100Mbps
Fuse Rating:
Anti-surge fuse 2A 20 x 5mm
Physical Specifications
Dimensions
(Raw):
48cm (W) x 15.8cm (D*) x 4.2cm (H) (1U)
19” (W) x 6.2” (D*) x 1.7” (H) (1U)
Dimensions
(Boxed):
59cm (W) x 27.5cm (D*) x 11cm (H)
23.2” (W) x 10.8” (D*) x 4.3” (H)
Weight:
Nett: 1.8kg Gross:
Nett: 4.0lb Gross:
2.3kg
5.1lb
Accessories
RB-RK3:
1U Rear panel rack kit for large
Redboxes
* Note that this product is deeper than standard Redboxes
35
Audio & Video Interfaces - Video Embedders & De-Embedders
RB-VHDMA8 3G/HD/SD-SDI De-Embedder, 8 Channel Analogue Output
Category: 3G/HD/SD-SDI Video
Embedders & De-Embedders.
Product Function: De-embeds from SDI
stream to 8 analogue outputs.
36
Typical Applications: De-embed audio
from live HD football feed to provide multilanguage radio commentaries, de-embed
audio to pass to analogue speaker system.
Features: Balanced & unbalanced analogue
outputs, independent output level controls,
2 x independent SDI outputs, Ethernet or
front panel control & configuration.
The RB-VHDMA8 is an 8 channel analogue
de-embedder contained in a single 19-inch
rack unit. The unit can selectively deembed up to 8 channels within any audio
group of an SDI video signal, to any of the
analogue outputs. After which, the video
becomes two independent paths, where
the audio groups can be selectively deleted
or passed through and then sent to the reclocked SDI outputs.
The de-embedding channel routing is
controlled via the front panel buttons and
indicators. There is also LED indication for SDI
input status and audio group presence.
The unit can be remote controlled via
Ethernet or serial port connections using the
Sonifex SCi software.
It has a triple rate SDI receiver with automatic
input rate detection and equalisation along
with two re-clocked and individually buffered
SDI outputs. It supports the full range of
Audio & Video Interfaces - Video Embedders & De-Embedders
SDI
Input
SDI
Receiver
Audio
Group
Detect
Audio
Pass /
Delete
SDI
Transmitter
Audio
De-embedder
Audio
Pass /
Delete
SDI
Transmitter
EQ
SDI
Output 1
CD
SDI
Output 2
CD
SCi Audio Output Levels Page.
Analogue
Outputs
Balanced
Unbalanced
Bank 1
DACs
Bank 2
DACs
4
4
8 Channel
Level
Control
8
Channel
Routing
RB-VHDMA8 System Block Diagram.
single link 3G, HD and SD standards from
NTSC and PAL up to 1080p 60Hz. There is
independent level control for each analogue
output channel, which can be adjusted from
-24dB through to +24dB in 0.5dB steps.
The analogue outputs have three full-scale
gain settings which can be set via jumpers
inside the unit. Allowable settings are
+12dBu, +18dBu and +24dBu reference FSD.
The balanced and unbalanced output
connections are paralleled, allowing one
type to be used per output.
Specification For RB-VHDMA8
Audio Specification
Signal to Noise:
Better than -106dB (RMS
A-weighted at 24-bit, balanced)
Distortion and Noise:
Better than -85dB THD+N at
1kHz (balanced)
De-embed Bank Select: Bank 1 or 2
De-embed Delay:
3G/HD/SD: 1.1 ms
Bank Channel Select:
Channels 1,2,3 or 4
Connections
Group Select:
Groups 1,2,3 or 4
Group Channel Select:
Group channels 1,2,3 or 4
Analogue Audio
Outputs:
Status:
1 x SDI input status LED
2 x SDI output LEDs
4 x Audio group status LEDs
8 output channels via BNC
(unbalanced) or D-type
socket (balanced)
Analogue Audio
8 x BNC
Front Panel Controls & Indicators
Audio Specifications
Max Output Level:
+24dBu (balanced)
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Gain Range:
12dBu, 18dBu or 24dBu ref FSD
(jumper selectable)
Connectors:
2 x 25-way D-type
Power Supply:
Universal filtered IEC,
continuously rated 85-264VAC
@47-63Hz, fused, max 14W
SCi Delay Page.
Equipment Type
RB-VHDMA8
3G/HD/SD-SDI De-embedder,
8 channel analogue outputs
SCi System Page.
37
Audio & Video Interfaces - Video Embedders & De-Embedders
RB-VHEMA8 3G/HD/SD-SDI Embedder, 8 Channel Analogue Inputs
Category: 3G/HD/SD-SDI Video
Embedders & De-Embedders.
Product Function: Embeds 8 analogue
inputs to 2 x SDI streams.
38
Typical Applications: Embed different
language commentaries to live HD football
feed, embed remixed 5.1 + stereo music
onto HD concert footage.
Features: Balanced & unbalanced analogue
inputs, independent input level controls,
2 x independent SDI outputs, Ethernet or
front panel control & configuration.
The RB-VHEMA8 is an 8 channel analogue
embedder contained in a single 19-inch
rack unit. The unit can selectively embed
up to 8 analogue channels onto either of
the two output video paths which are sent
to the re-clocked SDI outputs. It also has
the capability to allow audio groups to be
deleted or passed through on each of the
two video paths prior to the embedding
process.
The embedding channel routing is controlled
via the front panel buttons and indicators.
There is also LED indication for SDI input
status and audio group presence.
The unit can be remote controlled via
Ethernet or serial port connections using the
Sonifex SCi software.
It has a triple rate SDI receiver with
automatic input rate detection and
equalisation along with two re-clocked
and individually buffered SDI outputs. It
supports the full range of single link 3G, SD
Audio & Video Interfaces - Video Embedders & De-Embedders
and HD standards from NTSC and PAL up to
1080p 60Hz.
Specification For RB-VHEMA8
Audio Specification
Distortion and Noise:
Better than -100dB THD+N at
1kHz (balanced)
Embed Delay:
SD: 600 us
3G/HD: 300 us
Front Panel Controls & Indicators
There is independent level control for each
analogue input channel, which can be
adjusted from -24dB through to +24dB in
0.5dB steps. The analogue inputs have three
full-scale gain settings which can be set via
jumpers situated inside the unit. Allowable
settings are +12dBu, +18dBu and +24dBu
for FSD.
The balanced and unbalanced input
connections are paralleled, allowing one
type to be used per input.
Embed Bank Select:
Bank 1 or 2
Bank Channel Select:
Channels 1, 2, 3 or 4
Group Select:
Groups 1, 2, 3 or 4
Group Channel Select:
Group channels 1,2,3 or 4
Analogue Audio
Inputs:
Status:
1 x SDI input status LED
2 x SDI output LEDs
4 x Audio group status LEDs
8 input channels via BNC
(unbalanced) or D-type socket
(balanced)
Analogue Audio
Connectors:
8 x BNC
2 x 25-way D-type
Power Supply:
Universal filtered IEC,
continuously rated 85-264VAC
@47-63Hz, fused, max 18W
Audio Specifications
Maximum Input Level:
+27dBu (balanced)
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Input Levels:
+12dBu, +18dBu or +24dBu for
FSD (jumper selectable)
Signal to Noise:
Better than -113dBFS (RMS
A-weighted at 24-bit, balanced)
Audio
Group
Detect
SDI
Input
EQ
SDI
Receiver
Connections
Audio
Pass /
Delete
Audio
Embedder
SDI
Transmitter
Audio
Pass /
Delete
Audio
Embedder
SDI
Transmitter
SCi Audio Inputs Page.
Equipment Type
RB-VHEMA8
3G/HD/SD-SDI embedder,
8 channel analogue inputs
SDI
Output 1
SCi Delay Page.
CD
SDI
Output 2
CD
Analogue
Inputs
Channel
Routing
RB-VHEMA8 System Block Diagram.
8
8 Channel
Level
Control
4
Bank 1
ADCs
4
Bank 2
ADCs
Balanced
Unbalanced
SCi Embed Page.
39
Audio & Video Interfaces - Video Embedders & De-Embedders
RB-VHCMA4 3G/HD/SD-SDI Embedder & De-Embedder 4 Channel Analogue I/O
Category: 3G/HD/SD-SDI Video
Embedders & De-Embedders.
Product Function: De-embeds from 1
SDI stream to 4 analogue outputs &
embeds 4 analogue inputs to 2 x SDI
streams.
Typical Applications: To add post
production audio processes,
commentary & sound effects to an HD
video feed (De-embed, add effects,
re-embed).
Features: Balanced & unbalanced
analogue inputs & outputs, independent
input & output level controls, 2 x
independent SDI outputs, Ethernet or
front panel control & configuration.
The RB-VHCMA4 is a 4-channel analogue
de-embedder and a 4-channel analogue
embedder combined into a single 19-inch
rack unit. The unit can selectively deembed up to 4 channels within any audio
group of an SDI video signal, to any of the
analogue outputs. After which, the video
becomes two independent paths where
the audio groups can be selectively deleted
or passed through. The unit then embeds
any of the 4 analogue input channels to
available groups within each of the two
video paths, which are then sent to the reclocked SDI outputs.
SDI
Input
EQ
SDI
Receiver
Delay
Audio
De-embedder
Analogue
Outputs
The de-embedding and embedding channel
routing is controlled via the front panel
buttons and indicators. There is also LED
indication for SDI input status and audio
group presence.
The unit can be remote controlled via
Ethernet or serial port connections using the
Sonifex SCi software.
It has a triple rate SDI receiver with
automatic input rate detection and
equalisation along with two re-clocked
and individually buffered SDI outputs. It
supports the full range of single link 3G, SD
and HD standards from NTSC and PAL up to
1080p 60Hz.
Audio
Pass /Delete
Audio
Embedder
SDI
Transmitter
Audio
Pass /Delete
Audio
Embedder
SDI
Transmitter
There is independent level control for each
analogue input and output channel, which
can be adjusted from -24dB through to
+24dB in 0.5dB steps. The analogue outputs
have three full-scale gain settings which can
be set via jumpers situated inside the unit.
Allowable settings are +12dBu, +18dBu
and +24dBu reference FSD Similarly, the
analogue inputs full-scale gain settings
can be set via jumpers inside the unit.
Allowable settings are +12dBu, +18dBu and
+24dBu for FSD.
The balanced and unbalanced connections
are paralleled, allowing one type to be used
per input or output.
SDI
Output 1
CD
SDI
Output 2
CD
Audio
Group
Detect
Analogue
Inputs
Balanced
Balanced
De-embed 4
Bank
DACs
4
Channel
Level
Control
Unbalanced
RB-VHCMA4 System Block Diagram.
40
4
Channel
Routing
Channel
Routing
4
4
Channel
Level
Control
4
Embed
Bank
ADCs
Unbalanced
Audio & Video Interfaces - Video Embedders & De-Embedders
Specification For RB-VHCMA4
De-embed bank or embed bank
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Impedance:
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Bank Select:
Bank Channel
Select:
Channels 1,2,3 or 4
Gain Range:
12dBu, 18dBu or 24dBu ref FSD
(jumper selectable)
Group Select:
Groups 1,2,3 or 4
Signal to Noise:
Group Channel
Select:
Group channels 1,2,3 or 4
Better than -106dB (RMS A-weighted
at 24-bit, balanced)
Status:
1 x SDI input status LED
2 x SDI output LEDs
4 x Audio group status LEDs
Distortion and
Noise:
Better than -85dB THD+N at 1kHz
(balanced)
Front Panel Controls & Indicators
De-embed Delay: 3G/HD/SD: 1.1 ms
Embed Delay:
Audio Specifications
Analogue Inputs
Maximum Input
Level:
SD: 600 us
3G/HD: 300 us
Connections
+27dBu (balanced)
Analogue
Audio Inputs:
4 input channels via BNC (unbalanced)
or D-type (balanced)
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Impedance:
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Analogue
Audio Outputs:
4 output channels via BNC (unbalanced
or D-type (balanced)
Input Levels:
+12dBu, +18dBu or +24dBu for FSD
(jumper selectable)
Analogue
8 x BNC
Audio Connectors: 2 x 25-way D-type socket
Signal to Noise:
Better than -113dBFS (RMS A-weighted
at 24-bit, balanced)
Power Supply:
Distortion and
Impedance:
Better than -100dB THD+N at 1kHz
(balanced)
Max Output
Level:
+24dBu (balanced)
SCi Embed Page.
Universal filtered IEC,
continuously rated 85-264VAC
@47-63Hz, fused, max 16W
Equipment Type
RB-VHCMA4
Analogue Outputs
SCi System Page.
3G/HD/SD-SDI embedder &
de-embedder, 4 channel analogue I/O
41
Audio & Video Interfaces - Video Embedders & De-Embedders
RB-VHCMD16 3G/HD/SD-SDI Embedder & De-Embedder 16 Channel Digital I/O
Category: 3G/HD/SD-SDI Video
Embedders & De-Embedders.
Product Function: De-embeds from
1 SDI stream to 16 digital outputs &
embeds 16 digital inputs to 2 x SDI
streams.
42
Typical Applications: To add post
production audio processes,
commentary & sound effects to an HD
video feed (De-embed, add effects,
re-embed).
Features: Selectable unbalanced or
balanced digital inputs & outputs,
independent input & output level
controls, 2 x independent SDI outputs,
Ethernet or front panel control &
configuration.
The RB-VHCMD16 is a 16-channel deembedder and 16 channel embedder
combined into a single 19-inch rack unit.
The unit can selectively de-embed any
channel within any audio group of an
SDI video signal, to any of the digital
outputs. After which, the video becomes
two independent paths where the audio
groups can be selectively deleted or passed
through. The unit then embeds any of the
digital input channels to available groups
within each of the two video paths, which
are then sent to the re-clocked SDI outputs.
The de-embedding and embedding channel
routing is controlled via the front panel
buttons and indicators. There is also LED
indication for SDI input status and audio
group presence.
The unit can be remote controlled via
Ethernet or serial port connections using the
Sonifex SCi software.
It has a triple rate SDI receiver with
automatic input rate detection and
equalisation along with two re-clocked
and individually buffered SDI outputs. It
supports the full range of single link 3G, HD
and SD standards from NTSC and PAL up to
1080p 60Hz.
Audio & Video Interfaces - Video Embedders & De-Embedders
Each digital input is normally sample rate
converted to 48kHz before embedding, so
that it is synchronous to the video input,
though sample rate conversion can be
bypassed on a per input basis allowing
SMPTE-337M data to be embedded. All
digital outputs are output at 48kHz,
synchronous to the video input. There
is independent level control for each
digital input and output channel, which
can be adjusted from -24dB to +24dB in
0.5dB steps.
The digital audio I/O connections are
transformer-coupled balanced line interfaces
and can be configured to be either 75ohm
(AES 3ID) or 110ohm (AES 3) impedance
through either a BNC or D-type connector.
These connections are paralleled, allowing
one type to be used per input or output.
Specification For RB-VHCMD16
Front Panel Controls & Indicators
Bank Select:
Bank 1, 2, 3 or 4
Bank Channel Select:
Channels 1,2,3 or 4
Function Select:
De-embed or embed
Group Select:
Groups 1,2,3 or 4
Group Channel Select:
Group channels 1,2,3 or 4
Status:
1 x SDI input status LED
2 x SDI output LEDs
4 x Audio group status LEDs
Audio Specifications
SCi Audio Inputs Page.
SCi De-Embed Page.
SDI
Input
Digital
Outputs
Balanced
EQ
Bank 3 4
Digital Audio
Transmitter
Bank 4
Digital 4
Audio TX
Delay
Audio
De-embedder
Bank 1 4
Digital Audio
Transmitter
Bank 2
Digital Audio 4
Transmitter
Unbalanced
SDI
Receiver
16
Channel 16
Level
Control
Channel
Routing
Output Sample Rate:
48kHz
Input Sample Rates:
32-192kHz, sample rate
converted to 48kHz
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Impedance:
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Signal Level
(un-terminated):
Balanced: 3Vp-p +/- 20%
Unbalanced: 2Vp-p +/- 20%
Dynamic Range:
138dB
Distortion and Noise:
<-137dB THD+N at 997Hz,
ref 0dBFS
De-embed Delay:
3G/HD/SD: 330 us
SRC Input Delay:
192 kHz: 1.3 ms
96 kHz: 1.83 ms
48 kHz: 2.9 ms
Embed Delay:
SD: 600 us + SRC Input Delay
3G/HD: 300 us + SRC Input Delay
SCi Embed & SDI Output Page.
Audio
Pass /Delete
Audio
Embedder
SDI
Transmitter
Audio
Pass /Delete
Audio
Embedder
SDI
Transmitter
Audio
Group
Detect
CD
CD
4
Channel
Routing
16
16
Channel
Level
Control
SDI
Output 1
Connections
SDI
Output 2
Bank 1
Digital Audio
RX+SRC
Bank 2
4 Digital Audio
RX+SRC
4
Bank 3
Digital Audio
RX+SRC
Digital
Inputs
Balanced
Digital Audio Outputs:
16 outputs via 8 BNCs (AES 3ID)
or D-type (AES/EBU)
Digital Audio Inputs:
16 inputs via 8 BNCs (AES 3ID) or
D-type (AES/EBU)
Digital Audio
Connectors:
16 x BNC
2 x 25-way D-type socket
Power Supply:
Universal filtered IEC,
continuously rated 85-264VAC
@47-63Hz, fused, max 13W
Equipment Type
RB-VHCMD16
3G/HD/SD-SDI embedder & deembedder 16 channel digital I/O
Unbalanced
Bank 4
4 Digital Audio
RX+SRC
RB-VHCMD16 System Block Diagram.
43
Audio & Video Interfaces - Video Embedders & De-Embedders With Dolby® Encoder
RB-VHEDD8 3G/HD/SD-SDI De-Embedder, Dolby® E Encoder & Re-Embedder
Category: 3G/HD/SD-SDI Video DeEmbedders & Re-Embedders With Dolby
Encode/Decode.
Product Function: De-embeds from 1
x SDI stream, Dolby E Encode process
using de-embedded channels or 8 digital
44
audio inputs, re-embeds any audio or
Dolby encoded channels to 2 x SDI streams.
Typical Applications: To embed audio
in Dolby E encoded format into 2 x SDI
streams.
Features: 8 x balanced or unbalanced
digital audio inputs, independent input
level controls, 2 x independent SDI outputs,
Dolby E bitstream output, 8 programmable
GPIOs, metadata input, Ethernet or front
panel control & configuration.
The RB-VHEDD8 is an SDI audio de-embedder and re-embedder with Dolby® E Encoding
capabilities. The unit de-embeds 16 channels of audio from the SDI input and these,
together with the external digital audio inputs via the BNC or D-type connectors, are
passed to the Dolby® Encoder. Dolby E encodes up to 8 channels of audio into two
channels of an AES/EBU digital audio stream which are then embedded onto any of the
available groups within each of the two SDI output paths. The outputs from the deembedder can also be re-embedded into the SDI outputs, along with the encoder inputs.
The encoded Dolby E bitstream is also available via a dedicated rear panel output.
The unit has a triple rate SDI receiver with
automatic input rate detection and
equalisation along with two re-clocked and
individually buffered SDI outputs which can
be configured independently. It supports
the full range of single link 3G, HD and SD
standards from NTSC and PAL up to 1080p
60Hz.
The unit maintains lip-sync through the
encode process by delaying the video,
embedded audio and any other ancillary
Audio & Video Interfaces - Video Embedders & De-Embedders With Dolby® Encoder
data, including embedded audio, to match
the encoding latency. There is minor
adjustment of this delay, to allow for errors
further up or down the processing chain.
The encoding process metadata can be
selected from either the external 9-pin
D-type, from metadata embedded into the
vertical blanking area of the video input
(SMPTE 2020), or by internal settings.
The unit is controlled locally through the
front panel display and remotely via an
Ethernet connection, using the Sonifex SCi
software. The embedding channel routing is
controlled using these methods also.
Each digital input channel has independent
level control, with adjustment from -24dB to
+24dB in 0.5dB steps. The digital
audio I/O connections are transformercoupled balanced line interfaces and can
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(AES3) impedance through BNC or D-type
connectors respectively. These connections
Metadata
VANC
De-embedder
Metadata
Input
SDI
Input
SDI
Receiver
Audio
De-embedder
SCi Audio I/O Page.
SCi Encoder Control Page.
SCi Encoder Internal Metadata Page.
are paralleled, allowing one type to be used
per input or output.
Output Alignment
Jitter:
<0.2UI (3G <0.3UI)
Digital Audio I/O:
Inputs:
Output Level:
800mV ±10%
8 x digital audio input channels via
4 x BNCs or 25 way D-type
socket(AES3)
Outputs:
2 output channels via 1 x BNC
or 25 way D-type socket (AES3)
Digital Audio
Connectors:
5 x BNC
1 x 25-way D-type socket
LTC Input:
1 x BNC (not used)
8 GPIs and 8 GPOs are available on a 25
way D-type socket, whose function can be
programmed using the menu, e.g. alarm
outputs for loss of input or encoder errors.
270Mbps, SMPTE-259M-C (SD)
1.485 or 1.4835Gbps, SMPTE-292M
(HD)
2.97 or 2.967Gbps, SMPTE-424M
(3G), SMPTE-425M-A
SDI Specification
SDI Input:
1 x BNC, 3G/HD/SD-SDI
SDI Outputs:
2 x BNC, 3G/HD/SD-SDI, re-clocked
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Audio
Pass /
Delete
Audio
Embedder
SDI
Output 1
SDI
Transmitter
CD
Audio
Embedder
EQ
Metadata:
SDI
Output 2
SDI
Transmitter
Embedded Audio:
Video Delay:
Adjustment:
Display:
Vacuum fluorescent display
System Navigation:
Rotary selector with integral pushswitch
8
Channel
Routing
& Level
Control
8
Dolby E
Encoder
2
Additional Connections
Ethernet Port:
10/100Mbps, RJ-45
Remote
Input/Output Port:
25-way ‘D’-type socket
Power Supply:
Universal filtered IEC, continuously
rated 85-264VAC @47-63Hz, fused
24-bit, 48kHz synchronous
SMPTE-272M-C & SMPTE-299M
Fuse Rating:
Anti-surge fuse 2A 20 x 5mm
SMPTE-2020M
SMPTE-RDD06, 9-Pin D-type socket
Dimensions
(Raw):
48cm (W) x 15.8cm (D*)x 4.2cm (H)
19” (W) x 6.2 ” (D*) x 1.7” (H) (1U)
Dimensions
(Boxed):
59cm (W) x 27.5cm (D) x 11cm (H)
23.2” (W) x 10.8” (D) x 4.3” (H)
Weight:
Nett: 1.8kg Gross: 2.3kg
Nett: 4.0lb Gross: 5.1lb
1 frame when frame rate <= 30 fps,
2 frames when frame rate > 30 fps.
Physical Specifications
±10 ms.
* Note that this product is deeper than standard Redboxes
Audio Specifications
AES RX/TX
and
Termination
Select
Operational Control
525/59.94 (SMPTE-125M)
625/50 (ITU-R BT.656)
720p/23.98, 24, 25, 29.97, 30, 50,
59.94, 60 (SMPTE-296M)
1035i/59.94, 60 (SMPTE-260M)
1080i/50, 59.94, 60 (SMPTE-274M)
1080p/23.98, 24, 25, 50, 59.94, 60
(SMPTE-274M)
1080pSF/23.98, 24, 25, 29.97, 30
(RP-211)
1080i/50 (SMPTE-295M)
1080p/50 (SMPTE-295M)
Delay Specification
CD
16
Digital
Audio
I/O
<15dB at 1500MHz
SDI Supported
Standards:
Supported Video
Formats:
Specification For RB-VHEDD8
Audio
Pass /
Delete
Delay
Return Loss:
Sample Rate:
48kHz for output
Equipment Type
Input Sample
Rates:
CH1/2 & CH3/4: 32-192kHz
CH5/6 & CH7/8: 32-48kHz
RB-VHEDD8
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Accessories
Signal Level
(Un-terminated):
Unbalanced: 1Vp-p ±20%
Balanced: 6.6Vp-p ±20%
RB-RK3:
3G/HD/SD-SDI Dolby® E encoder &
embedder
1U Rear panel rack kit for large
Redboxes
RB-VHEDD8 Block Diagram.
45
Audio & Video Interfaces - Video Embedders & De-Embedders With Dolby® Decoder
RB-VHDDD8 3G/HD/SD-SDI De-Embedder Dolby® E & Dolby Digital Decoder & Re-Embedder
Category: 3G/HD/SD-SDI Video DeEmbedders & Re-Embedders With Dolby
Encode/Decode.
Product Function: De-embeds 16
channels from 1 x SDI stream, decodes
any embedded Dolby E or D audio,
46
outputs audio to 12 channels and reembeds to 2 x SDI streams.
Typical Applications: To de-embed audio
in Dolby E encoded format from an SDI
stream and re-embed into 2 x SDI streams.
Features: 12 x balanced or unbalanced
digital audio outputs, independent output
level controls, 2 x independent SDI outputs,
Dolby E bitstream output, 8 programmable
GPIOs, metadata output, Ethernet or front
panel control & configuration.
The RB-VHDDD8 is an SDI audio de-embedder and re-embedder with Dolby E & Dolby
Digital Decoding capabilities. It de-embeds all 16 available audio inputs and decodes a
selected Dolby E or Dolby Digital bitstream embedded in the SDI input. The outputs from
the decoder and the de-embedder can then be re-embedded onto either of the two SDI
outputs and also transmitted on BNCs or D-types. The encoded Dolby bitstream is also
available via a dedicated output on the rear panel.
The unit has a triple rate SDI receiver
with automatic input rate detection and
equalisation along with two re-clocked and
individually buffered SDI outputs which are
configured independently. It supports the full
range of single link 3G, HD and SD standards
from NTSC and PAL up to 1080p 60Hz.
The decoder metadata output is transmitted
using RS-485 via the external 9-pin D-type
and can also be embedded into the vertical
blanking space (SMPTE-2020) onto either of
the two SDI outputs.The unit is controlled
locally through the front panel display but
can be remote controlled via an Ethernet
Audio & Video Interfaces - Video Embedders & De-Embedders With Dolby® Decoder
59.94, 60 (SMPTE-296M)
1035i/59.94, 60 (SMPTE-260M)
1080i/50, 59.94, 60 (SMPTE-274M)
1080p/23.98, 24, 25, 50, 59.94, 60
(SMPTE-274M)
1080pSF/23.98, 24, 25, 29.97, 30
(RP-211)
1080i/50 (SMPTE-295M)
1080p/50 (SMPTE-295M)
Embedded Audio:
24-bit, 48kHz synchronous
SMPTE-272M-C and SMPTE-299M
Metadata:
SMPTE-2020M
SMPTE-RDD06, 9-Pin D-type
Delay Specification
SCi Decoder Page.
SCi External Audio Outputs Page.
SCi Embed Control Page.
Video Delay:
1 frame when frame rate <= 30 fps,
2 frames when frame rate > 30 fps.
Adjustment:
±10 ms.
Audio Specifications
Audio
Pass /
Delete
SDI
Input
SDI
Receiver
Audio
De-embedder
Audio
Embedder
Audio
Pass /
Delete
Delay
Audio
Embedder
Metadata
VANC
Embed
Metadata
VANC
Embed
Output Sample Rate: 48kHz
SDI
Output 1
SDI
Transmitter
CD
SDI
Output 2
SDI
Transmitter
EQ
CD
2
16
Dolby E &
Dolby Digital
Decoder
RB-VHDDD8 Block Diagram.
connection using the Sonifex SCi software,
through which metadata can also be viewed.
The unit maintains lip-sync through the
decode process by delaying the video,
embedded audio and any other ancillary
data to match the decoding latency. There is
minor adjustment of this delay, to allow for
errors up or down the processing chain.
Each digital output channel has independent
level control which can be adjusted from
-24dB through to +24dB in 0.5dB steps.
10
Channel
Routing
& Level
Control
AES
Transmitter
&
Termination
Select
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Signal Level
(Un-terminated):
Unbalanced: 1Vp-p +/- 20%
Balanced: 6.4Vp-p +/- 20%
Digital Audio
Outputs:
12 output channels via 6 x BNC or
25 way D-type socket (AES3)
Digital Audio
Connectors:
6 x BNC
1x 25-way D-type socket
Operational Control
Digital
Audio
Outputs
Display:
Vacuum fluorescent display
System Navigation:
Rotary selector with integral pushswitch
Additional Conections
Metadata
Output
The digital audio output connections are
transformer-coupled balanced line interfaces
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through either a BNC or via the D-type
connector. These output connections are
paralleled, allowing one type to be used per
output.
8 GPIs and 8 GPOs are available on a 25
way D-type socket whose function can be
programmed using the menu, e.g. alarm
outputs for loss of input or decoder errors.
Specification For RB-VHDDD8
SDI Specification
SDI Input:
1 x BNC, 3G/HD/SD-SDI
SDI Outputs:
2 x BNC, 3G/HD/SD-SDI, re-clocked
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ϳϱɏ
Output Alignment
Jitter:
<0.2UI (3G <0.3UI)
Output Level:
800mV ±10%
Ethernet Port:
10/100Mbps, RJ-45
Remote
Input/Output Port:
25-way ‘D’-type socket
Power Supply:
Universal filtered IEC, continuously
rated 85-264VAC @47-63Hz, fused
Fuse Rating:
Anti-surge fuse 2A 20 x 5mm
Physical Specifications
Dimensions
(Raw):
48cm (W) x 15.8cm (D*)x 4.2cm (H)
19” (W) x 6.2” (D*) x 1.7” (H) (1U)
Dimensions
(Boxed):
59cm (W) x 27.5cm (D) x 11cm (H)
23.2” (W) x 10.8” (D) x 4.3” (H)
Weight:
Nett: 1.8kg
Nett: 4.0lb
Gross: 2.3kg
Gross: 5.1lb
Return Loss:
<15dB at 1500MHz
* Note that this product is deeper than standard Redboxes
SDI Supported
Standards:
270Mbps, SMPTE-259M-C (SD)
1.485 or 1.4835Gbps, SMPTE-292M
(HD)
2.97 or 2.967Gbps, SMPTE-424M
(3G), SMPTE-425M-A
Equipment Type
Supported Video
Formats:
525/59.94 (SMPTE-125M)
625/50 (ITU-R BT.656)
720p/23.98, 24, 25, 29.97, 30, 50,
RB-RK3:
RB-VHDDD8
3G/HD/SD-SDI Dolby® E & Dolby
Digital decoder & de-embedder
Accessories
1U Rear panel rack kit for large
Redboxes
47
Audio & Video Interfaces - Dolby® E & Dolby® Digital Encoders & Decoders
RB-DEED8 Dolby® E Encoder 8 Channel, Digital Inputs
The RB-DEED8 is a Dolby® E Encoder. It
encodes up to 8 digital audio channels into
a single Dolby bitstream output. There is
also an extra Dolby output for connection
to monitoring equipment.
Category: Dolby Encoders & Decoders.
Product Function: Encodes up to 8
digital audio channels into a single Dolby
E bitstream output.
Typical Applications: To produce a
Dolby E encoded stream for distribution
around a facility as a stereo AES3 pair.
Features: 8 x balanced or unbalanced
digital audio inputs, independent input
level controls, 2 x Dolby E bitstream
outputs, genlock input, 8 programmable
GPIOs, metadata input, Ethernet or front
panel control & configuration.
SCi Encoder Metadata Status Page.
48
Each digital input channel has independent
level control, which can be adjusted from
-24dB through to +24dB in 0.5dB steps. The
digital inputs and outputs can be selected to
be either balanced or unbalanced AES/EBU
specification, and can be connected through
either BNC or D-type connectors. These I/O
connections are paralleled, allowing one
type to be used per input or output.
Metadata can be input to the encoder using
RS-485 via the external 9-pin D-type socket
on the rear panel and can also be viewed
using the SCi software.
A video input is used to genlock the audio
output and Dolby® E encoding to a known
video reference. This input autodetects
between NTSC, PAL or Tri-level sync.
8 general purposes inputs and 8 outputs
are available on a rear panel 25 way D-type
socket whose function can be programmed
using the menu, e.g. alarm outputs for loss
of input or encoder errors.
The unit is controlled locally using the
front panel display and control knob but
can be remote controlled via an ethernet
connection using the Sonifex SCi software.
Specification For RB-DEED8
Specification
Output Sample Rate: 48kHz
Input Sample Rates:
CH1/2 & CH3/4: 32-192kHz
CH5/6 & CH7/8: 32-48kHz
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Impedance:
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Signal Level
(un-terminated):
Unbalanced: 1Vp-p +/- 20%
Balanced: 6.6Vp-p +/- 20%
Digital Audio I/O:
Inputs:
8 input channels via 4 x BNCs or
25 way D-type socket (AES3)
Dolby Output 1:
1 x Dolby bitstream output via BNC
Dolby Output 2:
1 x Dolby bitstream output via BNC
or 25 way D-type socket (AES3)
Digital Audio
Connectors:
6 x BNC
1 x 25-way D-type socket
Metadata:
SMPTE-RDD06, 9 way D-type socket
Video Reference
Reference Type:
Autodetect NTSC, PAL or Tri-level
sync
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ϳϱɏ
Connector:
1 x BNC
LTC Input:
1 x BNC (not used)
Dolby® and the double-D symbol are registered trademarks of Dolby Laboratories.
Operational Control
SCi Encoder Metadata Control Page.
SCi Encoder Control Page.
SCi Remote Outputs Page.
Audio & Video Interfaces - Dolby® E & Dolby® Digital Encoders & Decoders
Display:
Vacuum fluorescent display
System Navigation:
Rotary selector with integral pushswitch
Metadata
Additional Connections
Ethernet Port:
10/100Mbps, RJ-45
Remote
Input/Output Port:
25-way ‘D’-type socket
Serial Port:
RS232, 9 way D-type
Power Supply:
Universal filtered IEC, continuously
rated 85-264VAC @47-63Hz, fused
Fuse Rating:
Anti-surge fuse 1A 20 x 5mm
CH12
Termination
Select
AES
Digital Audio
Receiver
CH34
Termination
Select
AES
Digital Audio
Receiver
CH56
Termination
Select
AES
Digital Audio
Receiver
CH78
Termination
Select
AES
Digital Audio
Receiver
VR
Video Sync
Input
Physical Specifications
Dimensions
(Raw):
48cm (W) x 15.8cm (D*)x 4.2cm (H)
19” (W) x 6.2” (D*) x 1.7” (H) (1U)
Dimensions
(Boxed):
59cm (W) x 27.5cm (D*) x 11cm (H)
23.2” (W) x 10.8” (D*) x 4.3” (H)
Weight:
Nett: 1.8kg
Nett: 4.0lb
Gross: 2.3kg
Gross: 5.1lb
Level
Control
Dolby E
Encoder
AES
Digital Audio
Transmitter
Termination
Select
DO
AES
Digital Audio
Transmitter
Termination
Select
DO
* Note that this product is deeper than standard Redboxes
Equipment Type
RB-DEED8
Dolby® E encoder 8 channel, digital
inputs
Accessories
RB-RK3:
1U Rear panel rack kit for large
Redboxes
PLL
Digital
Audio
IO
RB-DEED8 Block Diagram.
49
Audio & Video Interfaces - Dolby® E & Dolby® Digital Encoders & Decoders
RB-DEDD8 Dolby® E & Dolby Digital Decoder 8 Channel, Digital Outputs
Category: Dolby Encoders & Decoders.
Product Function: Decodes up to 10
digital audio channels from a Dolby E or
Dolby Digital bitstream.
Typical Applications: To decode 5.1 or
7.1 audio for a surround sound amplifier.
Features: 10 x balanced or unbalanced
digital audio outputs (8 digital audio, 2
Aux), independent output level controls,
1 x Dolby E or D bitstream input, 1 x
Dolby E or D bitstream output, genlock
input, 8 programmable GPIOs, metadata
output, Ethernet or front panel control &
configuration.
Dolby® and the double-D symbol are registered trademarks of Dolby Laboratories.
50
The RB-DEDD8 is a Dolby® E & Dolby Digital
Decoder. It decodes an incoming Dolby
bitstream and transmits the decoded
outputs on BNC or D-type connectors on
the rear panel. The Dolby input is also
looped through to allow connection to
other Dolby receiver equipment.
Each digital output channel has independent
level control, which can be adjusted from
-24dB through to +24dB in 0.5dB steps. The
digital inputs and outputs can be selected to
be either balanced or unbalanced AES/EBU
specification, and can be connected through
either BNC or D-type connectors. These I/O
connections are paralleled, allowing one
type to be used per input or output.
The metadata output from the decoder is
transmitted using RS-485 via the external
9-pin D-type socket on the rear panel and
can also be viewed using the SCi software.
A video input is used to genlock the audio
outputs and Dolby® E decoding to a known
video reference. This input autodetects
between NTSC, PAL or Tri-level sync.
8 general purposes inputs and 8 outputs
are available on a rear panel 25 way D-type
socket whose function can be programmed
using the menu, e.g. alarm outputs for loss
of input or decoder errors.
The unit is controlled locally through the
front panel display but can be remote
controlled via an ethernet connection using
the Sonifex SCi software.
Audio & Video Interfaces - Dolby® E & Dolby® Digital Encoders & Decoders
Specification For RB-DEDD8
Metadata:
Specification
Video Reference
Output Sample Rate: 48kHz
Reference Type:
Input Sample Rates:
32-48kHz
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Impedance:
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Signal Level
(Un-terminated):
Unbalanced: 1Vp-p ±20%
Balanced: 6.6Vp-p ±20%
Digital Audio I/O:
Dolby Input:
1 x Dolby bitstream input via BNC
Outputs:
SMPTE-RDD06, 9 way D-type socket
Power Supply:
Universal filtered IEC, continuously
rated 85-264VAC @47-63Hz, fused
Autodetect NTSC, PAL or Tri-level
sync
Fuse Rating:
Anti-surge fuse 1A 20 x 5mm
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ϳϱɏ
Connector:
1 x BNC
Dimensions
(Raw):
48cm (W) x 15.8cm (D*)x 4.2cm (H)
19” (W) x 6.2” (D*) x 1.7” (H) (1U)
LTC Output:
1 x BNC (not used)
Dimensions
(Boxed):
59cm (W) x 27.5cm (D*) x 11cm (H)
23.2” (W) x 10.8” (D*) x 4.3” (H)
Weight:
Nett: 1.8kg
Nett: 4.0lb
Operational Control
8 x Digital audio output channels &
2 x Auxiliary audio output (stereo
downmix) channels via 5 x BNCs or
25 way D-type socket
Display:
Vacuum fluorescent display
System Navigation:
Rotary selector with integral pushswitch
Additional Connections
Ethernet Port:
10/100Mbps, RJ-45
Dolby Output:
1 x Dolby bitstream output via BNC
or D-type
Remote
Input/Output Port:
25-way ‘D’-type socket
Digital Audio
Connectors:
7 x BNC
1 x 25 way D-type socket
Serial Port:
RS232, 9 way D-type socket
Digital
Audio
IO
DI
VR
Termination
Select
Video Sync
Input
AES
Digital Audio
Receiver
PLL
Dolby E &
Dolby Digital
Decoder
Channel
Routing &
Level Control
Physical Specifications
Gross: 2.3kg
Gross: 5.1lb
SCi Decoder PCM Metadata Page.
* Note that this product is deeper than standard Redboxes
Equipment Type
RB-DEDD8
Dolby® E & Dolby Digital decoder 8
channel, digital outputs
Accessories
RB-RK3:
1U Rear panel rack kit for large
Redboxes
AES
Digital Audio
Transmitter
Termination
Select
DO
AES
Digital Audio
Transmitter
Termination
Select
CH12
AES
Digital Audio
Transmitter
Termination
Select
CH34
AES
Digital Audio
Transmitter
Termination
Select
CH56
AES
Digital Audio
Transmitter
Termination
Select
CH78
AES
Digital Audio
Transmitter
Termination
Select
AUX
SCi Audio I/O Page.
Metadata
RB-DEDD8 Block Diagram.
SCi Decoder Control Page.
51
Audio & Video Interfaces - Digital Audio Converters
RB-ADDA Combined A/D and D/A Converter
Category: Digital Audio Converters.
Product Function: Converting stereo
analogue audio to digital & vice-versa.
Typical Applications:
• Interfacing professional or domestic
audio devices e.g. ipods, cassette
recorders or telephone balance units
into digital environments.
Features:
• Balanced & unbalanced inputs.
• Balanced & unbalanced outputs.
• AES/EBU & S/PDIF I/O.
• Sample rates up to 24 bit 96kHz.
The RB-ADDA A/D and D/A converter is a
1U rack-mount which produces an AES/EBU
or S/PDIF level digital audio output from a
balanced XLR or unbalanced phono stereo
audio input. It also produces a stereo
balanced XLR or unbalanced phono output
from an incoming AES/EBU or S/PDIF digital
input signal.
The unit operates in four modes:
Master Mode - In this mode the unit
receives an analogue audio signal, which
is digitised and formatted for digital serial
transmission (IEC958). The necessary clock
signals are generated internally from an
on board master clock at a selectable rate
(32kHz, 44.1kHz, 48kHz or 96kHz).
Slave Mode - In this mode the unit
automatically detects the presence of a
digital audio sync signal, if present at the
digital input, and synchronises the digital
output to it. If no sync is present, no output
will be generated.
Auto Mode - Here the unit synchronises to
the digital audio sync signal if present at the
digital input and uses the internal master
clock only if no sync input signal is detected.
In this case, the internal master clock is used
at the selected sample rate.
Auto Lock Mode - This operates like
the auto mode except that if no sync input
signal is detected, it will use the internal
master clock to sync to the sample rate
which was last clocked to.
When operating in sync modes, the front
panel power LED flashes whenever the unit
is not synchronised to the incoming digital
signal.
52
The analogue inputs have left and right level
controls using pre-set potentiometers and
DIP switches allowing a signal range from
+9dBu to +27dBu. The RCA phono inputs
have a further 10dB gain incorporated to
give a total gain range of -1dBu to +17dBu
for full-scale digits.
The analogue outputs have an output
level control, allowing full-scale settings
selectable from +12dBu, +18dBu or +24dBu.
There are factory-set internal level controls
for the analogue outputs allowing gain
adjustment of ±1dB.
There are buttons to select either the AES/
EBU or S/PDIF input or output for the D/A
and A/D sections respectively.
The output bit depth can be selected from
16, 20 or 24 bits. Inputs of a different bit
depth to the output are dithered using a
noise filter.
For the digital output, there is a switch
available to define the content of the
channel status bits embedded within the
digital audio stream. The status bits can be
forced to either Professional or Consumer
Mode.
Audio & Video Interfaces - Digital Audio Converters
Additionally, if de-emphasis is selected, the
RB-ADDA will decode 50/15μs emphasis
when indicated by certain channel status
bits in the incoming digital audio data.
Specification For RB-ADDA
Maximum
Output Level:
+24dBu balanced output, +14dBu
unbalanced output
The RB-ADDA has a calibration routine for
optimum performance, which allows the
noise floor and dynamic range to improve
by 1-2dB.
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Impedance:
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Audio Specification
Digital to Analogue Conversion
D/A Audio Specification For RB-ADDA
Gain Range:
Adjustable input gain of ±3dB on
12dBu, 18dBu or 24dBu, ref FSD
Input Level
Adjust:
DIP switch & pre-set pots
Distortion
and Noise:
>96dB THD + N at 1kHz
Sample Rates:
Master rates of 32kHz, 44.1kHz, 48kHz
or 96kHz, or can synchronise to
incoming 32kHz to 100kHz sample rate
Analogue
Inputs:
2 x XLR 3 pin (balanced)
2 x RCA phono (unbalanced)
Bit Depth:
16, 20 or 24 bits via DIP switch
Modes &
Frequencies:
16 way rotary DIP switch
Channel
Status Bits:
Forced to consumer mode or
professional mode, via DIP switch
Output Level
Adjust:
DIP switch
A/D Connections
Dynamic Range:
>100dB
Digital Outputs:
Gain Range:
Selectable 12dBu, 18dBu or 24dBu
output level, ref FSD
1 x AES/EBU XLR 3 pin male
1 x S/PDIF RCA phono
Mains Input:
Filtered IEC, switchable 110-120V, or
220-240V, fused, 10W max
Fuse Rating:
Anti-surge fuse 100mA 20 x 5mm
(230VAC)
Anti-surge fuse 250mA 20 x 5mm
(115VAC)
D/A Connections
The calibration cycle operates by calibrating
the gain and the zero reference of the A/D
converter.
Digital Inputs:
1 x AES/EBU XLR 3 pin female
1 x S/PDIF RCA phono
Analogue
Outputs:
2 x XLR 3 pin male (balanced)
2 x RCA phono (unbalanced)
Analogue to Digital Conversion
A/D Audio Specification For RB-ADDA
Maximum
Input Level:
+27dBu balanced inputs, +17dBu
unbalanced inputs
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Impedance:
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balanced
Dynamic Range:
>110dB
Operational Controls
Analogue
Input Select:
XLR or phono, via push-switch
Digital
Output Select:
AES/EBU or S/PDIF, via push-switch
Digital
Input Select:
AES/EBU or S/PDIF, via push-switch
De-emphasis
On/Off:
DIP switch
Equipment Type
RB-ADDA:
Combined A/D and D/A converter
Physical Specification
Dimensions:
(Raw):
48cm (W) x 10.8cm (D) x 4.2cm (H) (1U)
19” (W) x 4.3” (D) x 1.7” (H) (1U)
(Boxed):
58.5cm (W) x 22.5cm (D) x 7cm (H)
23” (W) x 8.9” (D) x 2.8” (H)
Weight:
Nett: 1.6kg Gross: 2.2kg
Nett: 3.5lbs Gross: 4.8lbs
53
Audio & Video Interfaces - Digital Audio Converters
RB-ADDA2 A/D and D/A Converter, 24 bit 192kHz
192
Category: Digital Audio Converters.
Description: Combined A/D and D/A
Converter (24 bit, 192kHz Capable) (1U).
Product Function: Converting stereo
analogue audio to digital & vice-versa.
Typical Applications:
• Interfacing professional or domestic
54
audio devices i.e. ipods or telephone
balance units into digital environments.
• Recording studios/post production
where easy access to settings is
required.
Features:
• Front panel switches.
• Coax/optical digital I/O.
• Separate sync for A to D and D to A
allows independent use.
• RS232 for remote setting/control via
Sonifex SCi software.
The RB-ADDA2 A/D and D/A converter
is a 1U rack-mount which produces an
AES/EBU, S/PDIF or TOSlink optical level
digital audio output from a balanced
XLR or unbalanced phono stereo audio
input. It also produces a stereo balanced
XLR or unbalanced phono output from
an incoming AES/EBU, S/PDIF or TOSlink
optical digital input signal.
The RB-ADDA2 is a high performance,
enhanced version of the RB-ADDA providing
the following additional features:
• It supports higher sample frequency rates up
to and including 176.4kHz and 192kHz.
• It has additional independent AES/EBU
and Word Clock synchronising inputs, so
that the A/D and D/A sections can operate
independently, with the digital outputs
synchronised to an external master reference
clock.
• It has TOSlink optical digital audio I/O.
• It has front panel push-button switches for all
the main settings. The buttons are arranged in
sets, where pressing the button advances the
current selection and LED indicator.
Audio & Video Interfaces - Digital Audio Converters
• An RS232 port allows RB-ADDA2 settings
to be controlled remotely. The front panel
LED indicators alter automatically when
using RS232 commands.
The A/D SOURCE push-button is used
to select from either the balanced or
unbalanced stereo analogue inputs and this
push-button also defines the input level for
full scale digits at one of +12dBFS, +18dBFS or
+24dBFS. These values can then be fine-tuned
by using rear-panel pre-set potentiometers
which give another ±3dB of gain adjustment,
allowing a signal range from +9dBu to
+27dBu. The RCA phono inputs have a further
10dB gain incorporated to give a total gain
range of -1dBu to +17dBu for full-scale digits.
For the digital output, there are three
push-button switches to select the sample
frequency, bit depth and status bit modes.
FREQUENCY allows selection of the master
sample frequency from one of 32kHz,
44.1kHz, 48kHz, 88.2kHz, 96kHz, 176.4kHz or
192kHz. BITS sets the output bit depth as one
of 16, 20 or 24 bits, and CS DATA defines the
content of the channel status bits embedded
within the digital audio stream. The status
bits can be forced to Professional Mode
(PRO), Consumer Mode (CON) or to follow
the mode of the input (FOLLOW).
The SYNC button is used to select the
synchronisation input, from Word Clock,
AES/EBU or the D/A input, and also the
synchronisation mode of the digital output.
The A/D section of the RB-ADDA2 operates in
four selectable modes:
Master Mode - In this mode the unit
receives an analogue audio signal, which
is digitised and formatted for digital serial
transmission (IEC958). The necessary clock
signals are generated internally from an
on board master clock at a selectable rate
(32kHz, 44.1kHz, 48kHz, 88.2kHz, 96kHz,
176.4kHz or 192kHz).
AES/EBU, S/PDIF or TOSlink optical and which
also sets the analogue output level to be
generated for full scale digits, from either
+12dBFS, +18dBFS or +24dBFS. There are
factory-set internal level controls for fine
tuning the analogue output gain adjustment.
If no digital audio source is present, the D/A
SOURCE button flashes. In both A/D and D/A
sections, audio is sent to all of the outputs
simultaneously. The RB-ADDA2 automatically
decodes 50/15μs emphasis if this is indicated
by certain channel status bits in the incoming
digital audio data. A red LED indicates when
power to the RB-ADDA2 is on.
Analogue
Input Level
for FSD:
+12dBFS, +18dBFS or +24dBFS, via A/D
SOURCE push-button
Analogue
Input Level
Adjust:
+9dBu to +27dBu via rear-panel
pre-set pots
Sample
Frequency
Rates:
32kHz, 44.1kHz, 48kHz, 88.2kHz, 96kHz,
176.4kHz or 192kHz, via FREQUENCY
push-button
Maximum
Output Level:
+24dBu balanced output, +14dBu
unbalanced output
Specification For RB-ADDA2
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Impedance:
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Auto Mode - Here the unit is synchronised
to an external source, using the digital audio
sync or D/A input signal from which the
clock signals are stripped, or to the TTL level
Word Clock. If no sync signal is present the
unit runs from the onboard master clock at
a rate selected by the front panel control
(32kHz, 44.1kHz, 48kHz, 88.2kHz, 96kHz,
176.4kHz or 192kHz).
Audio Specification
Dynamic Range:
>110dB
Analogue to Digital Conversion
A/D Audio Specification For RB-ADDA2
D/A Connections
Auto Lock Mode - This operates like the
auto mode except that if no sync signal is
present the unit runs at the closest master
clock rate to the last locked incoming
signal. The FREQUENCY LED indicates the
synchronised sample frequency.
A/D Connections
Serial RS232:
1 x 9 pin D-type plug
When operating in sync modes, the SYNC
button flashes whenever the unit is not
synchronised to the incoming digital signal.
Mains Input:
Filtered IEC, continuously rated
85-264VAC @ 47-63Hz, 10W max
Fuse Rating:
Anti-surge fuse 1A 20 x 5mm
* Note that this product is deeper than standard Redboxes.
Slave Mode - Here the unit is synchronised
to an external source, using the digital audio
sync or D/A input signal from which the
clock signals are stripped, or to the TTL level
Word Clock. The FREQUENCY LED indicates
the synchronised sample frequency and if no
sync is present, no output is generated.
The D/A section has a SOURCE push-button
which selects the digital input source from
Bit Depth:
16, 20 or 24 bits, via BITS push-button
Channel
Status Bits:
Consumer mode, professional mode or
follow input, via CS DATA push-button
Sync Input
Select:
AES/EBU, Word Clock or D/A input, via
SYNC push-button
Sync Mode
Select:
Master, slave, auto, auto lock, via SYNC
push-button
Digital to Analogue Conversion
D/A Audio Specification For RB-ADDA2
Digital Inputs:
1 x AES/EBU XLR 3 pin female
1 x S/PDIF RCA phono
1 x TOSlink optical
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balanced
Analogue
Outputs:
2 x XLR 3 pin male (balanced)
2 x RCA phono (unbalanced)
Dynamic Range:
>110dB
D/A Operational Controls
Gain Range:
Adjustable input gain of ±3dB on selected
+12dBu, +18dBu or +24dBu, ref FSD
Digital
Input Select:
AES/EBU, S/PDIF or TOSlink optical, via
push-button
Distortion
& Noise:
>96dB THD + N at 1kHz
Analogue
Output Level
for FSD:
Selectable +12dBu, +18dBu or +24dBu
output level, ref FSD, via D/A SOURCE
push-button
Analogue
Inputs:
2 x XLR 3 pin (balanced)
2 x RCA phono (unbalanced)
Equipment Type
Sync Inputs:
1 x AES/EBU XLR 3 pin female
1 x TTL Word clock BNC
Digital Outputs:
1 x AES/EBU XLR 3 pin male
1 x S/PDIF RCA phono
1 x TOSlink optical
Maximum
Input Level:
+27dBu balanced inputs, +17dBu
unbalanced inputs
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Impedance:
RB-ADDA2:
Combined A/D and D/A converter,
24 bit 192kHz
Physical Specification
Dimensions
(Raw):
48cm (W) x 15.8cm (D*) x 4.2cm (H) (1U)
19” (W) x 6.2” (D*) x 1.7” (H) (1U)
(Boxed):
59cm (W) x 27.5cm (D*) x 11cm (H)
23.2” (W) x 10.8” (D*) x 4.3” (H)
Weight:
Nett: 1.6kg Gross: 2.3kg
Nett: 3.5lbs Gross: 5lbs
A/D Operational Controls
Analogue
Input Source:
Balanced XLR or unbalanced phono, via
A/D SOURCE push-button
55
Audio & Video Interfaces - Digital Audio Converters
RB-SC1 Sample Rate Converter
Category: Digital Audio Converters.
Description: Sample Rate Converter
(24 bit, 96kHz Capable).
Product Function: Transferring digital
audio between digital equipment.
Typical Applications: Converting an
audio library from 44.1kHz to 48kHz,
transferring digital audio from a recorder
at 48kHz to a digital mixer at 96kHz.
56
Ideal for the transfer of digital audio
between different digital equipment,
the RB-SC1 sample rate converter
standardises the sample rate of a digital
audio signal to one of 32kHz, 44.1kHz,
48kHz or 96kHz, or to a synchronising
input.
Master Mode - In this mode the digital
output sample rate is simply set by, and locked
to, the internal on-board clock generator. No
sync signal is used or required.
Both audio inputs and outputs have pushbutton switches to select either AES/EBU
or S/PDIF. The synchronising input can be
selected from one of AES/EBU, S/PDIF or
TTL word clock.
There are also switches available to define
the content of the channel status bits
embedded within the digital audio stream.
The channel status bits will be forced to
Auto Lock Mode - In this mode no output
Professional Mode for sample rates above
will be generated until lock is achieved with
48kHz as they are not supported by the
a sync signal. The digital output sample rate
now follows the sync input. If the sync signal is Consumer Mode. For sample rates of 32kHz,
removed then the output sample rate will be 44.1kHz and 48kHz, the status bits can be
There are four modes of operation of the
RB-SC1 dependent on how you want to
synchronise the output to the input :
Auto Sync Mode - In this mode the digital
output sample rate follows the sync input.
When the sync signal is not present the
output sample rate will be set by, and locked
to, the internal on-board clock generator at a
frequency determined by the switch position.
set by, and locked to, the internal on-board
clock generator at the closest frequency
available to the previous sync input.
Slave Mode - In this mode the digital
output sample rate follows the sync input.
When the sync signal is not present the
digital output is turned off.
Audio & Video Interfaces - Digital Audio Converters
either set to follow the input signal type,
or can be forced to either Professional or
Consumer Mode.
Specification For RB-SC1
Dynamic Range:
120dB
As well as indicating that power is present
on the unit, the LED on the front panel has
a secondary role to indicate the status of
the digital inputs. Fast flashing between
red and amber indicates a loss of a digital
input signal and slow flashing between
red and amber indicates the absence of a
synchronising input when not in Master
Mode.
Distortion
and Noise:
Audio Specification
Output Select:
Push-button switch between AES/EBU
and S/PDIF
-114dB THD + N at 1kHz, ref 0dB FS
Sync Select:
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Push-button switch between AES/EBU
and S/PDIF, with DIP switch selection
between TTL and the other two inputs.
Operational
Modes:
Sample
Frequency
Range:
Master rates of 32kHz, 44.1kHz, 48kHz
or 96kHz, or can synchronise to
incoming 32kHz to 100kHz sample rate
Master mode, auto sync mode, auto
lock mode and slave mode, set via
rotary switch
Status Bits:
Bit Depth:
Up to and including 24 bit
Forced to consumer mode, professional
mode, or set to follow input with DIP
switch selection.
Mains Input:
Filtered IEC, continuously rated
85-264VAC @ 47-63Hz, max 10W
Anti-surge fuse 100mA 20 x 5mm
(230VAC)
Anti-surge fuse 250mA 20 x 5mm
(115VAC)
Rear Panel Connections and Controls
Inputs:
2 x AES/EBU XLR 3 pin female (audio
and sync)
2 x S/PDIF RCA phono (audio and sync)
1 x TTL BNC female (sync)
Fuse Rating:
Outputs:
1 x AES/EBU XLR 3 pin male
1 x S/PDIF RCA phono
Equipment Type
Input Select:
Push-button switch between AES/EBU
and S/PDIF
RB-SC1:
Physical Specification
Dimensions
(Raw):
28cm (W) x 10.8cm (D) x 4.2cm (H) (1U)
11” (W) x 4.3” (D) x 1.7” (H) (1U)
Dimensions
36cm (W) x 20.5cm (D) x 6cm (H)
(Boxed):
14.2” (W) x 8” (D) x 2.4” (H)
Weight:
Nett: 1.0kg Gross: 1.4kg
Nett: 2.2lbs Gross: 3.1lbs
Sample rate converter
RB-SC1 Sample Rate Converter Flowchart.
57
Audio & Video Interfaces - Digital Audio Converters
RB-SC2 Dual Sample Rate Converter
Category: Digital Audio Converters.
Description: Dual Sample Rate Converter
(24 bit, 192kHz Capable).
Product Function: Transferring digital
audio between digital equipment.
Typical Applications: Converting an
audio library from 44.1kHz to 48kHz,
transferring digital audio from a recorder
at 48kHz to a digital mixer at 192kHz.
58
The RB-SC2 sample rate converter is a 1U
rack-mount which produces AES/EBU,
S/PDIF and TOSlink optical level digital
audio outputs from a balanced AES/EBU,
S/PDIF and TOSlink optical level digital
audio inputs. The sample rate of the
outputs can be set by an internal clock or
from various external synchroniser sources.
The RB-SC2 is a high performance, enhanced
version of the RB-SC1 providing the
following additional features:
• It supports higher sample frequency rates
up to and including 176.4kHz and 192kHz.
• It has 2 independent sample rate
converter circuits that use a common
clock source to set the output sample
rate.
• It has 2 optional video synchronising
boards. These set the output sample rate
to 48kHz that is synchronised to either an
analogue video signal or SDI digital video
signal (HD or SD).
• A special X-Lock mode allows the unit to
function as a full bi-directional sample
rate converter.
• It has TOSlink optical digital audio inputs
and outputs.
• It has front panel push-button switches
for all the main settings. The buttons
are arranged in sets, where pressing the
button advances the current selection
and LED indicator.
• A serial RS232 port is included so that
the RB-SC2 settings can be controlled
remotely. The front panel LED indicators
alter automatically when using RS232
commands.
For the digital outputs, there are three
push-button switches to select the sample
frequency (FREQUENCY), channel status
bit type (CSDATA), and sync source and
mode of operation (SYNC).
The FREQUENCY button allows selection
of the master sample frequency from
one of 32kHz, 44.1kHz, 48kHz, 88.2kHz,
96kHz, 176.4kHz or 192kHz. The CS DATA
button defines the content of the channel
status bits embedded within the digital
audio stream, and can be forced to either
Professional Mode (PRO), Consumer Mode
(CON) or to follow the mode of the input
(FOLLOW).
The SYNC button is used to select the
synchronisation input, from the AES/EBU
sync input, the Wordclock input or, for
X-Lock, the other digital input. The X-Lock
synchronisation allows the unit to act as a
bi-directional sample rate converter with
the output of sample rate converter 1
syncing the input of sample rate converter
2 and vice versa so that they follow each
other.
The application for the X-Lock mode is so
that the RB-SC2 can be inserted between
2 digital devices which run at different
sample rates, such as a PC recorder and a
digital player. Using the RB-SC2 in X-Lock
mode ensures that the 2 devices remain
synchronised at all times regardless of the
sample rate of the 2 devices.
The SYNC button will also select the
operating mode of the unit as described
below. If an optional video sync board is
fitted then 2 sync LEDs light together to
show the active video sync.
Master Mode - In this mode the unit
receives a digital audio signal, which
is passed to the sample rate converter
and then re-formatted for the digital
serial transmitter (IEC958). The sample
rate converter clock signal is generated
internally from an on board master clock at
a selectable rate (32kHz, 44.1kHz, 48kHz,
88.2kHz, 96kHz, 176.4kHz or 192kHz).
Slave Mode - In this mode the unit is
synchronised to an external source, using
the digital audio sync, or to the TTL level
Word Clock. The FREQUENCY LED will
indicate the synchronised sample frequency
and if no sync is present, no output will be
generated.
Auto Mode - Here, the unit is synchronised
to an external source, using the digital
Audio & Video Interfaces - Digital Audio Converters
192
RB-SC2 shown with optional synchronisation board.
audio sync, or to the TTL level Word Clock.
If no sync signal is present the unit runs
from the onboard master clock at a rate
selected by the front panel control (32kHz,
44.1kHz, 48kHz, 88.2kHz, 96kHz, 176.4kHz
or 192kHz).
Specification For RB-SC2
Auto Lock Mode - This operates like the
auto mode except that if no sync signal
is present the unit will run at the closest
master clock rate to the last locked incoming
signal. The FREQUENCY LED will indicate the
synchronised sample frequency.
Dynamic Range:
When operating in sync modes, the SYNC
button flashes whenever the unit is not
synchronised to the incoming sync signal.
Connections
There are 2 further push-button switches
(INPUT 1 & INPUT2) that are used to select
the input connector used for each of the
2 sample rate converter circuits. These
switches select between AES/EBU, S/PDIF
and TOSLink optical connectors.
A red LED indicates power to the RB-SC2.
will accept a composite signal
of NTSC (525), PAL (625) & SECAM
(625) signals covered by
SMPTE-170-M (NTSC) and ITU-R
BT.470-6 (PAL & SECAM).
Audio Specification
138dB typical A-Weighted.
Distortion & Noise: -134dB THD + N at 1kHz, ref 0 dBFS
Sample Freq
Range:
32kHz – 196kHz
Input Sample
Width:
Up to and including 24 Bits.
Output Sample
Width:
24 Bits.
Digital Inputs:
2 x AES/EBU XLR 3 pin female
2 x S/PDIF RCA phono
2 x TOSLink optical input
Digital Outputs:
2 x AES/EBU XLR 3 pin plug
2 x S/PDIF RCA phono socket
2 x TOSLink optical output
Sync Inputs:
Video Sync
Specs:
1 x AES/EBU XLR 3 pin female
1 x Word Clock BNC
1 x Video Input (optional)
The Digital video sync board will
accept 270Mbps SD-SDI and
HD-SDI signals covered by
SMPTE-259-M-C (SD) and
SMPTE-292M (HD).
The Analogue video sync board
Equipment Type
RB-SC2:
Dual Stereo Sample Rate Converter,
24 bit 192kHz
Physical Specifications
Dimensions
(Raw):
48cm (W) x 10.8cm (D*) x 4.2cm (H) (1U)
19” (W) x 4.3” (D*) x 1.7” (H) (1U)
Dimensions
(Boxed):
59cm (W) x 27.5cm (D) x 11cm (H)
23.2” (W) x 10.8” (D) x 4.3” (H)
Consumer mode, professional
mode or follow input, via CS
DATA push-button
Weight:
Nett: 1.4kg Gross: 2.0kg
Nett: 3.1lb Gross: 4.4lb
Digital Select:
AES/EBU, S/PDIF or TOSlink Input
optical, via INPUT1 or INPUT2
push-buttons
RB-SYA:
Analogue video sync board
(NTSC, PAL & SECAM)
Sync Input
Select:
AES/EBU, Word Clock, X-Lock or
Video, via SYNC push-button
RB-SYD:
Digital video sync board (SD-SDI
Sync Mode
Select:
Master, slave, auto or auto lock,
via SYNC push-button
RB-RK3:
1U Rear panel rack kit for large
Redboxes
Operational Controls
Master
Frequency
Select:
32, 44.1, 48, 88.2, 96,176.4 or 192kHz
via FREQUENCY push-button
Channel
Status Bits:
Other Connections
Mains Input:
Universal filtered IEC,
continuously rated 85-264VAC
@47- 63Hz, max 10W
Fuse Rating:
Anti-surge fuse 1A 20 x 5mm
Serial Port:
RS232 9 Pin D-type socket
Accessories
& HD-SDI)
Note that this product is deeper than standard Redboxes
59
Audio & Video Interfaces - Digital Audio Converters
RB-DAC1 Digital To Analogue Converter
Using 24 bit, 96kHz capable devices, the
RB-DAC1 D/A Converter is a 1U rackmount which produces a stereo balanced
XLR or unbalanced phono output from
an incoming AES/EBU or S/PDIF digital
input signal. There is also a headphone
output with volume control for monitoring
purposes.
The analogue outputs have an output
level control, allowing full-scale settings
selectable from +12dBu, +18dBu or +24dBu.
The RCA phono outputs have a further
8.5dBu attenuation incorporated.
60
There is a button to select either the AES/
EBU or S/PDIF input for the D/A converter,
which is located on the rear panel.
Additionally, if de-emphasis is selected, the
RB-DAC1 will decode 50/15μs emphasis
when indicated by certain channel status
bits in the incoming digital audio data.
When operating, the front panel power
LED flashes whenever the unit is not
synchronised to the incoming digital signal.
Category: Digital Audio Converters.
Product Function: Converts a digital
audio signal to an analogue signal.
Typical Applications: To connect the
output of a CD player to an analogue
amplifier, or the program output of
a digital mixer to a pair of powered
speakers.
Features: AES/EBU & S/PDIF digital
inputs, headphone amplifier, configurable
full-scale settings.
Specification For RB-DAC1
Gain Select:
DIP switch
Audio Specification
De-emphasis
On/Off:
DIP switch
Maximum
Output Level:
+24dBu balanced, +14dBu unbalanced
+12dBu headphone
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Digital Inputs:
1 x AES/EBU XLR 3 pin female
1 x S/PDIF RCA phono
Dynamic Range:
>100dB
Gain Range:
Selectable 12dBu, 18dBu or 24dBu
output level, ref FSD
Analogue
Outputs:
2 x XLR 3 pin male (balanced)
2 x RCA phono (unbalanced)
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Headphone
Output:
1 x ¼” (6.35mm) A/B-gauge
3-pole stereo jack sockets
Mains Input:
Distortion
& Noise:
<0.01% THD + N @1kHz, ref 0dBu
Filtered IEC, 110V-120V, or 220-240V,
fused, 10W max
Fuse Rating:
Sample Freq.
Range:
30kHz - 100kHz
Anti-surge fuse 100mA 20 x 5mm
(230VAC)
Anti-surge fuse 250mA 20 x 5mm
(115VAC)
Connections
Operational Controls
Equipment Type
Digital Input
Select:
RB-DAC1:
AES/EBU or S/PDIF, via push-button
Digital to analogue converter
Physical Specification
Dimensions
(Raw):
48cm (W) x 10.8cm (D) x 4.2cm (H) (1U)
19” (W) x 4.3” (D) x 1.7” (H) (1U)
Dimensions
(Boxed):
58.5cm (W) x 22.5cm (D) x 7cm (H)
23” (W) x 8.9” (D) x 2.8” (H)
Weight
Nett: 1.4kg Gross: 2.0kg
Nett: 3.1lbs Gross: 4.4lbs
Audio & Video Interfaces - Digital Audio Converters
RB-SP1 Digital Splitter and Combiner
The RB-SP1 can be used for interfacing
stereo and mono signals to digital mixing
desks by splitting the left and right signals of
a stereo XLR to two separate XLR’s, and vice
versa by combining them.
The RB-SP1 can interface between them,
either combining the signals from 2 XLR’s into
1, or splitting the signal from 1 XLR into 2.
The RB-SP1 (digital splitter & combiner) is
also used to interface legacy double sampling
pieces of equipment. Some older equipment
uses 2 x AES/EBU connectors for double
sampling with each connector carrying an
audio signal at a normal frame rate, whilst
other equipment has a single connector
using twice the frame rate.
Additionally, a sample rate converter on the
second digital input can be used to convert
the sample rate of the secondary input to
that of the primary input. The RB-SP1 can
handle sample rates up to 96kHz and sample
sizes of 16, 20 and 24 bit.
Category: Digital Audio Converters.
Product Function: Splits a stereo digital
handbook for further information on what
this product can do.
Both inputs and outputs can be selected as
either AES/EBU or S/PDIF with the resultant
digital level following the switch selection.
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Sample
Freq Range:
30-100kHz (i.e. including 32kHz,
44.1kHz, 48kHz, 64kHz, 88.2kHz
and 96kHz)
Dynamic Range:
120dB
Connections
Audio Inputs:
2 x AES/EBU XLR 3 pin female
(balanced)
2 x S/PDIF RCA phono female
(unbalanced) (Input button selects
between AES/EBU and S/PDIF)
Audio Outputs:
2 x AES/EBU XLR 3 pin male (balanced)
2 x S/PDIF RCA phono female
(unbalanced), (Output button selects
between AES/EBU and S/PDIF)
Mains Input:
Filtered IEC, continuously rated
85-264VAC @ 47-63Hz, max 10W
Fuse Rating:
Anti-surge fuse 1A 20 x 5mm
Specification For RB-SP1
Audio Specification
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There are two types of operation : Split 96,
and Stereo/Mono. These each have three
different switch modes : Split, Bypass
and Combine. Download the product
audio signal to mono and combines two
mono signals to make a stereo signal.
Typical Applications: To combine two
mono digital channels, e.g. telco, to a
stereo input for a mixer.
Features: 2 x AES/EBU & S/PDIF I/O, up to
96kHz operation.
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Signal Level:
3V/10V peak to peak min/max
(AES/EBU)
0.5V ±20% peak to peak (S/PDIF)
Bit Depth:
Up to and including 24 bit
Distortion
& Noise:
-114dB THD + N at 1kHz, ref 0dB FS
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Equipment Type
RB-SP1:
Digital splitter & combiner
Physical Specification
Dimensions
(Raw):
48cm (W) x 10.8cm (D) x 4.2cm (H) (1U)
19” (W) x 4.3” (D) x 1.7” (H) (1U)
Dimensions
(Boxed):
58.5cm (W) x 22.5cm (D) x 7cm (H)
23” (W) x 8.9” (D) x 2.8” (H)
Weight
Nett: 1.6kg Gross: 2.2kg
Nett: 3.5lbs Gross: 4.8lbs
61
Audio & Video Interfaces - Digital Audio Converters Add-On Boards
Synchronisation Add-On Boards
There are four optional synchronisation boards which can be used with the RB-SC2 and RB-TGHD(B & X) to synchronise the outputs to analogue video (RB-SYA), digital video (RB-SYD),
AES/EBU audio (RB-SYE) and word clock (RB-SYW) respectively.
RB-SYA Analogue Video Sync Board
The Analogue video sync board will accept
a composite signal of NTSC (525), PAL (625)
& SECAM (625) signals covered by SMPTE170-M (NTSC) and ITU-R BT.470-6 (PAL &
SECAM).
RB-SYD Digital Video Sync Board
The Digital video sync board will accept
270Mbps SD-SDI and HD-SDI signals covered
by SMPTE-259-M-C (SD) and SMPTE-292M
(HD).
RB-SYE Audio Sync Board
The AES/EBU sync board can be used to
synchronise the outputs of the RB-TGHD(B &
X) and can accept a digital audio input signal
with a sample frequency between 32kHz
and 192 kHz. When using the RB-SYE sync
board, the Channel Status information that is
encoded in the input data signal is copied to
all digital output channels on the RB-TGHD.
RB-SYW Word Clock Sync Board
The Word Clock sync board will accept a
distributed clock running at the desired
sample frequency between 32 kHz and 192
kHz. The signal can be differential or single
ended TTL level.
RB-SYA Analogue Video Sync Board For RB-SC2
(PAL, NTSC, SECAM).
RB-SYD Digital Video Sync Board For RB-SC2 (HDSDI, SD-SDI).
RB-SYE AES/EBU Sync Board.
RB-SYW Word Clock Sync Board For RB-TGHD.
Category: Synchronisation Add-On Boards.
62
Product Function: Allows the RB-SC2 and
RB-TGHD(B & X) to be synchronised to
various inputs.
Typical Applications: To sync the products
to an analogue video sync input, or AES/
EBU, or Word Clock.
Audio & Video Interfaces - Synchronisers, Delays & Silence Detectors
RB-DS2R Remote Switch Panel For RB-DS2
Category: Synchronisers, Delays &
Silence Detectors.
Product Function: Remote control unit
for up to 4 x RB-DS2 units.
Typical Applications: Remote control
of multiple RB-DS2 units from another
location, where quick delay time changes
are needed.
Features: Quick and simple control of
delay times in frames from 0 to 7
The RB-DS2R is a remote panel for
controlling up to 4 separate RB-DS2 units
from a single 1U 19” rack-mount panel.
Specification For RB-DS2R
Audio Specification
Rear Panel Connections
Remote I/O Port: 4 x 15-way ‘D’-type socket, 8 GPI outputs
The unit has 4 rotary switches, each with 8
available selections numbered 0 to 7, used
for selecting the delay time in frames.
Front Panel Controls
On the rear of the unit are 4 x 15-way
D-type connectors used for connecting
directly to the remote input of 4 x RB-DS2
units.
Physical Specification
The RB-DS2R is a passive unit, i.e. there is no
power supply.
Selector Switches: 4 x 8 way rotary selectors
Equipment Type
RB-DS2R:
Remote switch panel for 4 x RB-DS2
Dimensions
(Raw):
48cm (W) x 10.8cm (D) x 4.2cm (H) (1U)
19” (W) x 4.3” (D) x 1.7” (H) (1U)
Dimensions
(Boxed):
58.5cm (W) x 22.5cm (D) x 7cm (H)
23” (W) x 8.9” (D) x 2.8” (H)Weight
Weight:
Nett: 1.2kg Gross: 1.6kg
Nett: 2.6lbs Gross: 3.5lbs
63
Audio & Video Interfaces - Synchronisers, Delays & Silence Detectors
RB-DS2 Stereo Delay Synchroniser & Time-Zone Delay
Category: Synchronisers, Delays &
Silence Detectors.
Product Function: Resynchronisation
of audio to video (lip-sync) following
conversion, transmission delay &
network delays.
64
Typical Applications: In the broadcast
chain to remove lip-sync errors, post
production to synchronize audio when
monitoring video signals, time zone delay
to provide +1 hour programme feed.
Features: Analogue and digital I/O. Can
be used as a fixed delay or for correction
on the fly. Memory expansion option
offers delay times in excess of 4 hours.
8 GPI’s offer remote access/switching of
pre-set delays.
The RB-DS2 is a stereo audio delay
synchroniser used for resynchronising
audio to video following delay processes
such as standards conversion, transmission
delay, logo insertion, video aspect ratio
conversion and network delays.
It can be used for fixed installations to
correct a permanent audio delay, or on an
intermittent basis to provide occasional
correction, for example for live links.
Accepting digital audio signals up to 96kHz,
24 bit, the sonic quality of the RB-DS2
is superb and silent switching is used to
provide the smoothest, cleanest audio delay
available.
Audio & Video Interfaces - Synchronisers, Delays & Silence Detectors
The RB-DS2 has both balanced analogue and
AES/EBU digital audio inputs and outputs
on 3 pin XLR connectors. It can act as a
combined A/D and D/A unit meaning that
analogue inputs can be delayed and output
as AES/EBU or vice-versa. It is a stereo delay,
but can also be used as a dual mono delay,
to process each audio path separately, or as
a mono delay only.
As standard the RB-DS2 can provide up to
10.5 seconds of delay at 96kHz sampling,
24 bit (42 secs at 48kHz, 16 bit). An internal
Compact Flash™ expansion allows up to 2GB
of memory to be accessed providing delay
times of over 4 hours, for example, to delay
a programme output across different timezones, or to shift a broadcast programme
by 1 hour for a satellite rebroadcast. Delay
times can be selected in samples, fields,
frames, metres, milliseconds and with the
Compact Flash™ expansion, in hh:mm:ss.
Frame and field definitions can be for PAL
(25 frame) or NTSC (30 frame) signals.
A front panel blue vacuum fluorescent
display with rotary controller is used for
selecting the various settings of the delay,
which include the source (analogue or
digital), channels, sample rate, sample bit
width, format (PAL or NTSC), delay units
and the delay itself. Additionally, input peak
digits can be selected from +12dBu, +18dBu
and +24dBu for FSD and two left and right
pre-set potentiometers on the rear panel
allow the input gain range to be altered by
±3dB around the selected peak digits.
Specification For RB-DS2
Audio Specification
Rear Panel Connections
A/D Specification
Analogue
Inputs:
2 x XLR 3 pin female (balanced) (L & R)
Analogue
Outputs:
2 x XLR 3 pin male (balanced) (L & R)
Digital Inputs:
1 x AES/EBU XLR 3 pin female
Digital Outputs:
1 x AES/EBU XLR 3 pin male
Maximum
Input Level:
+28dBu
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Analogue &
Digital Input
Levels:
Selectable +12dBu, +18dBu, +24dBu
for FSD
Analogue
Pre-set Input
Gain Range:
Adjustable 3dB loss to 3dB gain
(L & R adjust)
Both analogue and digital outputs can be
separately muted and a front panel Bypass
button disengages electro-mechanical relays
to divert both analogue and digital inputs
to their outputs. This is also disengaged
automatically when a power-fail occurs.
Signal to Noise:
Better than -101dBFS (RMS A-weighted
at 24bit)
Dynamic Range:
> 110dB
Distortion
& Noise:
All of the settings in the unit can be saved
to one of 8 configuration settings. These
Configs can be viewed, edited, saved and
loaded, and also remotely loaded by using
one of the 8 GPI contacts, meaning that any
setting, such as delay time or Bypass, can be
altered instantaneously using a GPI signal.
The analogue output gain range can be
altered in software from -6dBu to +24dBu
output level, ref FSD.
The RB-DS2 also has an RS232 serial port
for remotely controlling the unit and there
are 4 remote outputs which can be used
for signalling. The front panel controls can
be locked-out for situations where remote
control is being used to run the unit, or
where physical security is required.
Remote I/O Port: 15-way ‘D’-type plug, 8 GPI inputs,
4 GPI outputs
Serial Comms
Port:
9-way ‘D’-type plug
Memory
Expansion:
Internal Compact Flash™ storage card
supporting up to 2GB. CF cards must be
PIO Type 4 or higher.
> 96dB THD + N at 1kHz
Mains Input:
Filtered IEC, continuously rated 85-264
VAC, 47-63Hz, fused, 60W peak, 30W
average
Maximum
Output Level:
+24dBu
Fuse Rating:
Anti-surge fuse 2A 20 x 5mm
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Impedance:
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Display:
Vacuum fluorescent display
Dynamic Range:
> 100dB
System
Navigation:
Rotary selector with integral
push-switch
Analogue
Output Gain
Range:
Selectable -6dBu to +24dBu output
level, ref FSD
Audio Bypass:
Via push-button
Sampling
Frequency:
Selectable 32kHz, 44.1kHz, 48kHz,
64kHz, 88.2kHz or 96kHz
D/A Specification
Front Panel Controls
Sample Width:
Selectable 16bit or 24bit
Channels:
Stereo or dual mono
Format (Fields
& Frames):
PAL, NTSC
Delay Units:
Samples, fields, frames, milliseconds,
hh:mm:ss, (with CF expansion)
Maximum
Delay:
10.5 seconds @ 96kHz, 24bit, stereo
(excluding memory expansion)
Equipment Type
RB-DS2:
Stereo delay synchroniser & time-zone
delay
Physical Specification
Dimensions
(Raw):
48cm (W) x 15.8cm (D *) x 4.2cm (H)
(1U)
19” (W) x 6.2” (D*) x 1.7” (H) (1U)
Dimensions
(Boxed):
59cm (W) x 27.5cm (D*) x 11cm (H)
23.2” (W) x 10.8” (D*) x 4.3” (H)Weight
Weight:
Nett: 1.6kg Gross: 2.3kg
Nett: 3.5lbs Gross: 4.8lbs
* Note that this product is deeper than standard Redboxes
65
Audio & Video Interfaces - Synchronisers, Delays & Silence Detectors
RB-PD2 Stereo Profanity Delay
Category: Synchronisers, Delays &
Silence Detectors.
Product Function: To introduce an audio
delay into a program output allowing
unsuitable or profain program material
to be removed instantly.
66
Typical Applications: Radio and TV phone
in programs and chat shows where content
needs to be censored.
Features: Both analogue and digital audio
I/O, automatic audio stretch algorithm
allows seemless building of the delay, no
pitch changes, up to 55 seconds of delay,
Drop and Dump buttons, Dump via jingle
playout or dropping of audio, GPI control
of buttons, GPIO can follow current delay
time.
The RB-PD2 is a stereo audio profanity
delay used for live broadcast programs to
prevent unwanted or obscene material
from being transmitted. It features an
automatic audio stretch algorithm that
allows between 2 and 55 (*) seconds of
delay to be built up live whilst “on air”,
whilst maintaining the correct pitch. The
delay can also be acquired whilst playing
a pre-selected audio file on a Compact
Flash™ memory card. When the program
is complete, the audio stretch algorithm
seamlessly reduces the delay to zero.
The RB-PD2 has both balanced analogue and
AES/EBU digital audio inputs and outputs on
3 pin XLR connectors and provides sample
rates up to 48kHz at 24 bit. It can act as a
combined A/D and D/A unit meaning that
analogue inputs can be output as AES/EBU
or vice-versa.
The delay can be initiated by pressing the
BUILD DELAY button on the front panel.
A front panel display shows the amount
of delay being built-up, up to the amount
initially selected.
Audio & Video Interfaces - Synchronisers, Delays & Silence Detectors
There are several ways to make sure that
any unwanted material is removed from the
audio at the outputs. A COUGH function,
activated from a dedicated front panel
button, allows locally generated sounds
being presented at the inputs, such as the
presenter coughing or equipment switching
noises, to be discarded.
The DUMP function, which is also activated
from a front panel button, has 2 different
modes. The first DUMP mode removes
a section of audio that has already been
buffered, by a pre-selected amount. The
second DUMP mode plays a pre-selected
audio file on the Compact Flash™ memory
card. When the file has finished playing, the
delay is then equal to the duration of the
file. The DUMP button can be used multiple
times to use up the built-up delay and once
used, the unit automatically starts to rebuild
the original delay time.
As a last resort, all the buffered audio
can be discarded by pressing and holding
the DUMP button which activates the DROP
function.
At the end of a radio show when you want
to broadcast live, the delay can be ramped
down by pressing the front panel EXIT DELAY
button.
A dedicated record mode allows audio
presented at either the analogue or digital
inputs to be recorded to a linear WAV
file on a Compact Flash™ memory card.
Additionally, the card format used is PC
readable, allowing pre-recorded linear WAV
files to be transferred easily from a PC or
other such device.
A front panel blue vacuum fluorescent
display with rotary controller is used
for selecting the various settings of the
profanity delay, which include the start
delay and dump modes, safe period, source
(analogue or digital), sample rate and
sample bit width as well as the required
delay time. The current delay value, in
seconds, is permanently displayed as is the
current status of the unit. Additionally, input
peak digits can be selected from +12dBu,
+18dBu and +24dBu for FSD and two left
and right pre-set potentiometers on the
rear panel allow the input gain range to be
altered by ±3dB around the selected peak
digits. The analogue output gain range
can be altered in software from -6dBu to
+24dBu output level, ref FSD. Both analogue
and digital outputs can be separately muted
and a front panel Bypass button disengages
electro-mechanical relays to divert both
analogue and digital inputs to their outputs.
This is also disengaged automatically when a
power-fail occurs.
The RB-PD2 features a remote port
supplying 8 inputs and 6 outputs, all of
which are freely assignable. The inputs can
be used to trigger any of the unit’s functions
such as build delay, activate cough or enter
record mode and start a new recording. The
outputs can provide external signalling to
indicate when certain events have occurred
such as the delay reaching the required
value or the outputs being muted. Also,
the 6 remote outputs can optionally be
made to follow 6 of the inputs with the
current programme delay inserted between
actuation on the input and actuation on the
output. This can be useful for timed events
that need to account for the delay built up
by the RB-PD2.
Because playback from a Compact Flash™
card can be triggered remotely, the RB-PD2
can also be used at transmitter sites to play
an emergency audio file via GPI in the event
of silence detection.
The front panel controls can be locked-out
for situations where remote control is being
used to run the unit, or where physical
security is required.
A red LED indicates when power to the
RB-PD2 is on.
Sampling Frequency:
Selectable 32kHz, 44.1kHz
or 48kHz
Sample Width:
Selectable 16bit or 24bit
Channels:
Stereo
Minimum Delay:
2 seconds
Maximum Delay:
Dependent on sample and
bit rates selected:
32kHz: 16 bit - 55 seconds
24 bit - 27.5 seconds
44.1kHz:16 bit - 40 seconds
24 bit - 20 seconds
48kHz: 16 bit - 37 seconds
24 bit - 18.5 seconds
Rear Panel Connections
Analogue Inputs:
2 x XLR 3 pin female (balanced)
(L & R)
Analogue Outputs:
2 x XLR 3 pin male (balanced)
(L & R)
Digital Inputs:
1 x AES/EBU XLR 3 pin female
Digital Outputs:
1 x AES/EBU XLR 3 pin male
Remote I/O Port:
15-way ‘D’-type plug, 8 GPI
inputs, 6 GPI outputs
Serial Comms Port:
9-way ‘D’-type plug
Mains Input:
Filtered IEC, continuously rated
85 - 264VAC, 47 - 63Hz, fused,
60W peak, 30W average
Fuse Rating:
Anti-surge fuse 2A 20 x 5mm
Front Panel Controls
* At 32kHz 16bit
Display:
Vacuum fluorescent display
Specification For RB-PD2
Direct Control
Push-Buttons:
Build Delay, Exit Delay, Cough,
Audio Bypass & Dump/Drop
A/D Specification
System Navigation:
Rotary selector with integral
push-switch
Removable Audio
Storage Device:
Compact Flash™ memory card port
(supporting up to 2GB) CF™ card
used must be PIO type 4 or higher
Maximum Input Level: +28dBu
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Analogue & Digital
Levels:
Selectable +12dBu, +18dBu, Input
+24dBu for FSD
Analogue Pre-set
Input Gain Range:
Adjustable 3dB loss to 3dB
gain (L & R adjust)
Signal to Noise:
Better than -101dBFS (RMS
A-weighted at 24bit)
Dynamic Range:
> 110dB
Distortion & Noise:
> 96dB THD + N at 1kHz
D/A Specification
Maximum Output
Level:
+24dBu
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Dynamic Range:
> 100dB
Analogue Output
Gain Range:
Selectable -6dBu to +24dBu
output level, ref FSD
Equipment Type
RB-PD2:
Stereo profanity delay
Physical Specification
Dimensions
(Raw):
48cm (W) x 15.8cm (D+) x 4.3cm (H)
(1U)
19” (W) x 6.2” (D+) x 1.7” (H)(1U)
Dimensions
(Boxed):
59cm (W) x 27.5cm (D*) x 11cm (H)
23.2” (W) x 10.8” (D*) x 4.3” (H)
Weight :
Nett: 1.7kg Gross: 2.3kg
Nett: 3.7lbs Gross: 5lbs
+
Note that this product is deeper than standard Redboxes
67
Audio & Video Interfaces - Synchronisers, Delays & Silence Detectors
RB-DD4 4 Channel Digital Audio Delay
Category: Synchronisers, Delays &
Silence Detectors.
Product Function: Resynchronisation
of audio to video (lip-sync) following
conversion, transmission delay &
network delays.
192
68
Typical Applications: In the broadcast
chain to remove lip-sync errors, post
production to synchronize audio when
monitoring video signals, time zone delay
to provide +1 hour programme feed.
Features: Digital I/O as AES/EBU, S/PDIF or
TOSlink; delays 4 channels independently,
front panel headphone monitoring, audio
presence display, remote operation with
SCi software, passive signal path and can
be used as a fixed delay or for correction
on the fly.
The RB-DD4 4 channel digital audio delay
allows you to delay 4 mono channels
of audio independently or together.
Each channel delay is user selectable
from multiples of common video frame
rates, or a user defined value set via the
serial interface. The unit is perfect for
synchronizing audio to video which has
been delayed by processing latency.
Using a front panel button, you can select
which channel needs to be delayed. There
is also an ‘ALL’ option which allows the
selected delay to be applied to all channels.
Then using another front panel button you
RB-DD4 4 Channel Digital Audio Delay (With RB-SYD Video Sync Board).
can select the length of one frame of delay
and the multiple of frames to delay by.
The connectivity is incredibly flexible,
allowing three different types of
connection to each input and output
including AES/EBU, S/PDIF and TOSLink.
All three different types of output can be
used simultaneously. There is a monitor
socket on the front panel which allows you
to listen to each mono channel, by front
panel selection. Pairs of channels can be
monitored (1 & 2 or 3 & 4) using a rear
panel stereo option. There is also an option
to attenuate the monitor by 12dB selectable
by rear panel DIPswitch. Audio presence is
detected and displayed for each channel
Audio & Video Interfaces - Synchronisers, Delays & Silence Detectors
around the INPUTS 1 & 2 and INPUTS 3 & 4
buttons.
The flexibility continues with many audio
synchronization options. The digital audio
output can be synchronized to either input,
an additional AES/EBU reference input, a
TTL wordclock BNC input or an analogue/
SDI video feed if used with an additional
RB-SYA or RB-SYD board. Also the output
can be synchronized to an on-board master
clock, with a selectable frame rate. There
are warning indicators on the front panel
for loss of lock on both inputs and for the
selected external synchronization. Selectable
synchronization modes are as follows:
Slave Mode - In this mode the digital
output sample rate follows the sync input.
When the sync signal is not present the
digital output is turned off.
A powerful feature of the RB-DD4 is that by
using the Sonifex SCi serial software, the
unit can be programmed for different delay
durations, levels and switching functions
so that you can program the unit for your
specific application. A rear panel DIPswitch
configures the unit to be controlled serially.
Contact Sonifex for further information if
you have a particular requirement that isn’t
catered for by the RB-DD4 as standard.
The RB-DD4 has been designed to have a
passive signal path through the main input,
Master Mode - In this mode the digital
so if power to the unit fails, signal inputs 1
output sample rate is simply set by, and
& 2 are routed to outputs 1 & 2 and signal
locked to, the internal on-board clock
generator. No sync signal is used or required. inputs 3 & 4 are routed to outputs 3 & 4.
This is essential for applications such as
Auto Sync Mode - In this mode the digital
installation at transmitter sites, where a
output sample rate follows the selected sync
power failure to the unit should not prevent
input. When the sync signal is not present the
the audio input signal from being output
output sample rate will be set by, and locked
to the transmitter. Please note that this
to, the internal on-board clock generator at
is not true for the TOSLink outputs which
the selected output frequency.
are muted.
Auto Lock Mode - The digital output
sample rate follows the sync input. If the sync
signal is removed then the output sample rate
will be set by, and locked to, the internal onboard clock generator at the closest frequency
available to the previous sync input.
Specification For RB-DD4
Sync Mode
Select:
Audio Specification
Master, Auto Sync, Auto Lock, Slave via
rear panel DIPSwitches
Stereo Features: Stereo monitor outputs via rear panel
DIPSwitches
Dynamic Range:
>138dB
Distortion
and Noise:
<-137dB THD + N at 1kHz, ref 0dB FS
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Signal Level:
Balanced: 3V/10V peak to peak min max
Unbalanced: Min 0.5V±20% peak to peak
Connections
Sample Freqs:
32, 44.1, 48, 88.2, 96,176.4 or 192kHz
Digital Inputs:
Bit Depth:
Up to and including 24 bit
Max Delay:
8 secs per mono channel at 32kHz
1.33 secs per mono channel at 192kHz
2 x AES/EBU XLR 3 pin female
2 x S/PDIF RCA phono
2 x TOSLink optical input
Digital Outputs:
Frame Rates
(Front Panel):
23.98fps, 24fps, 25fps, 29.97fps, 30fps,
50fps, 59.94fps, 60fps
2 x AES/EBU XLR 3 pin plug
2 x S/PDIF RCA phono socket
2 x TOSLink optical output
Frame Rates
(SCi Software):
525/29.97, 625/25, 720/60p,
720/59.94p, 720/50p, 720/30p,
720/29.97p, 720/25p, 720/24p,
720/23.98p, 1035/60i, 1035/59.94i,
1080/60i, 1080/59.94i, 1080/50i,
1080/30p, 1080/29.97p, 1080/25p,
1080/24p, 1080/23.98p, 1080/30pSF,
1080/29.97pSF, 1080/25pSF,
1080/24pSF, 1080/23.98pSF, 1080/60p,
1080/59.94p, 1080/50p
Sync Inputs:
1 x AES/EBU XLR 3 pin female
1 x Word Clock BNC
1 x Video Input (optional)
Delay Settable:
1 to 19 frames (front panel controls)
Frames, lines, fields, milliseconds and
samples up to total delay time (SCi
software)
Monitor
Attenuation:
12dB monitor attenuation via rear
panel DIPSwitches
Serial Mode:
Enter serial control mode via rear panel
DIPSwitches
Boot Mode:
Boot up base code or firmware via rear
panel DIPSwitches
Remote I/O Port: 15 way D-type plug
Serial Port:
RS232, 9 pin D-type socket
Mains Input:
Universal filtered IEC, continuously
rated [email protected] 63Hz, max 10W
Fuse Rating:
Anti-surge fuse 2A 20 x 5mm
Equipment Type
RB-DD4:
4 channel digital audio delay
Physical Specifications
Front Panel Operational Controls & Indicators
Dimensions
(Raw):
48cm (W) x 10.8cm (D*) x 4.2cm (H) (1U)
19” (W) x 4.3” (D*) x 1.7” (H) (1U)
Digital Input
Select:
AES/EBU, S/PDIF or TOSlink optical
via INPUTS 1 & 2 or INPUTS 3 & 4
push-buttons
Dimensions
(Boxed):
59cm (W) x 27.5cm (D*) x 11cm (H)
23.2” (W) x 10.8” (D*) x 4.3” (H)
Weight:
Delay Control:
Delay time selection system via front
panel push button
Nett: 1.4kg Gross: 2.0kg
Nett: 3.1lb Gross: 4.4lb
Accessories
Monitor
Select Control:
Headphone monitor channel select
RB-SYA:
Analogue video sync board
(NTSC, PAL & SECAM)
Indicators:
Input presence indicators via bicolour
LEDS around each push button
RB-SYD:
Digital video sync board
(SD-SDI & HD-SDI)
RB-RK3:
1U Rear panel rack kit for large
Redboxes
Rear Panel Operational Controls
Master Select:
32, 44.1, 48, 88.2, 96,176.4 or 192kHz
Frequency via rear panel DIPSwitches
Sync Source
Select:
INPUTS 1&2, INPUTS 3&4, AES Sync,
Word Clock, Video Sync via rear panel
DIPSwitches
* Note that this product is deeper than standard Redboxes
69
Audio & Video Interfaces - Synchronisers, Delays & Silence Detectors
RB-AEC Acoustic Echo Canceller
Category: Synchronisers, Delays &
Silence Detectors.
Product Function: Remove acoustic
echo in a presenter’s earpiece caused
by microphones picking up audio from
loudspeakers in delay.
Typical Applications: In a TV production
environment where presenters are fed a
signal which has some form of acoustic
delay or echo. Any situation where
adaptive echo cancellaton is required.
Features: Built-in web server interface,
analogue and digital I/O, automatic
operation once configured.
The RB-AEC 1U rack-mount is an acoustic
echo canceller primarily designed for the
benefit of studio personnel for television
and radio. When a studio presenter’s
microphone signal is played out through
a monitor speaker in the control room,
it can be picked up by the control room
microphone(s) and returned to the
presenter’s earpiece as an undesirable
echo.
In circumstances where green screen
video processing is taking place, the delay
can be greater than 200ms. Additionally,
the dimensions, occupancy and distance
between mouth and microphone can further
Before
Studio
Monitor/
Earpiece
Much like during a conference call
configuration between two rooms, each
room has a microphone and speaker to
conduct a conversation. When an occupant
of one room speaks, it takes a certain
length of time before it is received in the
After
Studio
Monitor/
Earpiece
Mic
Approx 200ms
Delay
Presenter
Receiving
Delayed
Audio
influence the echo. The RB-AEC is used to
remove the entire control room monitor
speaker output from the presenter’s feed
by adapting to the environment in which
the control room microphones are placed.
Although acoustic echo cancellation is
more commonly implemented in telephony
systems, the Sonifex RB-AEC is designed to
produce broadcast quality cancellation.
Presenter Receiving
Artifact Free Audio
RB-AEC
(C)
T(z)
Vol
Monitor
Control Room/
Gallery
70
Approx 200ms
Delay
(A)
(B)
TB
Mic
Mic
Vol
Monitor
TB
Mic
Speaker
Output
Control Room/
Gallery
Speaker
Output
second room. Without a suitable solution
this ‘delayed’ signal can then be captured
by the microphone in the second room and
returned back to the first room as an echo.
In the particular example of TV production,
as well as the processing/transmission
delay, sound reflections from the control
room monitor speaker into the control room
microphone(s) cause the studio earpiece to
suffer further delay. The sound reflections
in the control room vary with the contents
of the room including any personnel
present. Also, different frequencies produce
varying reflections across various types of
surfaces and magnitudes within the room.
For a 15m distance between speaker and
microphone the delay is as much as 40ms.
The DSP solution offered by the RB-AEC
can dynamically compensate for varying
configurations.
Operation of the RB-AEC
The post-processed transmission output
program from the studio (A) is sent to the
RB-AEC as an analogue or digital audio signal
(the stereo input is auto-sensing) which acts
as a mix-minus to the input signal (B) from
the Control Room. The RB-AEC removes the
unwanted acoustic echoes so that the audio
sent to the presenter’s earpiece (C) is free of
echoes and reflection artifacts.
Audio & Video Interfaces - Synchronisers, Delays & Silence Detectors
Specification For RB-AEC
Audio Specification
Audio Input
(Program):
1 x mono analogue or AES/ EBU
digital on XLR 3-pin female
(autoselecting)
Audio Input
(From Control
Room):
1 x mono analogue
or AES/EBU digital on XLR
3-pin female (autoselecting)
Max Level
(0dB Input Gain):
+18dBu (analogue)/0dBFS
(digital)
CMRR:
Maximum Output
Level:
+18dBu (analogue)/0dBFS (digital)
Controls
Configuration:
1 x Ethernet port and 1 x rear
panel 4-way dipswitch
Reset:
1 x front panel recessed button
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AES/EBU Output
Sample Rates:
Selectable 32kHz - 192kH
Distortion:
<0.02% (1kHz, +8dBu output)
RB-AEC:
Noise:
-84dB RMS, unity gain ref
+8dBu output
Physical Specification
>60dB typical
Frequency
Response:
20Hz-12kHz +0/-0.5dB
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Rejection Ratio
(Input to Output):
AES/EBU Input:
32kHz to 192kHz
Typically 20dB on complex
waveforms, reference peak level
of 0dB
Input Gain:
0, +6, +12 or +18dB digital
gain (switchable)
Remote I/O Port:
9 way D-Type socket
Ethernet Port:
1 x RJ45 with status LEDs
Audio Outputs
(Analogue):
2 x mono analogue on
XLR 3-pin male
Mains Input:
Filtered IEC, continuously rated 85
264VAC @ 47-63Hz, 10W max
Audio Outputs
(AES/EBU):
1 x stereo digital AES/EBU
on XLR 3-pin male
Fuse Rating:
Anti-surge fuse 1A 20 x 5mm
Equipment Type
Dimensions (Raw):
Acoustic echo canceller
48cm (W) x 10.8cm (D) x 4.2cm (H)
(1U)
19” (W) x 4.3” (D) x 1.7” (H) (1U)
Dimensions (Boxed): 58.5cm (W) x 22.5cm (D) x 7cm (H)
23” (W) x 8.9” (D) x 2.8” (H)
Weight:
Nett: 1.5kg Gross: 2.0kg
Nett: 3.3lbs Gross: 4.4lbs
71
Audio & Video Interfaces - Synchronisers, Delays & Silence Detectors
RB-SD1 Silence Detection Unit
Category: Synchronisers, Delays &
Silence Detectors.
Product Function: To detect silence in
an analogue audio signal and switch to a
second source.
Typical Applications: At a radio
transmitter site, before the input to
the transmitter. A local audio source
will become the secondary input in
case silence is detected from the studio
source. At a radio studio in case of
failure of broadcast playout server, it can
switch in an emergency playback system.
Features: Adjustable silence detect
threshold level and silence duration,
passive signal path, switching of external
equipment on silence detection,
automatic or manual operation, mono
or stereo operation, front panel LED
indications of alarm status.
72
The RB-SD1 silence detection unit is a 1U
rack-mount device used to monitor an
unattended stereo studio feed and in the
event of the signal going “quiet” after a
given period the unit will switch through an
alternative stereo audio signal. This signal
could be a recorded message (e.g. “normal
service will be resumed”, etc), a feed from
a CD player or minidisc machine, or an
alternative recorded programme. Controls
are provided to start external equipment
and to provide remote status indication.
It has 2 balanced stereo audio inputs with
the maximum input level being +28dBu.
Each input is user-defined as either the main
source or auxiliary source and both sources
are monitored for failure, each having a
remote failure alarm. In the event of the
main source dropping below a pre-set level
for a pre-determined amount of time, the
unit will automatically switch through to the
auxiliary signal.
The unit can operate in 2 modes - automatic
or manual. In both modes it will automatically
switch over to the auxiliary source on
detecting silence. When the main signal
is again detected it will either return to
the main signal automatically or manually
depending on the mode chosen. The RB-SD1
has a number of remote operational features.
Remote outputs provide separate relay
contact closures for failure of the main and
auxiliary inputs. You can also control remotely
all of the front panel switches for source
selection, mode selection and signal Restore.
You can remotely start and stop another
piece of equipment on alarm failure and
main signal Restore respectively. Also, the
longest silence time (2min 5sec) can be set
remotely, which is useful if you are expecting
to broadcast a long silence.
The unit can be configured to alarm when
either the left or right channel of the main
input source fails, or if the whole stereo signal
fails. There are also options to set the remote
The silence detect level is adjustable between
start output as momentary or latched, to
-60dBu and -15dBu in 3dB steps via a 16
disable switching to the auxiliary input on
position rotary switch on the rear panel. The
alarming and to increase the gain on the
silence interval can be adjusted between
auxiliary input so that an unbalanced input
2 seconds to 30 seconds in 2 second steps,
can be used, for example, from a domestic
or, alternatively, set to 2 minutes 5 seconds
minidisc player.
also via a 16 position rotary switch on the
rear panel. The audio outputs use stereo
Front panel LED indicators show individually
professional balanced XLR-3 male connectors. left and right programme and alarm
conditions for both the main and auxiliary
inputs. The status of the source, mode and
alarm state are also shown on the front
panel with LED indicators.
Additionally, the RB-SD1 can be
programmed for specific applications which
can be defined on power-up of the unit.
Contact Sonifex for further information
if you have a particular requirement.
(Refer to the handbook on the website for
information on current configurations).
The RB-SD1 has been designed to have a
passive signal path through the main input,
so if power to the unit fails, the signal input
will still be routed through to the output.
This is essential for applications such as
installation at transmitter sites, where a
power failure to the unit should not prevent
the audio input signal from being output to
the transmitter.
Audio & Video Interfaces - Synchronisers, Delays & Silence Detectors
Specification For RB-SD1
Audio Specification
Rear Panel Connections and Controls
Front Panel Controls and Indicators
Maximum
Input Level:
+28dBu
Inputs (Main
& Auxiliary):
4 x XLR 3 pin female (balanced,
auxiliary can be unbalanced)
Controls
Source select, mode select and restore
(With Indicators):
Maximum
Output Level:
+28dBu
Output:
2 x XLR 3 pin male (balanced)
Indicator:
Remotes:
15 way D-type plug
Frequency
Response:
20Hz - 20kHz ±0.1dB
Alarm
Threshold:
-15dBu to -60dBu in 3dB steps via
rotary switch
Gain:
+8dB (for unbal input B - optional)
As input for balanced input, <0.05%
ref +8dBu output for unbalanced input
2 - 30 seconds in 2 second intervals &
125 second option via rotary switch
RB-SD1:
Distortion:
Silence Detect
Duration:
Detection Type:
Mono or stereo, via DIP switch
Silence Switch
Defeat:
Disable/enable silence switching, via
DIP switch
Dimensions
(Raw):
48cm (W) x 10.8cm (D) x 4.2cm (H) (1U)
19” (W) x 4.3” (D) x 1.7” (H) (1U)
Remote Start:
Latched or momentary, via DIP switch
Dimensions
(Boxed):
58.5cm (W) x 22.5cm (D) x 7cm (H)
23” (W) x 8.9” (D) x 2.8” (H)
Mains Input:
Filtered IEC, switchable 110-120V, or
220-240V, fused, 9W max
Weight:
Nett: 1.4kg Gross: 2.0kg
Nett: 3.1lbs Gross: 4.4lbs
Fuse Rating:
Anti-surge fuse 100mA 20 x 5mm
(230VAC)
Anti-surge fuse 250mA 20 x 5mm
(115VAC)
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Output
Impedance:
As input, except when using
unbalanced auxiliary input where
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Noise:
<-87dB, unity gain, ref +8dBu output
for unbal input
Program and alarm indicators for left
and right source for both main and
auxiliary channels
Equipment Type
Silence detection unit
Physical Specification
73
Audio & Video Interfaces - Synchronisers, Delays & Silence Detectors
RB-SD1IP Silence Detection Unit With Ethernet & USB
Category: Synchronisers, Delays &
Silence Detectors.
Product Function: To detect silence in
an analogue audio signal and switch to a
second source, or third USB source.
Typical Applications: At a radio
transmitter site, before the input to the
transmitter. A local audio source can
become the secondary input in case
silence is detected and if this fails, USB
audio can play. At a radio studio in case
of failure of broadcast playout server,
it can switch in an emergency playback
system.
Features: IP Enabled with web browser
control interface, SNMP V1 compatible,
trap generation for non-networked
hardware via GPIO port, USB playback
functionality, adjustable silence detect
threshold level and silence duration,
passive signal path, switching of external
equipment on silence detection,
automatic or manual operation, mono
or stereo operation, front panel LED
indications of alarm status.
74
The RB-SD1IP silence detection unit is an
upgraded version of the existing RB-SD1.
The unit is a 1U rack mount device used to
monitor an unattended stereo studio feed
and in the event of the signal going “quiet”
after a given period the unit will switch
through an alternative stereo audio signal.
This signal could be a recorded message
(e.g. “Normal service will be resumed”, etc),
a feed from a flashcard player, audio from a
connected USB flash drive or an alternative
recorded program. Controls are provided
to start external equipment and to provide
remote status indication.
New Features:
The RB-SD1IP offers all of the functionality
of the standard RB-SD1 with several extra
capabilities. Ethernet connectivity provides
the ability to set up and control the unit
via a browser based GUI. The network
capabilities allow the user to more finely
control silence level (-60dBu to 0dBu in
3dBu steps) and silence duration (1 second
to 24 hours). You can also remotely lock/
unlock the front panel controls on the
unit and opt to use either the hardware
configured settings or web based settings.
In addition to the front panel LEDs the GUI
home page offers a real time view of signal
levels and alarm statuses.
Using the new browser GUI, left and right
channels can be treated independently and
remote relay triggers can be configured
as one of many events including the new
GPIO pins. You can also choose to lock/
unlock the use of the remote pins to control
the unit and firmware updates can also be
performed using the web GUI.
SNMP V1 is implemented so that the unit
can be controlled by existing network
management systems. The addition of 6
extra GPIO pins to the rear panel allows
customisable functionality, including the
use of the RB-SD1IP network interface to
generate SNMP Traps on behalf of other,
non-networked, hardware.
The RB-SD1IP has been fitted with a USB
interface on the front panel and can act as
a host in two ways. Firstly the USB port can
be used to upgrade the firmware on the
unit from a USB flash drive. Secondly, such
a drive can hold a pre-recorded message
which the unit can play out in the event that
both main and auxiliary signals fall silent.
RB-SD1IP Home Page.
RB-SD1IP Levels Page.
Audio & Video Interfaces - Synchronisers, Delays & Silence Detectors
Specification For RB-SD1IP
Rear Panel Connections and Controls
Silence Switch Defeat:
Audio Specification
Inputs (Main & Auxiliary): 4 x XLR 3 pin female (balanced,
auxiliary can be unbalanced)
Disable/enable silence
switching, via DIP switch or GUI
Remote Start:
Latched or momentary, via DIP
switch or GUI
Ethernet:
10/100Mbps on 1xRJ45 socket
with status LEDs
Mains Input:
Filtered IEC, continuously rated
85-264VAC @47-63hz 10W max
Fuse Rating:
Anti-surge fuse 1A 20 x 5mm
(250VAC)
Maximum Input Level:
+28dBu
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Output:
Maximum Output Level:
+28dBu
Remotes:
15 way D-Type plug
Output Impedance:
As input, except when using
unbalanced auxiliary input
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GPIO:
9 way D-Type socket
Alarm Threshold:
-15dBu to -60dBu in 3dB steps
via rotary switch 0dBu to -60dBu
in 3dB steps via web GUI
Silence Detect Duration:
2 - 30 seconds in 2 second
intervals & 125 second option
via rotary switch 1 second –
24 hours using web GUI
Frequency Response:
20Hz - 20kHz ±0.1dB
Gain:
+8dB (for unbal input B-optional)
Noise:
<-93dB, unity gain, ref +8dBu
output for unbal input
Distortion:
As input for balanced input,
<0.02% @ 1kHz ref +8dBu
output for unbalanced input
Detection Type:
2 x XLR 3 pin male (balanced)
Mono or Stereo, via DIP switch
Mono, Stereo, or dual mono via
web GUI
Front Panel Controls and Indicators
Controls (With Indicators): Source Select, Mode Select and
Restore
Indicator:
Program and Alarm indicators
for left and right source for both
Main and Auxiliary channels,
power indicator
Reset:
Recessed push button
USB Port:
1 x USB A socket
USB
The RB-SD1IP can act as a host for low powered USB
Mass Storage devices in order to playback Audio files as
an emergency backup system for when both Main and
Auxiliary sources fail.
File System(s):
FAT & FAT32
Supported Audio:
.wav extension (16 bit Stereo
PCM @ 48KHz, 24KHz, 12KHz,
44.1KHz, 22.050KHz, 11.025KHz,
32KHz, 16KHz, 8KHz)
Note: Additional Audio support may be added in future
updates
Equipment Type
RB-SD1IP:
Silence Detection Unit With
Ethernet & USB
Physical Specification
Dimensions (Raw):
48cm (W) x 10.8cm (D) x 4.2cm
(H) (1U)
19” (W) x 4.3” (D) x 1.7” (H) (1U)
Dimensions (Boxed):
58.5cm (W) x 22.5cm (D) x 7cm
(H)
23” (W) x 8.9” (D) x 2.8” (H)
Weight:
Nett: 1.4kg Gross: 2.0kg
Nett: 3.1lbs Gross: 4.4lbs
75
Audio & Video Interfaces - Synchronisers, Delays & Silence Detectors
RB-DSD1 Digital Silence Detection Unit
Category: Synchronisers, Delays &
Silence Detectors.
Product Function: To detect silence in
a digital audio signal and switch to a
second source.
76
192
Typical Applications: At a radio transmitter
site, before the input to the transmitter.
A local audio source can become the
secondary input in case silence is detected.
Features: Multiple digital audio I/O
formats, adjustable silence detect
threshold level and silence duration,
passive signal path, switching of external
equipment on silence detection, automatic
or manual operation, front panel LED
indications of alarm status.
The RB-DSD1 digital silence detection unit
works in a similar way to the Sonifex RBSD1 analogue silence detection unit, but
has AES/EBU, S/PDIF and TOSlink inputs
and outputs instead of analogue inputs and
output respectively. Designed to switch
from one input to another in the event of
loss of audio, the unit is ideal at transmitter
sites, or after the master output of a
studio, to switch in another audio source,
or simultaneous broadcast, should a master
source fail.
RB-DSD1 Digital Silence Detection Unit Shown With Optional Sync Input Board.
The unit can switch:
• On loss of level of the main input.
• On loss of level on one channel of the
main input.
• On loss of synchronisation lock of the
main input.
The RB-DSD1 has 2 x digital stereo audio
inputs, each one selectable via front panel
MAIN and AUX push buttons, from either
AES/EBU balanced XLRs, S/PDIF unbalanced
phonos or TOSlink unbalanced optical
inputs. Sample rate converters on each
Audio & Video Interfaces - Synchronisers, Delays & Silence Detectors
input mean that sources of different sample
rates can be used with the output sample
rate being defined independently. Each
input is user-defined as either the main
source or auxiliary source and both sources
are monitored for failure, each having a
remote failure alarm. The colour of the
MAIN and AUX push-buttons indicate which
input is the main (green) and auxiliary (red)
input, with a flashing LED indicating loss of
synchronisation.
In the event of the main source dropping
below a pre-set level for a pre-determined
amount of time, the unit will automatically
switch through to the auxiliary signal. The
silence detect level is adjustable between
-39dBfs and -84dBfs in 3dBfs steps via front
panel DIPSwitches. The silence interval can
be adjusted between 0 seconds and 252
seconds in 2 second steps via another
front panel DIPswitch block. A small cover
panel can be screwed in place to obscure
the DIPSwitches to prevent tampering of
the settings.
There are 2 stereo outputs to allow for
distribution of the selected input to
multiple outputs. Each output is available
as simultaneous AES/EBU balanced XLRs,
S/PDIF unbalanced phonos or TOSlink
unbalanced optical outputs. The output
sample rates are selectable via rear panel
DIPSwitches from one of 32kHz, 44.1kHz,
48kHz, 88.2kHz, 96kHz, 176.4kHz or
192kHz.
The unit has TTL wordclock BNC and AES/
EBU XLR synchronising inputs as standard
and optionally, the RB-SYA and RB-SYD
synchronisation boards can be fitted to
synchronise the unit to analogue or digital
video signals. A front panel DIPswitch
block is used to decide whether the unit
is synchronised to Input1, Input2, the
AES/EBU sync input, the wordclock sync
input or an optional video sync board.
A front panel SYNC button selects the
synchronisation mode of the unit and the
button flashes whenever the unit is not
synchronised to an incoming sync signal.
Selectable sync modes are as follows:
Master Mode - In this mode the digital
output sample rate is simply set by, and
locked to, the internal on-board clock
generator. No sync signal is used or
required.
Auto Sync Mode - In this mode the
digital output sample rate follows the
selected sync input. When the sync signal
is not present the output sample rate
will be set by, and locked to, the internal
on-board clock generator at the selected
output frequency.
Auto Lock Mode - The digital output
sample rate follows the sync input. If the
sync signal is removed then the output
sample rate will be set by, and locked to,
the internal on-board clock generator at the
closest frequency available to the previous
sync input.
Slave Mode - In this mode the digital
output sample rate follows the sync input.
When the sync signal is not present the
digital output is turned off.
The unit can operate in 2 modes automatic or manual, selectable using a
rear panel DIPswitch. In both modes it will
automatically switch over to the auxiliary
source on detecting silence. When the main
signal is again detected it will either return
to the main signal automatically or manually
depending on the mode chosen. In manual
mode, the front panel RESTORE button is
used to return to the main signal.
The RB-DSD1 has a number of remote
operational features. Remote outputs
provide separate relay contact closures for
failure of the main and auxiliary inputs.
You can also remotely select between
auto and manual mode (with tally output),
action the signal RESTORE, set the silence
detection delay to be 2mins 5 seconds and
define which input is the main input (with
tally output). You can remotely start and
stop another piece of equipment on alarm
failure and there is an option to set the
remote start output as either momentary
or latched.
The unit can be configured to alarm when
either the left or right channel of the main
input source fails, or if the whole stereo
signal fails. Additionally, if one channel of a
stereo signal is lost, you can define whether
to mute the lost channel, or whether to
mix the remaining channel to the lost side,
effectively creating a mono signal. If the
main source synchronisation is lost, you
can define whether the unit switches to the
auxiliary input immediately, or whether to
treat the signal as silence to be detected
and then switched based on the unit’s
silence detection settings.
Front panel LED indicators by the
MAIN and AUX buttons show individually
left and right programme and alarm
conditions for both the main and auxiliary
inputs.
A powerful feature of the RB-DSD1 is that
by using the Sonifex SCi serial software, the
unit can be programmed for different delay
durations, levels and switching functions so
that you can programme the unit for your
specific application. A front panel DIPswitch
configures the unit to be controlled serially
77
Audio & Video Interfaces - Synchronisers, Delays & Silence Detectors
Specification For RB-DSD1
and a front panel LED indicates serial
operation. Contact Sonifex for further
information if you have a particular
requirement that isn’t catered for by the
RB-DSD1 as standard.
The RB-DSD1 has been designed to have a
passive signal path through the main input,
so if power to the unit fails, signal input
1 is routed to output 1 and signal input
2 is routed to output 2. This is essential
for applications such as installation at
transmitter sites, where a power failure to
the unit should not prevent the audio input
signal from being output to the transmitter.
SCi Indicator Page.
Audio Specification
Remote Start:
Latched or momentary, via DIPswitch
Dynamic Range:
>138dB
Channel Loss:
Distortion
and Noise:
<-137dB THD + N at 1kHz, ref 0dB FS
Mute channel or mix remaining, via
DIPswitch
Sync Loss:
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Switch immediately or treat as silence
delay, via DIPswitch
Signal Level:
Balanced: 3V/10V peak to peak min/max
Unbalanced: Min 0.5V±20% peak to peak
Sample
Frequencies:
32, 44.1, 48, 88.2, 96,176.4 or 192kHz
Bit Depth:
Up to and including 24 bit
Connections
Digital Inputs:
2 x AES/EBU XLR 3 pin female
2 x S/PDIF RCA phono
2 x TOSLink optical input
Digital Outputs:
2 x AES/EBU XLR 3 pin plug
2 x S/PDIF RCA phono socket
2 x TOSLink optical output
Sync Inputs:
1 x AES/EBU XLR 3 pin female
1 x Word Clock BNC
1 x Video Input (optional)
Front Panel Operational Controls & Indicators
Digital Input
Select:
AES/EBU, S/PDIF or TOSlink optical
via INPUT 1 or INPUT 2 push-buttons
Sync Input
Select:
AES/EBU, wordclock, INPUT 1, INPUT 2
or video board, via front panel DIPswitch
Sync Mode
Select:
Master, slave, auto or auto lock,
via SYNC push-button
Alarm
Threshold:
-39dBfs to -84dBfs in 3dBfs steps via
front panel DIPSwitches
Equipment Type
Silence Detect
Duration:
0 - 252 seconds in 2 second intervals
via front panel DIPSwitches
Physical Specifications
Detection Type:
Mono or stereo, via front panel
DIPswitch
Restore Control: Manual restore button & mode
indication LED
SCi Miscellaneous Page.
78
Indicators:
Program and alarm indicators for left
and right sources for both main and
auxiliary inputs
Remote I/O Port: 15 way D-type plug
Serial Port:
RS232, 9 way D-Type
Mains Input:
Universal filtered IEC, continuously rated
85-264VAC @47- 63Hz, max 10W
Fuse Rating:
Anti-surge fuse 2A 20 x 5mm
RB-DSD1:
Digital silence detection unit
Dimensions
(Raw):
48cm (W) x 10.8cm (D*) x 4.2cm (H) (1U)
19” (W) x 4.3” (D*) x 1.7” (H) (1U)
Dimensions
(Boxed):
59cm (W) x 27.5cm (D*) x 11cm (H)
23.2” (W) x 10.8” (D*) x 4.3” (H)
Weight:
Nett: 1.4kg Gross: 2.0kg
Nett: 3.1lb Gross: 4.4lb
Accessories
Rear Panel Operational Controls
RB-SYA:
Master
Frequency
Select:
32, 44.1, 48, 88.2, 96,176.4 or 192kHz
via rear panel DIPSwitches
Analogue video sync board (NTSC, PAL
& SECAM)
RB-SYD:
Digital video sync board (SD-SDI
& HD-SDI)
Input Select:
Main input from INPUT 1 or INPUT 2 via
DIPswitch
RB-RK3:
1U Rear panel rack kit for large
Redboxes
Restore Mode:
Automatic or manual, via DIPswitch
* Note that this product is deeper than standard Redboxes
Audio & Video Interfaces - Synchronisers, Delays & Silence Detectors
RB-DSD8 8 Channel Silence Switcher
Category: Synchronisers, Delays &
Silence Detectors.
Product Function: A multi channel (4
stereo) silence detector which supports
both analogue and digital signals and
can switch between multiple 5.1 or 4 x
stereo inputs on silence detection.
Typical Applications: At a radio
transmitter site, as a quad stereo
silence input detector and switcher.
Two separate 5.1/7.1 mixer outputs
can be connected and one switched to
the other in the event of failure of the
primary system.
Features: Analogue and digital I/O,
very flexible configuration via front
panel or Ethernet, channels can switch
independently or can be linked to act
together, adjustable silence detect
threshold level and silence duration,
passive signal path, switching of external
equipment on silence detection,
automatic or manual operation,
mono or stereo operation, front panel
LED indications of alarm status, dual
redundant power supplies.
The RB-DSD8 8 channel silence switcher
works in a similar way to the Sonifex
RB-SD1 and RB-DSD1 but allows for 4
stereo channels of audio. These stereo
audio channels can be either analogue
or digital and can be used independently
to give 4 stereo silence detectors or they
can be linked to handle multichannel
audio inputs, e.g. for 5.1 and 7.1 surround
systems. Designed to switch from one input
(or set of inputs) to another in the event of
loss of audio, the unit is ideal at transmitter
sites, or after the master output of a
studio, to switch in another audio source,
or simultaneous broadcast, should a master
source fail.
The unit can switch:
• On loss of level of the main input.
• On loss of level on one channel of the
main input.
• On loss of synchronisation lock of a main
digital input.
The 2 x 8 channel audio inputs can accept
both digital and analogue connections,
with the unit automatically recognising a
digital input. For the 8 channel outputs, by
setting the appropriate DIPSwitches, each
stereo output can be designated as either
an analogue pair or as a digital output, thus
making the RB-DSD8 incredibly flexible and
suitable for many different applications.
The unit level settings are in dBFS but when
using analogue signals the equivalent full
scale value can be set to +24dBu, +18dBu, or
+12dBu by DIPSwitches.
Each stereo pair has individual settings and
controls and when stereo signals are linked,
the foremost pair determines the switching
characteristics and controls to be used. Each
stereo pair has an AES LED that shows the
status of the digital audio on that channel
and a Selection LED to show that input is
currently being sent to each respective
output. Two Presence LEDS for the left and
right inputs of each stereo pair indicate the
input level of the channels.
The unit can switch between sources
manually or automatically at the push of
a button. If switching manually, silence
detection is disabled and the user chooses
when to switch using the main or backup
buttons. If switching automatically, the
unit switches between the two sources
automatically upon the detection of silence.
Each pair can be set to switch manually
or automatically and the current setting
is indicated by the Mode LED. Link/Select
buttons are used to group channels together
to access multichannel operation and switch
simultaneously. Each pair has a Link/Select
button which illuminates blue when active.
Pressing and holding the first Link/Select
button with any other Link/Select button
causes all inputs up to that point to be
selected.
The RB-DSD8 has a ‘slave mode’ that allows
you to connect two RB-DSD8 units and
control them simultaneously from one unit.
The silence detect level is adjustable
between -39dBFS and -84dBFS in 3dBFS
steps via DIPSwitches and this level is
compared to peak signals. The silence
interval can be adjusted between 2 seconds
and 254 seconds in 2 second steps via
DIPSwitches.
A powerful feature of the RB-DSD8 is that
by using the Sonifex SCi serial software, the
unit can be programmed for different delay
durations, levels and switching functions so
that you can set up the unit for your specific
application. A DIPSwitch configures the unit
to be controlled serially which is indicated
by a front panel LED. You can control the
unit remotely using either USB or Ethernet.
The RB-DSD8 has been designed with dual
redundant power supplies. This means
that if either power supply fails, the other
is ready to take over. In the extremely
unlikely event that both fail, the unit has
been designed with a passive signal path
through the main input. This is essential
for applications such as installation at
79
Audio & Video Interfaces - Synchronisers, Delays & Silence Detectors
transmitter sites, where a power failure to
the unit should not prevent the audio input
signal from being output to the transmitter.
Contact Sonifex for further information if
you have a particular requirement that isn’t
catered for by the RB-DSD8 as standard.
Clocking & Synchronisation
All digital input signals are routed to a
sample rate converter allowing mixed
incoming sample rates to be used. The
output sample rates are selectable from a
80
predefined master clock of 32kHz, 44.1kHz,
48kHz, 88.2kHz, 96kHz, 176.4kHz or 192kHz
or the clock can be derived from a sync
input. When analogue inputs are selected,
the analogue to digital converters are also
clocked at that sample rate.
DIPSwitches choose the synchronisation
mode and the sync source from TTL
wordclock or AES/EBU through the
dual-purpose synchronising input as
standard. A front panel indicator shows
the status of the synchronization input.
Selectable sync modes are as follows:
Master Mode - In this mode the digital
output sample rate is simply set by, and
locked to, the internal on-board clock
generator. No sync signal is used or required.
Auto Lock Mode - In this mode no output
is generated until lock is achieved with a
sync signal. The digital output sample rate
now follows the sync input. If the sync signal
is removed then the output sample rate is
set by, and locked to, the internal on-board
clock generator at the closest frequency
available to the previous sync input.
Slave Mode - Here the digital output
sample rate follows the sync input. When
the sync signal is not present the digital
output is turned off.
Audio & Video Interfaces - Synchronisers, Delays & Silence Detectors
Specification For RB-DSD8
Audio Specification - Digital
Rear Panel - Operational Controls
Equipment Type
Dynamic Range:
>138dB
Silence Threshold:
RB-DSD8:
Distortion & Noise:
<-137dB THD + N at 1kHz,
ref 0dBFS
-39dBfs to -84dBfs in 3dBfs
steps, via rear panel DIPSwitches
Silence Duration:
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0 - 254 seconds in 2 second
intervals duration, via rear panel
DIPSwitches
Dimensions
(Raw):
48cm(W) x 22cm(D) x 4.2cm(H) 1U
19” (W) x 8.7” (D) x 1.7” (H) 1U
Signal Level:
Balanced: 3V/10V peak to peak
min/max
Stereo/Mono
Switching:
Stereo or mono, via rear panel
DIPSwitch
Dimensions
(Boxed):
55cm(W) x 28cm(D) x 17cm(H)
21.7”(W) x 11”(D) x 6.7”
32, 44.1, 48, 88.2, 96,176.4 or
192kHz
Master Select:
Nett: 2.3kg Gross: 3.8kg
Nett: 5.1lb Gross: 8.4lb
Bit Depth:
Up to and including 24 bit
32, 44.1, 48, 88.2, 96,176.4 or
192kHz Output sample rate, via
rear panel DIPSwitches
Weight:
Sample Rates:
Ignore Silence:
Each channel can be set to ignore
silences, via rear panel DIPswitches
Audio Specification - Analogue
Channel Specific Settings Webpage.
Maximum Input
Level:
+24dBu
Remote Control
Enable:
Enabled or disabled, via rear panel
DIPswitch
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Dynamic Range:
>110dB
Sync Mode &
Source Select:
Distortion & Noise:
>82dB THD + N at 1kHz
Sync in master mode or sync from
MAIN input1, AES or wordclock
sync input in auto or slave mode,
via rear panel DIPswitches
Common Mode
Rejection:
>60dB , ref 0dBu
Remote Start:
Latched or momentary, via
DIPswitch
Input Lock Loss:
Switch immediately or treat as
silence delay, via rear panel
DIPswitch
Digital or Analogue
Output:
Digital or analogue, via rear panel
DIPswitches
Full Scale Line Up:
24, 18 or 12 dBu = 0dBFS, via rear
panel DIPswitches
Boot Mode:
Boot in boot or normal via rear
panel DIPswitch
Front Panel Operational Controls
Switch Mode Select:
Via AUTO, MANUAL or SLAVE
push-buttons
Manual Source
Via MAIN and BACKUP
Select:
push-buttons
Group Selection :
Via LINK/SELECT push-buttons
Front Panel Indicators
Presence LEDs:
For all input channels
Link LEDs:
Show which channels are
controlled concurrently
Mode LEDs:
Indicate the current mode selected
for each group
Selection LEDs:
Indicate whether MAIN or BACKUP
is selected
AES LEDs:
Show the state of the digital input
to each group
Remote I/O Port:
25 way D-type female
PSU LEDs:
Show the state of each power
supply
SCi port:
USB or ethernet
Mains Input:
Remote Control LED:
Show if remote control is selected
2 x Universal filtered IEC,
continuously rated 85-264VAC
@47- 63Hz, max 60W
External Sync LED:
Show the state of any sync inputs
used.
Fuse Rating:
2 x Anti-surge fuse 2A 20 x 5mm
Status Webpage.
8 channel silence switcher
Physical Specifications
* Note that this product is deeper than standard Redboxes
Accessories
RB-RK3:
1U Rear panel rack kit for large
Redboxes
Connections
Digital/Analogue
Inputs:
2 x 8 channel inputs on
2 x 25 pin D-type male
Digital/Analogue
Outputs:
1 x 8 channel outputs on
1 x 25 pin D-type female
Sync Inputs:
1 x BNC (Wordclock or AES)
81
Audio & Video Interfaces - Synchronisers, Delays & Silence Detectors
RB-FS42 Audio Failover Switcher, 4 Main I/O, 2 Standby I/O
•
Category: Synchronisers, Delays &
Silence Detectors.
Product Function: To route the source
audio signals in to a standby encoder
in event of encoder fail. Or to route the
destination audio signals from a standby
decoder in event of decoder fail.
Typical Applications: As a failover
switcher for multi channel audio
transport over E1 or IP, typically as
performed by APT Oslo, Prodys Nureus,
where a N+1 topology is adopted.
Features:
• Relay based switching.
• Dual DC, or AC, power supplies (select
when ordering).
• Encoder site (Set via IP): 4 stereo
•
•
•
•
•
•
•
•
program inputs, 4 + 2 stereo program
outputs where each program input/
Output carries: analogue L/R, stereo
AES/EBU & RS232.
Encoder site: Each program output has
an alarm detect GP input.
Decoder Site (Set via IP): 4 +2 stereo
program inputs, 4 program outputs
where each program input carries:
analogue L/R, stereo AES/EBU & RS232.
Decoder Site: Each program input has an
alarm detect GP input.
AES/EBU transparent (for Dolby E
transport).
Passive throughput in event of power
outage.
GPO output for signalling RB-FS42 alarm
conditions: PSU 1/2 fail, Standby 1/2
active, Summary Alarm.
Automatic or Manual reversion modes.
LED indicators on front panel.
IP control, including Web GUI and SNMP.
The RB-FS42 4 + 2 audio failover switcher is a smaller channel count version of the
RB-FS82, offering a lower number of inputs and outputs. It is a more cost effective solution
where failover is needed for a smaller number of sources/destinations and supports all of
the features of the RB-FS82.
The device has 4 main + 2 standby, stereo analogue audio, AES/EBU digital audio and
RS232 connections (both inputs and outputs) and can be configured via Ethernet for two
main operational applications:
82
1. For switching of program sources to a
standby destination in the event of a
destination failure (‘Standbys to outputs’).
Typically this would be audio encoders at a
program distribution head end (for audio
over IP, E1 or other bearer networks), with
“N” x programs feeding “N” x encoders.
If an encoder fails the audio destined for
that encoder gets routed to a standby
encoder so ensuring the continuity of
audio to network transport.
2. Switching of program sources, including
standby sources, to destinations in the
event of source failure (‘Standbys to
inputs’). Typically this would be audio
decoders at a transmission site with “N” x
programs and “N” x decoders feeding “N”
x transmitters. If a decoder fails, the audio
from a standby decoder, or other audio
source such as an mp3 player, overrides
the signal path to the transmitter so
ensuring continuity on air.
The RB-FS42 supports any configuration of
ƵƉƚŽϰŵĂŝŶƉƌŽŐƌĂŵƐŝŐŶĂůƉĂƚŚƐ;EчϰͿ
and there are 2 standby program signal
paths, in either mode of operation. Each
program path simultaneously switches
analogue L/R audio, AES/EBU digital audio
and RS232 data. Each of these signals is
wired on D-Type connectors on the rear
panel.
All signal paths are passive and therefore
completely transparent utilising relay based
switching. This has the benefit of a “straight
wire” topology during normal (alarm free)
operation and also during any power
outage to the device. An additional benefit
of the passive signal path is AES/EBU bit
transparency allowing throughput of AES/
EBU AC3 Dolby E TM signals.
To ensure the passive nature of the device,
switching is determined by alarm (General
Purpose) inputs, with this alarm signalling
in turn being normally provided by the
encoder or decoders (or other devices) at
site. Recognising the mission critical nature
of the system, a high grade of relay is used
in the RB-FS42.
The passive design ensures continuity of
audio in the event of any power outage.
However the RB-FS42 also includes dual
redundant power supplies (85V-264V AC
with a 12V 1A DC backup as standard). The
RB-FS42-DC model accepts two DC power
inputs (24 - 48V DC) via locking 2.5mm pin
power connectors and both power supplies
are monitored by the unit. This means that
if either power supply fails, the other is
ready to take over. In the extremely unlikely
event that both fail, the unit’s passive signal
path ensures a straight wire connection for
all 4 program feeds (analogue, AES/EBU &
Audio & Video Interfaces - Synchronisers, Delays & Silence Detectors
RS232). This is essential for applications such
as installation at transmitter sites, where a
power failure to the unit will not prevent the
audio input signal from being output to each
of the supported 4 transmitters.
Specification For RB-FS42
Audio Specification - Digital
The RB-FS42 uses passive fixed switching relays which don’t
affect the overall audio performance
Analogue Outputs: 4 x outputs on 1 x 25 way
D-type female
Fuse Rating (AC):
Digital Outputs:
RB-FS42:
Audio failover switcher, 4 + 2 inputs
RB-FS42DC:
Audio failover switcher, 4 + 2 inputs,
2 x DC inputs
RS232 Inputs:
4 x RS232 communication lines on
1 x 25 way D-type female
RS232 Outputs:
4 x RS232 communication lines on
1 x 25 way D-type female
Manual Switching: Via Restore 1 & Restore 2
push-buttons
GPI/O:
6 Inputs & 5 outputs on
1 x 25 pin D-Type female
Front Panel Indicators
Standby
1&2 Inputs:
2 x Analogue differential stereo inputs
2 x Stereo digital inputs
2 x RS232 communication line pairs
on 1 x 25 way D-type female
Ethernet Port:
10/100Mbps on 1 x RJ45 socket for
IP control, SNMP and web GUI
Mains Input (AC):
1 x Universal filtered IEC, continuously
rated 85-264VAC @47- 63Hz, max
20W, plus 1 x 12V 1A DC supply,
2.5mm locking socket fused.
2 x 18V-50V 20W max, DC supply,
2.5mm locking socket fused.
Audio Specification - Analogue
Crosstalk:
>86dB
Front Panel Operational Controls
Please see the RB-FS82 description (Page
84) for more information on LED indicators,
GPIO functions, Main and Standby routing,
alarm clearing and SNMP management
features - the RB-FS42 is identical in this
respect.
Contact Sonifex for further information if
you have a particular requirement that isn’t
catered for by the RB-FS42 as standard.
4 x outputs on 1 x 25 way
D-type female
Power LEDs:
2 x Power indicators
Channel Status
LEDs:
8 x Standby status indicators,
2 per channel
Standby Restore
LEDs:
2 x illuminated buttons
Rear Panel Connections
Analogue Inputs:
4 x differential stereo inputs across
1 x 25 way D-type female
Digital Inputs:
4 x inputs on 1 x 25 way
D-type female
or (Dual DC):
1 x Anti-surge fuse 2A 20 x 5mm
Equipment Type
Physical Specifications
Dimensions
(Raw):
48cm(W) x 22cm(D) x 4.2cm(H) 1U
19” (W) x 8.7” (D) x 1.7” (H) 1U
Dimensions
(Boxed):
55cm(W) x 28cm(D) x 17cm(H)
21.7”(W) x 11”(D) x 6.7”
Weight:
Nett: 2.1kg Gross: 3.5kg
Nett: 4.6lb Gross: 7.7lb
* Note that this product is deeper than standard Redboxes
Accessories
RB-RK3:
1U Rear panel rack kit for large
Redboxes
83
Audio & Video Interfaces - Synchronisers, Delays & Silence Detectors
RB-FS82 Audio Failover Switcher, 8 Main I/O, 2 Standby I/O
•
Category: Synchronisers, Delays &
Silence Detectors.
Product Function: To route the source
audio signals in to a standby encoder
in event of encoder fail. Or to route the
destination audio signals from a standby
decoder in event of decoder fail.
Typical Applications: As a failover
switcher for multi channel audio
transport over E1 or IP, typically as
performed by APT Oslo, Prodys Nureus,
where a N+1 topology is adopted.
Features:
• Relay based switching.
• Dual DC, or AC, power supplies (select
when ordering).
• Encoder site (Set via IP): 8 stereo
•
•
•
•
•
•
•
•
program inputs, 8 + 2 stereo program
outputs where each program input/
Output carries: analogue L/R, stereo
AES/EBU & RS232.
Encoder site: Each program output has
an alarm detect GP input.
Decoder Site (Set via IP): 8 +2 stereo
program inputs, 8 program outputs
where each program input carries:
analogue L/R, stereo AES/EBU & RS232.
Decoder Site: Each program input has an
alarm detect GP input.
AES/EBU transparent (for Dolby E
transport).
Passive throughput in event of power
outage.
GPO output for signalling RB-FS82 alarm
conditions: PSU 1/2 fail, Standby 1/2
active, Summary Alarm.
Automatic or Manual reversion modes.
LED indicators on front panel.
IP control, including Web GUI and SNMP.
The RB-FS82 8 + 2 audio failover switcher is
1. For switching of program sources to
an important tool in many critical areas in
a standby destination in the event
telecommunications and broadcast chains.
of a destination failure (‘Standbys to
The device has 8 main + 2 standby, stereo
outputs’). Typically this would be audio
analogue audio, AES/EBU digital audio and
encoders at a program distribution head
RS232 connections (both inputs and outputs)
end (for audio over IP, E1 or other bearer
and can be configured via Ethernet for two
networks), with “N” x programs feeding
main operational applications:
“N” x encoders. If an encoder fails the
84
audio destined for that encoder gets
routed to a standby encoder so ensuring
the continuity of audio to network
transport.
2. Switching of program sources, including
standby sources, to destinations in the
event of source failure (‘Standbys to
inputs’). Typically this would be audio
decoders at a transmission site with
“N” x programs and “N” x decoders
feeding “N” x transmitters. If a decoder
fails, the audio from a standby decoder,
or other audio source such as an mp3
player, overrides the signal path to the
transmitter so ensuring continuity on air.
The RB-FS82 supports any configuration
ŽĨƵƉƚŽϴŵĂŝŶƉƌŽŐƌĂŵƐŝŐŶĂůƉĂƚŚƐ;Eч
8) and there are 2 standby program signal
paths, in either modes of operation. Each
program path simultaneously switches
analogue L/R audio, AES/EBU digital audio
and RS232 data. Each of these signals is
wired on D-type connectors on the rear
panel.
All signal paths are passive and therefore
completely transparent utilising relay based
switching. This has the benefit of a “straight
wire” topology during normal (alarm free)
operation and also during any power
outage to the device. An additional benefit
of the passive signal path is AES/EBU bit
transparency allowing throughput of AES/
EBU AC3 Dolby E (TM) signals.
To ensure the passive nature of the device,
switching is determined by alarm (General
Purpose) inputs, with this alarm signalling
in turn being normally provided by the
encoder or decoders (or other devices) at
site. Recognising the mission critical nature
of the system, a high grade of relay is used
in the RB-FS82.
The passive design ensures continuity of
audio in the event of any power outage.
However the RB-FS82 also includes dual
redundant power supplies (AC with a DC
backup as standard, or dual DC by ordering
RB-FS82DC). This means that if either power
supply fails, the other is ready to take over.
In the extremely unlikely event that both
fail, the unit’s passive signal path ensures a
straight wire connection for all 8 program
feeds (analogue, AES/EBU & RS232). This is
essential for applications such as installation
at transmitter sites, where a power failure
to the unit will not prevent the audio input
signal from being output to each of the
supported 8 transmitters.
A row of LEDs on the front panel confirm the
unit status, with each individual program
path indicated as being in alarm with
Audio & Video Interfaces - Synchronisers, Delays & Silence Detectors
either Standby 1 or Standby 2 programs
clearly confirmed as actively over-riding
the failed signal. Alarm LEDs on the front
panel are also indicated for power supply
1 failure and power supply 2 failure and
these are also mirrored by the device’s own
General Purpose Outputs so facilitating easy
interfacing of the device with the addition of
a summary alarm status GPO. In the event
of alarm clearing, the unit will automatically
revert to normal operation, but a manual
reversion mode is also provided, allowing for
engineering investigation without the unit
‘hunting’ between different signal paths.
Two buttons on the front panel, RESTORE 1
and RESTORE 2, allow manual restoration
of the previously failed signal paths, away
from Standby 1 and Standby 2 respectively
and these can be remotely controlled over
Ethernet.
To facilitate integration with site
management systems the RB-FS82 supports
SNMP control and is configured by a simple
web based GUI.
Contact Sonifex for further information if
you have a particular requirement that isn’t
catered for by the RB-FS82 as standard.
Front Panel Indicators
Power LEDs:
2 x Power indicators
Channel Status
LEDs:
16 x Standby status indicators,
2 per channel
Standby Restore
LEDs:
2 x illuminated buttons
Analogue Inputs:
Digital Inputs:
Fuse Rating (AC):
8 x inputs on 1 x 25 way
D-type female
RB-FS82:
Audio failover switcher, 8 + 2 inputs
RB-FS82DC:
Audio failover switcher, 8 + 2 inputs,
2 x DC inputs
Analogue Outputs: 8 x inputs on 1 x 25 way
D-type female
8 x RS232 communication lines on
1 x 25 way D-type female
The RB-FS82 uses passive fixed switching relays which don’t
affect the overall audio performance
RS232 Outputs:
8 x RS232 communication lines on
1 x 25 way D-type female
Audio Specification - Analogue
GPI/O:
10 Inputs & 5 outputs on
1 x 25 pin D-Type female
Standby
1&2 Inputs:
2 x Analogue differential stereo inputs
2 x Stereo digital inputs
2 x RS232 communication line pairs
on 1 x 25 way D-type female
>86dB
Front Panel Operational Controls
Manual Switching: Via Restore 1 & Restore 2
push-buttons
1 x Universal filtered IEC, continuously
rated 85-264VAC @47- 63Hz, max
20W, plus 1 x 12V 1A DC supply,
2.5mm locking socket fused.
2 x 18V-50V 20W max, DC supply,
2.5mm locking socket fused.
8 x differential stereo inputs across
2 x 25 way D-types female
RS232 Inputs:
Crosstalk:
Mains Input (AC):
or (Dual DC):
8 x outputs on 1 x 25 way
D-type female
Audio Specification - Digital
10/100Mbps on 1 x RJ45 socket for
IP control, SNMP and web GUI
Rear Panel Connections
Digital Outputs:
Specification For RB-FS82
Ethernet Port:
1 x Anti-surge fuse 2A 20 x 5mm
Equipment Type
Physical Specifications
Dimensions
(Raw):
48cm(W) x 22cm(D) x 4.2cm(H) 1U
19” (W) x 8.7” (D) x 1.7” (H) 1U
Dimensions
(Boxed):
55cm(W) x 28cm(D) x 17cm(H)
21.7”(W) x 11”(D) x 6.7”
Weight:
Nett: 2.2kg Gross: 3.6kg
Nett: 4.8lb Gross: 8.0lb
* Note that this product is deeper than standard Redboxes
Accessories
RB-RK3:
1U Rear panel rack kit for large
Redboxes
85
Audio & Video Interfaces - Matching Converters
RB-UL1 Single Stereo Unbalanced To Balanced Converter
The RB-UL1 is a single stereo unit for
interfacing domestic or semi-professional
unbalanced equipment, such as a CD
player, to professional balanced line levels.
Category: Matching Converters.
Product Function: Converting
unbalanced audio signals to balanced
audio signals.
Typical Applications: Interfacing
domestic unbalanced audio devices e.g.
a CD player or ipod to balanced audio
equipment.
Features: Pre-set left and right gain
controls, Neutrik XLR output connectors.
86
The two RCA unbalanced inputs have an
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balanced XLR-3 outputs with an output
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The output gain can be individually adjusted
for left and right channels by using pre-set
potentiometers accessible through the rear
panel.
Specification For RB-UL1
Audio Specification
Mains Input:
Maximum
Input Level:
Filtered IEC, switchable 110-120V, or
220-240V, fused, 6W max
Fuse Rating:
Anti-surge fuse 100mA 20 x 5mm
(230VAC)
Anti-surge fuse 250mA 20 x 5mm
(115VAC)
+28dBu
/ŶƉƵƚ/ŵƉĞĚĂŶĐĞ͗хϭϬŬɏ
Maximum
Output Level:
+28dBu
KƵƚƉƵƚ
Impedance:
фϱϬɏ
Distortion:
0.01% THD @ 1kHz, ref +8dBu output
Noise:
-100dB, unity gain, ref +8dBu output
&ƌĞƋƵĞŶĐLJ
Response:
ϮϬ,njƚŽϮϬŬ,njцϬ͘ϭĚ;ϲϬϬɏůŽĂĚ͕
ref 1kHz)
Gain Range:
Balanced output : -15dBu to +15dBu,
ref -15dBu into unbalanced RCA input
Connections
Inputs:
2 x RCA phono (Unbalanced)
Outputs:
2 x XLR 3 pin male (Balanced)
Equipment Type
RB-UL1:
Single stereo unbalanced to balanced
converter
Physical Specification
Dimensions
(Raw):
28cm (W) x 10.8cm (D) x 4.2cm (H) (1U)
11” (W) x 4.3” (D) x 1.7” (H) (1U)
Dimensions
(Boxed):
36cm (W) x 20.5cm (D) x 6cm (H)
14.2” (W) x 8” (D) x 2.4” (H)
Weight:
Nett: 1.00kg Gross: 1.45kg
Nett: 2.2lbs Gross: 3.2lbs
Audio & Video Interfaces - Matching Converters
RB-UL2 Dual Stereo Unbalanced To Balanced Converter
Category: Matching Converters.
Product Function: Converting
unbalanced audio signals to balanced
audio signals for two stereo channels.
Typical Applications: Interfacing
domestic unbalanced audio devices e.g.
a CD player or ipod to balanced audio
equipment.
Features: Pre-set left and right gain
controls, Neutrik XLR output connectors.
The RB-UL2 is a dual stereo unit for
interfacing domestic or semi-professional
unbalanced equipment to professional
balanced line levels.
Specification For RB-UL2
Connections
Audio Specification
Inputs:
4 x RCA phono (Unbalanced)
Maximum
Input Level:
+28dBu
Outputs:
4 x XLR 3 pin male (Balanced)
Mains Input:
/ŶƉƵƚ
Impedance:
хϭϬŬɏ
Filtered IEC, switchable 110-120V, or
220-240V, fused, 6W max
All connections are on the rear panel. Four
RCA unbalanced inputs have an impedance
ŽĨϭϬŬɏĂŶĚĂƌĞƌŽƵƚĞĚƚŽĨŽƵƌďĂůĂŶĐĞĚ
XLR-3 outputs with an output impedance
ŽĨфϱϬɏ͘
Fuse Rating:
KƵƚƉƵƚ
Impedance:
фϱϬɏ
Maximum
Output Level:
+28dBu
Anti-surge fuse 100mA 20 x 5mm
(230VAC)
Anti-surge fuse 250mA 20 x 5mm
(115VAC)
Distortion:
0.01% THD @ 1kHz, ref +8dBu output
Noise:
-100dB, unity gain, ref +8dBu output
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Response:
ϮϬ,njƚŽϮϬŬ,njцϬ͘ϭĚ;ϲϬϬɏůŽĂĚ͕
ref 1kHz)
Gain Range:
Balanced output : -15dBu to +15dBu,
ref -15dBu into unbalanced RCA input
The output gain can be individually adjusted
for left and right channels by using pre-set
potentiometers accessible through the rear
panel, allowing you to feed both balanced
and unbalanced equipment.
Equipment Type
RB-UL2:
Dual stereo unbalanced to balanced
converter
Physical Specification
Dimensions
(Raw):
28cm (W) x 10.8cm (D) x 4.2cm (H) (1U)
11” (W) x 4.3” (D) x 1.7” (H) (1U)
Dimensions
(Boxed):
36cm (W) x 20.5cm (D) x 6cm (H)
14.2” (W) x 8” (D) x 2.4” (H)
Weight:
Nett: 1.05kg Gross: 1.5kg
Nett: 2.3lbs Gross: 3.3lbs
87
Audio & Video Interfaces - Matching Converters
RB-UL4 Quad Stereo Unbalanced To Balanced Converter
Category: Matching Converters.
Product Function: Converting
unbalanced audio signals to balanced
audio signals for four stereo channels.
Typical Applications: Interfacing
domestic unbalanced audio devices e.g.
a CD player or ipod to balanced audio
equipment.
Features: Pre-set left and right gain
controls, Neutrik XLR output connectors.
88
The RB-UL4 is a 1U rack-mount quad stereo
unit for interfacing domestic or semiprofessional unbalanced equipment to
professional balanced line levels.
Specification For RB-UL4
Connections
Audio Specification For RB-UL4
Inputs:
8 x RCA phono (Unbalanced)
Maximum
Input Level:
+28dBu
Outputs:
8 x XLR 3 pin male (Balanced)
Mains Input:
Maximum
Output Level:
+28dBu
Filtered IEC, switchable 110-120V, or
220-240V, fused, 6W max
All connections are on the rear panel.
The eight RCA unbalanced inputs have an
ŝŵƉĞĚĂŶĐĞŽĨϭϬŬɏĂŶĚĂƌĞƌŽƵƚĞĚƚŽĞŝŐŚƚ
balanced XLR-3 outputs with an output
ŝŵƉĞĚĂŶĐĞŽĨфϱϬɏ͘
Fuse Rating:
/ŶƉƵƚ
Impedance:
хϭϬŬɏ
KƵƚƉƵƚ
Impedance:
фϱϬɏ
Anti-surge fuse 100mA 20 x 5mm
(230VAC)
Anti-surge fuse 250mA 20 x 5mm
(115VAC)
Distortion:
0.01% THD @ 1kHz, ref +8dBu output
Noise:
-100dB, unity gain, ref +8dBu output
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Response:
ϮϬ,njƚŽϮϬŬ,njцϬ͘ϭĚ;ϲϬϬɏůŽĂĚ͕
ref 1kHz)
Gain Range:
Balanced output : -15dBu to +15dBu,
ref -15dBu into unbalanced RCA input
The output gain can be individually adjusted
for left and right channels by using pre-set
potentiometers accessible through the rear
panel.
Equipment Type
RB-UL4:
Quad stereo unbalanced to balanced
converter
Physical Specification
Dimensions
(Raw):
48cm (W) x 10.8cm (D) x 4.2cm (H) (1U)
19” (W) x 4.3” (D) x 1.7” (H) (1U)
Dimensions
(Boxed):
58.5cm (W) x 22.5cm (D) x 7cm (H)
23” (W) x 8.9” (D) x 2.8” (H)
Weight:
Nett: 1.3kg Gross: 1.9kg
Nett: 2.9lbs Gross: 4.2lbs
Audio & Video Interfaces - Matching Converters
RB-LU4 Quad Stereo Balanced To Unbalanced Converter
The RB-LU4 is a 1U rack-mount quad stereo
unit for interfacing professional balanced
line levels to domestic or semi-pro
unbalanced equipment, e.g. for connecting
a pro satellite receiver to a consumer
hi-fi system.
All connections are on the rear panel.
The eight balanced XLR-3 inputs have an
ŝŵƉĞĚĂŶĐĞŽĨϮϬŬɏĂŶĚĂƌĞƌŽƵƚĞĚƚŽĞŝŐŚƚ
unbalanced RCA outputs with an output
ŝŵƉĞĚĂŶĐĞŽĨфϱϬɏ͘
The output gain can be individually adjusted
for left and right channels by using pre-set
potentiometers accessible through the
rear panel.
A LED power indicator on the front panel
displays the power supply connection.
Connections
Specification For RB-LU4
Audio Specification
Maximum
Input Level:
+28dBu
/ŶƉƵƚ
хϮϬŬɏďĂůĂŶĐĞĚďƌŝĚŐŝŶŐ
Impedance (XLR):
Inputs:
8 x XLR 3 pin female (Balanced)
Outputs:
8 x RCA phono (Unbalanced)
Mains Input:
Filtered IEC, switchable 110-120V, or
220-240V, fused, 6W max
Fuse Rating:
Anti-surge fuse 100mA 20 x 5mm
(230VAC)
Anti-surge fuse 250mA 20 x 5mm
(115VAC)
Equipment Type
KƵƚƉƵƚ
фϱϬɏ
Impedance (RCA):
RB-LU4:
Maximum
Output Level:
+22dBu
Physical Specification
Distortion:
0.01% THD @ 1kHz, ref +8dBu output
Dimensions
(Raw):
48cm (W) x 10.8cm (D) x 4.2cm (H) (1U)
19” (W) x 4.3” (D) x 1.7” (H) (1U)
Noise:
-100dB, unity gain, ref +8dBu output
Common Mode
Rejection:
>66dB typically
Dimensions
(Boxed):
58.5cm (W) x 22.5cm (D) x 7cm (H)
23” (W) x 8.9” (D) x 2.8” (H)
Weight:
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Response:
ϮϬ,njƚŽϮϬŬ,njцϬ͘ϭĚ;ϲϬϬɏůŽĂĚ͕
ref 1kHz)
Nett: 1.3kg
Nett: 2.9lbs
Gain Range:
Unbalanced Output : -28dBu to
0dBu, ref 0dBu into balanced XLR input
Quad stereo balanced to unbalanced
converter
Gross: 1.9kg
Gross: 4.2lbs
Category: Matching Converters.
Product Function: Converting balanced
audio signals to unbalanced audio
signals for four stereo channels.
Typical Applications: Interfacing a
professional satellite receiver to an
unbalanced hi-fi system.
Features: Pre-set left and right output
gain controls, Neutrik XLR input
connectors.
89
Audio & Video Interfaces - Matching Converters
RB-BL2 Single Stereo Bi-Directional Matching Converter
Category: Matching Converters.
Product Function: Converting
unbalanced audio signals to balanced
audio signals and vice-versa for one
stereo channel.
Typical Applications: Providing
bi-directional matching conversion
between an unbalanced and balanced
piece of equipment, typically between
an unbalanced portable recorder &
balanced USB or flashcard playback
device.
Features: Pre-set left and right output
gain controls, Neutrik XLR input and
output connectors.
The RB-BL2 is a bi-directional stereo unit
for interfacing domestic or semi-pro
unbalanced equipment to professional
balanced line levels, and vice-versa.
The two XLR-3 electronically balanced inputs
ŚĂǀĞĂŶŝŵƉĞĚĂŶĐĞŽĨϮϬŬɏďƌŝĚŐŝŶŐĂŶĚ
are routed to two unbalanced RCA (phono)
ŽƵƚƉƵƚƐǁŝƚŚĂŶŽƵƚƉƵƚŝŵƉĞĚĂŶĐĞŽĨфϱϬɏ͘
The two RCA unbalanced inputs have an
ŝŵƉĞĚĂŶĐĞŽĨϮϬŬɏĂŶĚĂƌĞƌŽƵƚĞĚƚŽƚǁŽ
balanced XLR-3 outputs with an output
ŝŵƉĞĚĂŶĐĞŽĨфϱϬɏ͘ůůĐŽŶŶĞĐƚŝŽŶƐĂƌĞŽŶ
the rear panel.
The output gain can be adjusted for
left and right channels by using pre-set
potentiometers accessible through the rear
panel.
90
Specification For RB-BL2
Audio Specification
Connections
Maximum
Input Level:
+28dBu
Inputs:
2 x RCA phono (Unbal), 2 x XLR 3 pin
female (Bal)
Maximum
Output Level:
+28dBu
Outputs:
2 x XLR 3 pin male (Bal), 2 x RCA phono
(Unbal)
/ŶƉƵƚ
хϭϬŬɏƵŶďĂůĂŶĐĞĚ
Impedance (RCA):
Mains Input:
Filtered IEC, switchable 110-120V, or
220-240V, fused, 6W max
/ŶƉƵƚ
хϮϬŬɏďĂůĂŶĐĞĚďƌŝĚŐŝŶŐ
Impedance (XLR):
Fuse Rating:
Anti-surge fuse 100mA 20 x 5mm
(230VAC)
Anti-surge fuse 250mA 20 x 5mm
(115VAC)
KƵƚƉƵƚ
фϱϬɏ
Impedance (RCA):
KƵƚƉƵƚ
фϱϬɏ
Impedance (XLR):
Distortion:
0.01% THD @ 1kHz, ref +8dBu output
Noise:
-100dB, unity gain, ref +8dBu output
Equipment Type
RB-BL2:
Single stereo bi-directional matching
converter
Physical Specification
Common
>66dB typically
Mode Rejection:
Dimensions
(Raw):
28cm (W) x 10.8cm (D) x 4.2cm (H) (1U)
11” (W) x 4.3” (D) x 1.7” (H) (1U)
&ƌĞƋƵĞŶĐLJ
Response:
ϮϬ,njƚŽϮϬŬ,njцϬ͘ϭĚ;ϲϬϬɏůŽĂĚ͕
ref 1kHz)
Dimensions
(Boxed):
36cm (W) x 20.5cm (D) x 6cm (H)
14.2” (W) x 8” (D) x 2.4” (H)
Gain Range:
Unbalanced Output : -28dBu to
0dBu, ref 0dBu into balanced XLR input
Balanced Output : -15dBu to +15dBu,
ref -15dBu into unbalanced RCA input
Weight:
Nett: 1.0kg Gross: 1.4kg
Nett: 2.2lbs Gross: 3.1lbs
Audio & Video Interfaces - Matching Converters
RB-BL4 Dual Stereo Bi-Directional Matching Converter
&ƌĞƋƵĞŶĐLJ
Response:
ϮϬ,njƚŽϮϬŬ,njцϬ͘ϭĚ;ϲϬϬɏůŽĂĚ͕
ref 1kHz)
Gain Range:
Unbalanced Output : -28dBu to
0dBu, ref 0dBu into balanced XLR
input
Balanced Output : -15dBu to +15dBu,
ref -15dBu into unbalanced RCA input
The RB-BL4 is a dual bi-directional stereo
unit for interfacing domestic or semi-pro
unbalanced equipment to professional
balanced line levels, and vice versa.
The output gain can be adjusted for
left and right channels by using pre-set
potentiometers accessible through the
rear panel.
The four XLR-3 electronically balanced
ŝŶƉƵƚƐŚĂǀĞĂŶŝŵƉĞĚĂŶĐĞŽĨϮϬŬɏďƌŝĚŐŝŶŐ
and are routed to four unbalanced RCA
(phono) outputs with an output impedance
ŽĨфϱϬɏ͘
Specification For RB-BL4
Connections
Audio Specification
Inputs:
Maximum
Input Level:
+28dBu
4 x RCA phono (Unbal), 4 x XLR 3 pin
female (Bal)
Outputs:
Maximum
Output Level:
+28dBu
4 x XLR 3 pin male (Bal), 4 x RCA phono
(Unbal)
Mains Input:
Filtered IEC, switchable 110-120V, or
220-240V, fused, 6W max
Fuse Rating:
Anti-surge fuse 100mA 20 x 5mm
(230VAC)
Anti-surge fuse 250mA 20 x 5mm
(115VAC)
The four RCA unbalanced inputs have an
ŝŵƉĞĚĂŶĐĞŽĨϮϬŬɏĂŶĚĂƌĞƌŽƵƚĞĚƚŽĨŽƵƌ
balanced XLR-3 outputs with an output
ŝŵƉĞĚĂŶĐĞŽĨфϱϬɏ͘ůůĐŽŶŶĞĐƚŝŽŶƐĂƌĞŽŶ
the rear panel.
/ŶƉƵƚ
хϭϬŬɏƵŶďĂůĂŶĐĞĚ
Impedance (RCA):
/ŶƉƵƚ
хϮϬŬɏďĂůĂŶĐĞĚďƌŝĚŐŝŶŐ
Impedance (XLR):
KƵƚƉƵƚ
фϱϬɏ
Impedance (RCA):
Equipment Type
KƵƚƉƵƚ
фϱϬɏ
Impedance (XLR):
RB-BL4:
Dual stereo bi-directional matching
converter
Distortion:
0.01% THD @ 1kHz, ref +8dBu output
Physical Specification
Noise:
-100dB, unity gain, ref +8dBu output
Common Mode
Rejection:
>66dB typically
Dimensions
(Raw):
48cm (W) x 10.8cm (D) x 4.2cm (H) (1U)
19” (W) x 4.3” (D) x 1.7” (H) (1U)
Dimensions
(Boxed):
58.5cm (W) x 22.5cm (D) x 7cm (H)
23” (W) x 8.9” (D) x 2.8” (H)
Weight:
Nett: 1.3kg Gross: 1.9kg
Nett: 2.9lbs Gross: 4.2lbs
Category: Matching Converters.
Product Function: Converting
unbalanced audio signals to balanced
audio signals and vice-versa for two
stereo channels.
Typical Applications: Providing
bi-directional matching conversion
between an unbalanced and balanced
piece of equipment, typically between
an unbalanced portable recorder &
balanced USB or flashcard playback
device.
Features: Pre-set left and right output
gain controls, Neutrik XLR input and
output connectors.
91
Audio & Video Interfaces - Matching Converters
RB-PA2 Dual Stereo RIAA Phono Amplifier
Category: Matching Converters.
Description: Dual Stereo RIAA Phono
Amplifier.
Product Function: A record player/
turntable pre-amp providing RIAA
equalisation.
Typical Applications: Specifically for use
with record decks/DJ turntables e.g. for
vinyl transfers.
Features: Superbly transparent signal,
powerful bass response, Neutrik XLR
connectors, adjustable output gain.
The RB-PA2 is a dual stereo RIAA equalised
phono amplifier. This record player preamp
amplifies the small signal from your pick
up cartridge (either moving magnet or high
output moving coil magnetic) and provides
the necessary RIAA equalisation required
for vinyl records, to match it to a line input
of your mixer, or amplifier. It uses a clean
RIAA response making it ideal for use with
broadcast turntables and DJ decks and
where vinyl records are being archived
to CD for high quality transfers, or for
converting albums to mp3.
92
It operates with phono cartridges with
a nominal output of 1mV to 25mV and
has been specifically tested to work with
the Technics SL-1200™ and 1210™ range
of turntables. Quality high specification
components have been used together with
low noise, fast acting operational amplifiers
to produce a superbly transparent signal
path. Additionally, the RB-PA2 runs without
a rumble filter to produce a powerful bass
response.
The RB-PA2 is a dual stereo unit to interface
up to two turntables. The inputs are phono
connectors with Neutrik XLR connectors
for the balanced outputs, which may be
wired unbalanced if required. Output gain
is individually adjustable for left and right
channels in the range 35dB to 65dB by using
pre-set potentiometers. Frequency response
is held within 0.5dB of the RIAA curve and
noise is typically better than -84dB RMS A
weighted at 40dB gain.
Headroom:
27dB for 3mV input
Dynamic Range:
90dB
Connections
Analogue Inputs: 4 x RCA Phono sockets.
Specification For RB-PA2
Analogue
Outputs:
Audio Specification
Earth Tag:
Grounding turret tag
/ŶƉƵƚ
Impedance:
хϰϳŬɏ
Mains Input:
Filtered IEC, switchable 110-120V,
or 220-240V, fused, 6W max.
Input
1mV to 31mV (ref 1kHz)
Sensitivity Range:
Fuse Rating:
Anti-surge fuse 100mA 20 x 5mm
(230VAC)
Anti-surge fuse 250mA 20 x 5mm
(115VAC)
Maximum
Output Level:
+28dBu
KƵƚƉƵƚ
impedance:
фϱϬɏ
Output
Gain Range:
Adjustable 28dB to 60dB gain via
4 multi-turn pots
Crosstalk:
Better than -80dBu at 1kHz
Frequency
Response:
4 x XLR 3 pin male (balanced) (L & R)
Equipment Type
RB-PA2:
Dual stereo RIAA phono amplifier
Physical Specification
Dimensions
(Raw):
28cm (W) x 10.8cm (D) x 4.2cm (H) (1U)
11” (W) x 4.3” (D) x 1.7” (H) (1U)
10Hz-15kHz (-3dB)
Dimensions
(Boxed):
36cm (W) x 20.5cm (D) x 6cm (H)
14.2” (W) x 8” (D) x 2.4” (H)
RIAA Accuracy:
Within 0.5dB of RIAA curve
Weight:
Distortion:
0.01% THD @ 1kHz, ref +8dBu output
Nett: 1.1kg Gross: 1.5kg
Nett: 2.4lbs Gross: 3.3lbs
Noise:
-77dB (CCIR Q-Pk, 20Hz-20kHz)
Audio & Video Interfaces - Matching Converters
RB-LI2 Stereo Line Isolation Unit
Category: Matching Converters.
Product Function: Used to isolate audio
signals from inter-area ground hum
loops.
Typical Applications: To reduce earth
hum problems, especially when the
equipment at each end of your cable
run is powered for either different mains
power supplies or different phases of
the same supply. To create an isolated
distribution point, or if you need to take
a feed from somewhere that doesn’t
have a distribution amplifier.
Features: Transformer isolation, loopthrough output, output gain control.
The RB-LI2 stereo line isolation unit is used
to isolate audio signals from inter-area
ground hum loops, which could be caused
by equipment being powered by different
mains power supplies, or different phases
on the same supply.
The input and output are connected
together through a transformer, which has
internal jumpers allowing the outputs to be
balanced about ground.
accessible through the rear panel. The gain
range of the stereo line isolation unit is
-15dB to +13.5dB (28.5dB).
This unit is useful where audio is required
to be driven over a relatively long length
of cable. By isolating the signal using
transformers, ground loop currents that
can be present in non-isolated signals, are
eradicated completely.
Specification For RB-LI2
There is also a loop-through output, so
that the RB-LI2 can be inserted into a line,
forming a transformer balanced distribution
point.
Audio Specification
The isolated outputs can be individually
adjusted using pre-set potentiometers,
Frequency
Response:
Output
Gain Range:
-15dB to +13.5dB
Connections
Inputs:
2 x XLR 3 pin female (Balanced)
Isolated Outputs: 2 x XLR 3 pin male (Balanced)
Loop Outputs:
2 x XLR 3 pin male (Balanced)
Mains Input:
Filtered IEC, switchable 110-120V,
or 220-240V, fused, 6W max
Fuse Rating:
Anti-surge fuse 100mA 20 x 5mm
(230VAC)
Anti-surge fuse 250mA 20 x 5mm
(115VAC)
Equipment Type
RB-LI2:
Stereo line isolation unit
Physical Specification
KƵƚƉƵƚ
impedance:
фϭϱϬɏ
Dimensions
(Raw):
28cm (W) x 10.8cm (D) x 4.2cm (H) (1U)
11” (W) x 4.3” (D) x 1.7” (H) (1U)
Distortion:
0.5% THD @ 40Hz, ref +17dBu output
Dimensions
(Boxed):
36cm (W) x 20.5cm (D) x 6cm (H)
14.2” (W) x 8” (D) x 2.4” (H)
Weight:
Nett: 1.1kg Gross: 1.5kg
Nett: 2.4lbs Gross: 3.3lbs
Common
<64dB typically
Mode Rejection:
10Hz to 36kHz ±0.5dB
93
Audio & Video Interfaces - Matching Converters
RB-PLI6 6 Way Mono Passive Line Isolation Unit
Category: Matching Converters.
Product Function: Used to isolate audio
signals from inter-area ground hum
loops for 6 mono or 3 stereo channels
Typical Applications: To reduce earth hum
problems, especially when the equipment
at each end of your cable run is powered
for either different mains power supplies
or different phases of the same supply. OB
trucks where multiple isolated audio feeds
are required to different locations on the
broadcast.
Features: Passive unit so no power is
required and it eliminates ground hum.
The RB-PLI6 passive line isolation unit is
used to isolate audio signals from inter-area
ground hum loops, which could be caused
by equipment being powered by different
mains power supplies or phases on the
same supply.
The input and output are connected
together through a transformer, and the
unit has internal jumpers allowing the
inputs and/or outputs to be balanced about
ground. The unit requires no mains power
for operation.
This unit is useful where audio is required
to be driven over a relatively long length
of cable. By isolating the signal using
transformers, ground loop currents which
can be present in non-isolated signals, are
eradicated completely.
94
Specification For RB-PLI6
Audio Specification
KƵƚƉƵƚ
impedance:
фϭϱϬɏ
Distortion:
0.5% THD @ 40Hz, ref +17dBu output
Common
>64dB typically
Mode Rejection:
Frequency
Response:
10Hz to 36kHz ±0.5dB
Connections
Inputs:
6 x XLR 3 pin female (Balanced)
Outputs:
6 x XLR 3 pin male (Balanced)
Equipment Type
RB-PLI6:
6 way mono passive line isolation unit
Physical Specification
Dimensions
(Raw):
48cm (W) x 10.8cm (D) x 4.2cm (H) (1U)
19” (W) x 4.3” (D) x 1.7” (H) (1U)
Dimensions
(Boxed):
58.5cm (W) x 22.5cm (D) x 7cm (H)
23” (W) x 8.9” (D) x 2.8” (H)
Weight:
Nett: 1.5kg Gross: 1.9kg
Nett: 3.3lbs Gross: 4.2lbs
Audio & Video Interfaces - Audio Distribution Amplifiers
RB-DA6 6 Way Stereo Distribution Amplifier
The RB-DA6 is a 1U rack-mount high
performance 6 way stereo distribution
amplifier for splitting a source into a
number of different outputs.
The RB-DA6 has 1 stereo input and 6 stereo
outputs. It can also be configured so that 1
mono input can be distributed to 12 outputs
by use of a switch which is recessed on the
front panel to prevent it being accidentally
knocked.
The XLR-3 inputs and outputs are
electronically balanced and can be wired
unbalanced. Each output is individually
buffered so that a short circuit on one
output won’t affect the others.
The left and right input gain controls
(normalising) are pre-set potentiometers
accessible through the front panel.
The output gain may be varied from -8dB
to 18dB which is useful for normalising
consumer and professional signals to give
outputs of -15dBu and 0dBu respectively.
Category: Audio Distribution Amplifiers.
Product Function: Audio distribution to
multiple destinations.
Typical Applications: Distribute line level
signals to multiple reporters in a broadcast
centre, distribute main programme output
from a mixer to other studios.
Features: Input level adjustment,
mono/stereo switch, Neutrik XLR audio
connectors.
Specification For RB-DA6
Connections
Audio Specification
Inputs:
Maximum
Input Level:
+28dBu
2 x XLR 3 pin female (balanced, can
be unbalanced)
Outputs:
Maximum
Output Level:
+28dBu
12 x XLR 3 pin male (balanced, can
be unbalanced)
Mains Input:
&ƌĞƋƵĞŶĐLJ
Response:
ϮϬ,njƚŽϮϬŬ,njцϬ͘ϭĚ;ϲϬϬɏůŽĂĚ͕
ref 1kHz)
Filtered IEC, switchable 110-120V, or
220-240V, fused, 6W max
Fuse Rating:
Gain Range:
Adjustable 8dB loss to 18dB gain
(L & R adjust)
Anti-surge fuse 100mA 20 x 5mm
(230VAC)
Anti-surge fuse 250mA 20 x 5mm
(115VAC)
Common
>66dB typically
Mode Rejection:
Equipment Type
/ŶƉƵƚ
impedance:
хϮϬŬɏďƌŝĚŐŝŶŐ
Physical Specification
KƵƚƉƵƚ
impedance:
фϱϬɏ
Dimensions
(Raw):
48cm (W) x 10.8cm (D) x 4.2cm (H) (1U)
19” (W) x 4.3” (D) x 1.7” (H) (1U)
Distortion:
0.01% THD @ 1kHz, ref +8dBu output
Noise:
-100dB unity gain, ref +8dBu output
Dimensions
(Boxed):
58.5cm (W) x 22.5cm (D) x 7cm (H)
23” (W) x 8.9” (D) x 2.8” (H)
Weight:
Nett: 1.3kg Gross: 1.9kg
Nett: 2.9lbs Gross: 4.2lbs
RB-DA6:
6 way stereo distribution amplifier
95
Audio & Video Interfaces - Audio Distribution Amplifiers
RB-DA6G 6 Way Stereo Distribution
Amplifier With Output Gain Control
Category: Audio Distribution Amplifiers.
Product Function: Audio distribution to
multiple destinations.
Typical Applications: Distribute line level
signals to multiple reporters in a broadcast
centre, distribute main programme output
from a mixer to other studios, create
multiple feeds of a recorded signal at
different output levels.
Features: Output level adjustment,
mono/stereo switch, Neutrik XLR audio
connectors.
The RB-DA6G is a 1U rack-mount high
performance 6 way stereo distribution
amplifier for splitting a source into a
number of different outputs. It is identical
to the RB-DA6 with the addition of
individual output gain adjustment, instead
of global stereo gain adjustment.
Specification For RB-DA6G
Audio Specification
Maximum
Input Level:
+28dBu
Maximum
Output Level:
+28dBu
&ƌĞƋƵĞŶĐLJ
Response:
ϮϬ,njƚŽϮϬŬ,njцϬ͘ϭĚ;ϲϬϬɏůŽĂĚ͕
ref 1kHz)
Gain Range:
Adjustable 8dB loss to 18dB gain
(12 adjustable pots)
Common
>66dB typically
Mode Rejection:
/ŶƉƵƚ
impedance:
96
хϮϬŬɏďƌŝĚŐŝŶŐ
KƵƚƉƵƚ
impedance:
фϱϬɏ
Distortion:
0.01% THD @ 1kHz, ref +8dBu output
Noise:
-88dB unity gain, ref +8dBu output
Connections
Inputs:
unbalanced)
2 x XLR 3 pin female (balanced, can be
Outputs:
12 x XLR 3 pin male (balanced, can be
unbalanced)
Mains Input:
Filtered IEC, switchable 110-120V, or
220-240V, fused, 6W max
Fuse Rating:
Anti-surge fuse 100mA 20 x 5mm
(230VAC)
Anti-surge fuse 250mA 20 x 5mm
(115VAC)
Equipment Type
RB-DA6G:
6 way stereo distribution amplifier
with output gain control
Physical Specification
Dimensions
(Raw):
48cm (W) x 10.8cm (D) x 4.2cm (H) (1U)
19” (W) x 4.3” (D) x 1.7” (H) (1U)
Dimensions
(Boxed):
58.5cm (W) x 22.5cm (D) x 7cm (H)
23” (W) x 8.9” (D) x 2.8” (H)
Weight:
Nett: 1.3kg Gross: 1.9kg
Nett: 2.9lbs Gross: 4.2lbs
Audio & Video Interfaces - Audio Distribution Amplifiers
RB-DA6P 6 Way Stereo Distribution Amplifier
With Phoenix Connectors
The RB-DA6P is a version of the RB-DA6
audio distribution amplifier which uses
Neutrik Phoenix style connectors instead
of XLR inputs and outputs. Each unit is
shipped with the mating connectors so that
wires can be simply terminated using a
small flat-blade screwdriver.
The inputs and outputs are electronically
balanced and can be wired unbalanced.
Each output is individually buffered so that
a short circuit on one output won’t affect
the others.
The left and right input gain controls
(normalising) are pre-set potentiometers
accessible through the front panel.
The output gain may be varied from -8dB
to 18dB which is useful for normalising
consumer and professional signals to give
outputs of -15dBu and 0dBu respectively.
Category: Audio Distribution Amplifiers.
Product Function: Audio distribution to
multiple destinations.
Typical Applications: Distribute line level
signals to multiple reporters in a broadcast
centre, distribute main programme output
from a mixer to other studios.
Features: Input level adjustment, mono/
stereo switch, Neutrik Phoenix style audio
connectors.
Specification For RB-DA6P
Connections
Audio Specification
Inputs:
Maximum
Input Level:
+28dBu
2 x Phoenix 3 pin (balanced, can
be unbalanced)
Outputs:
Maximum
Output Level:
+28dBu
12 x Phoenix 3 pin (balanced, can
be unbalanced)
Mains Input:
&ƌĞƋƵĞŶĐLJ
Response:
ϮϬ,njƚŽϮϬŬ,njцϬ͘ϭĚ;ϲϬϬɏůŽĂĚ͕
ref 1kHz)
Filtered IEC, switchable 110-120V, or
220-240V, fused, 6W max
Fuse Rating:
Gain Range:
Adjustable 8dB loss to 18dB gain
(L & R adjust)
Anti-surge fuse 100mA 20 x 5mm
(230VAC)
Anti-surge fuse 250mA 20 x 5mm
(115VAC)
Common
>66dB typically
Mode Rejection:
/ŶƉƵƚŝŵƉĞĚĂŶĐĞ͗ хϮϬŬɏďƌŝĚŐŝŶŐ
Equipment Type
RB-DA6P:
6 way stereo distribution amplifier with
Phoenix connectors
KƵƚƉƵƚ
impedance:
фϱϬɏ
Physical Specification
Distortion:
0.01% THD @ 1kHz, ref +8dBu output
Dimensions
(Raw):
48cm (W) x 10.8cm (D) x 4.2cm (H) (1U)
19” (W) x 4.3” (D) x 1.7” (H) (1U)
Noise:
-100dB unity gain, ref +8dBu output
Dimensions
(Boxed):
58.5cm (W) x 22.5cm (D) x 7cm (H)
23” (W) x 8.9” (D) x 2.8” (H)
Weight:
Nett: 1.3kg Gross: 1.9kg
Nett: 2.9lbs Gross: 4.2lbs
97
Audio & Video Interfaces - Audio Distribution Amplifiers
RB-DA6R 6 Way Stereo Distribution Amplifier With RJ45 Connectors
Category: Audio Distribution Amplifier.
Product Function: Audio distribution to
multiple destinations over CAT5 cabling.
Typical Applications: Distribute line
levels signals to multiple reporters in
a broadcast centre, distribute main
programme output from a mixer to
other studios using a pre-wired CAT5
infrastructure, create multiple feeds of a
recorded signal.
Features:
• Input level adjustment (RB-DA6R).
• Individual output gain adjustment for
each channel (RB-DA6RG).
• Mono/stereo switch.
• Multiple input connections.
• Fast wiring using RJ45 connectors and
Cat 5 cabling
The RB-DA6R is a 1U rack-mount high
performance 6 way stereo distribution
amplifier for splitting a source into a number
of different outputs. The amplifier provides
multiple balanced audio outputs using RJ45
connectors wired to the StudioHub+TM
standard *.
By using RJ45 outputs, it allows simpler CAT5
cabling to used to connect the amplifier to
other products.
The RB-DA6R has one stereo input which
is switchable via a rear panel push switch
between paralleled balanced inputs (2 x
XLR sockets or 1 x stereo input on RJ45) and
unbalanced inputs (1 pair of stereo phono
sockets).
* StudioHub+TM is a registered trademark of Radio Systems Inc
98
The unit has 6 stereo outputs on 6 x RJ45
connectors. The unit can also be configured
so that 1 mono input can be distributed
to 12 outputs by use of a switch which is
recessed on the front panel to prevent it
being accidentally knocked.
The inputs and outputs are electronically
balanced and can be wired unbalanced.
Each output is individually buffered so that a
short circuit on one output won’t affect the
others. The left and right input gain controls
(normalising) are pre-set potentiometers
accessible through the front panel.
The output gain may be varied from -8dB
to 18dB which is useful for normalising
consumer and professional signals to give
outputs of -15dBu and 0dBu respectively.
Audio & Video Interfaces - Audio Distribution Amplifiers
RB-DA6RG 6 Way Stereo Distribution Amplifier With RJ45 Connectors & Output Gain Control
The RB-DA6RG is a 1U rack-mount high
performance 6 way stereo distribution
amplifier for splitting a source into a
number of different outputs.
It is identical to the RB-DA6R with
the addition of individual output gain
adjustment, instead of global stereo gain
adjustment.
Specification For RB-DA6R & RB-DA6RG
Audio Specification
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Equipment Type
Maximum
Input Level:
+28dBu
Distortion:
0.01% THD @ 1kHz, ref +8dBu output
RB-DA6R:
Noise:
-100dB unity gain, ref +8dBu output
Maximum
Output Level:
+28dBu
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Response:
ϮϬ,njƚŽϮϬŬ,njцϬ͘ϭĚ;ϲϬϬɏůŽĂĚ͕
ref 1kHz)
Gain Range
(RB-DA6R):
Gain Range
(RB-DA6RG):
Adjustable 8dB loss to 18dB gain
(L & R adjust)
Adjustable 8dB loss to 18dB gain
(12 adjustable pots)
Common
>66dB typically
Mode Rejection:
* StudioHub+TM is a registered trademark of
Radio Systems Inc
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хϭϬŬɏ;ƵŶďĂůĂŶĐĞĚͿ
Connections
Inputs:
2 x XLR 3 pin female, 1 x RJ45 socket
balanced)
2 x phono sockets (unbalanced)
Outputs:
6 x RJ45 sockets
(all balanced, can be unbalanced)
Mains Input:
Filtered IEC, switchable 110-120V, or
220-240V, fused, 6W max
Fuse Rating:
Anti-surge fuse 100mA 20 x 5mm (230VAC)
Anti-surge fuse 250mA 20 x 5mm (115VAC)
RB-DA6RG:
6 way stereo distribution amplifier with
RJ45 connectors
6 way stereo distribution amplifier with
RJ45 connectors & output gain control
Physical Specification
Dimensions
(Raw):
48cm (W) x 10.8cm (D) x 4.2cm (H) (1U)
19” (W) x 4.3” (D) x 1.7” (H) (1U)
Dimensions
(Boxed):
58.5cm (W) x 22.5cm (D) x 7cm (H)
23” (W) x 8.9” (D) x 2.8” (H)
Weight:
Nett: 1.3kg Gross: 1.9kg
Nett: 2.9lbs Gross: 4.2lbs
Accessories
RB-RK3:
1U Rear panel rack kit for large Redboxes
99
Audio & Video Interfaces - Audio Distribution Amplifiers
RB-DA4x5 4 Input, 4 x 5 Output Distribution Amplifier/Mixer
Category: Audio Distribution Amplifier.
Product Function: Mixing & distribution
of 4 mono analogue audio inputs to 4
sets of 5 mono outputs.
100
Typical Applications: Distribute line
level signals to multiple reporters in a
broadcast centre, at a radio station to
distribute a PC automation output to
up to 5 studios, to distribute a main
programme output from a mixer to other
studios, to create multiple mix minus
feeds or to create multiple stereo to
mono outputs.
Features:
• 4 input, 20 output distribution amplifier.
• Audio prescence LED’s for inputs or
outputs.
• Balanced I/O that can be wired
unbalanced.
• Outputs a 1kHz test tone e.g to check
transmission lines.
• Front panel cover to secure settings.
• Input and output gain adjustment.
The RB-DA4x5 is a 1U rack-mount
combined distribution amplifier and mixer.
It has 4 mono analogue audio inputs on
female XLR and 4 groups of 5 outputs on 15
way D-type connectors.
Each output group has a five way frontpanel DIP switch assigned to it which is
used to select the input(s) to send to the
output group. This enables each of the four
inputs, or a 1kHz 0dBu tone, to be mixed to
the output group.
Audio & Video Interfaces - Audio Distribution Amplifiers
The inputs and outputs are electronically
balanced and can be wired unbalanced.
Each input has adjustable gain using a preset potentiometer, providing a gain range of
-8dB to +18dB.
Each output is individually buffered so that
a short circuit on one output won’t affect
the others. The output gain of each group
can similarly be adjusted between -8dB and
+18dB using pre-set potentiometers.
The DIP switch settings and gain controls
are recessed beneath a front-mounting
screw-on cover so that settings can not be
accidently altered, for secure applications.
Four bright front-panel signal present LEDs
show the levels of either the inputs or
the output groups by pressing the AUDIO
PRESENCE button. The LEDs will show green
illumination from -12db through to 0db,
amber from 0db through to +6db and red
for inputs and outputs at +6db and over.
Balanced XLR
Input 1
Gain
Balanced XLR
Input 2
Gain
Output 1
Switch Bank
Balanced XLR
Input 3
Gain
Meter 1
Output
Balanced XLR
Input 4
Audio Specification
Maximum
Input Level:
+28dBu
Maximum
Output Level:
+28dBu
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Response:
ϮϬ,njƚŽϮϬŬ,njцϬ͘ϭĚ;ϲϬϬɏůŽĂĚ͕
ref 1kHz)
Input Gain
Range:
Adjustable 8dB loss to 18dB gain
(channel 1-4 adjust)
Output Gain
Range:
Adjustable 8dB loss to 18dB gain
(group 1-4 adjust)
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impedance:
хϮϬŬɏďƌŝĚŐŝŶŐ
KƵƚƉƵƚ
impedance:
фϱϬɏ
Distortion:
0.01% THD @ 1kHz, ref +8dBu output
Noise:
-100dB unity gain, ref +8dBu output
Common Mode
Rejection:
>66dB typically
Connections
Output 1
Switch Bank
Gain
Specification For RB-DA4X5
Gain
Meter 2
Output
Gain
Inputs
(4 x Mono):
4 x XLR 3 pin female (balanced, can be
unbalanced)
Outputs
(20 x Mono):
4 x 15 way D-type male plug (balanced,
can be unbalanced)
Mains Input:
Filtered IEC, switchable 110-120V, or
220-240V, fused, 9W max
Fuse Rating:
Anti-surge fuse 100mA 20 x 5mm
(230VAC)
Anti-surge fuse 250mA 20 x 5mm
(115VAC)
Reference
Tone
Equipment Type
Meter In 1
RB-DA4x5:
Meter In 2
Meter Out 1
Meter Out 2
Meter 1
Meter 2
Meter In 3
Meter In 4
Meter Out 3
Meter Out 4
Meter 3
Physical Specification
Output 1
Switch Bank
Dimensions
Output 1
Switch Bank
Gain
Meter 3
Output
Meter 4
4 Input, 4 x 5 output distribution
amplifier/mixer
Gain
Meter 4
Output
(Raw):
48cm (W) x 15.8cm (D*) x 4.2cm
(H) (1U)
19” (W) x 6.2” (D*) x 1.7 (H) (1U)
Dimensions
(Boxed):
59cm (W) x 27.5cm (D*) x 11cm (H)
23.2” (W) x 10.8” (D*) x 4.3” (H)
Weight:
Nett: 1.6kg Gross: 2.2kg
Nett: 3.5lbs Gross: 4.8lbs
* Note that this product is deeper than standard Redboxes
RB-DA4x5 Block Diagram.
101
Audio & Video Interfaces - Audio Distribution Amplifiers
RB-DDA6A 6 Way Stereo AES/EBU Digital Distribution Amplifier
Category: Audio Distribution Amplifiers.
Product Function: 1 input, 6 output AES/
EBU digital audio distribution amplifier.
Typical Applications: Distribution of
audio from a digital mixing desk to
multiple digital recorders, or feeding
multiple studios with an output from a
flashcard player.
Features: Neutrik XLR input and output
connectors, sample rates up to 96kHz.
102
The RB-DDA6A digital distribution amplifier
is used for distributing digital audio data in
AES/EBU format, repeating both the audio
data and the status information of the input
whilst re-normalising to standard digital
audio levels.
24 bit. So, it can be used for standard CD
signal distribution at 16 bit 44.1kHz, as well
as for high quality 24 bit 96kHz recording.
Fuse Rating:
Specification For RB-DDA6A
RB-DDA6A:
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Impedance:
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It has a single XLR-3 female AES/EBU audio
input which is distributed to 6 XLR-3 male
AES/EBU outputs.
KƵƚƉƵƚ
Impedance:
ϭϭϬɏцϮϬйďĂůĂŶĐĞĚ
Sample
Freq Range:
30-100kHz (i.e. including 32kHz,
44.1kHz, 48kHz, 64kHz, 88.2kHz
& 96kHz)
Applications include distributing audio from
a digital mixing desk to multiple digital
recorders, or feeding multiple studios with an
output from a flashcard player.
Signal Level:
2V/7V peak to peak min/max
It can accept input sample rates in the range
of 30kHz - 100kHz, and bit rates of 16, 20 and
Audio Specification
Connections
Input:
1 x AES/EBU XLR 3 pin female
(Balanced)
Outputs:
6 x AES/EBU XLR 3 pin male (Balanced)
Mains Input:
Filtered IEC, switchable 110-120V, or
220-240V, fused, 6W max
Anti-surge fuse 100mA 20 x 5mm
(230VAC)
Anti-surge fuse 250mA 20 x 5mm
(115VAC)
Equipment Type
6 way stereo AES/EBU digital
distribution amplifier
Physical Specification
Dimensions
(Raw):
28cm (W) x 10.8cm (D) x 4.2cm (H) (1U)
11” (W) x 4.3” (D) x 1.7” (H) (1U)
Dimensions
(Boxed):
36cm (W) x 20.5cm (D) x 6cm (H)
14.2” (W) x 8” (D) x 2.4” (H)
Weight :
Nett: 0.95kg Gross: 1.4kg
Nett: 2.1lbs Gross: 3.2lbs
Audio & Video Interfaces - Audio Distribution Amplifiers
Specification For RB-DDA6A-2P
Audio Specification
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RB-DDA6A-2P 6 Way Stereo AES/EBU Digital Distribution Amplifier
with Dual Power Supplies
The RB-DDA6A-2P digital distribution
amplifier is used for distributing digital
audio data in AES/EBU format, repeating
both the audio data and the status
information of the input whilst renormalising to standard digital audio levels.
It has a single XLR-3 female AES/EBU audio
input which is distributed to 6 XLR-3 male
AES/EBU outputs.
Applications include distributing audio from
a digital mixing desk to multiple digital
recorders, or feeding multiple studios with
an output from a flashcard player.
It can accept input sample rates in the range
of 30kHz - 100kHz, and bit rates of 16, 20
and 24 bit. So, it can be used for standard
CD signal distribution at 16 bit 44.1kHz,
as well as for high quality 24 bit 96kHz
recording.
The unit provides redundancy protection
by using two power supply units that can
be supplied from separate mains feeds. The
unit automatically load shares between the
two supplies so that if either supply should
fail the unit will continue to work correctly.
If both supplies fail the input is connected
directly to output 1.
The condition of the two supplies is
indicated on the front panel by two red LEDs
which illuminate to indicate the correct
function of the supply. On the
alarm output connector there is a
changeover relay to indicate the status
of each supply. This is normally closed to
indicate power fail and once the supply
is working correctly the relay activates to
make a normally open contact.
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KƵƚƉƵƚ/ŵƉĞĚĂŶĐĞ͗
ϭϭϬɏцϮϬйďĂůĂŶĐĞĚ
Sample Freq Range:
30-100kHz (i.e. including
32kHz, 44.1kHz, 48kHz, 64kHz,
88.2kHz and 96kHz
Signal Level:
2V/7V peak to peak min/max
Connections
Audio Inputs:
1 x AES/EBU XLR 3 pin female
(balanced)
Audio Outputs:
6 x AES/EBU XLR 3 pin male
(balanced)
Alarm Outouts:
2 x Relay output indicators on
1 x 9 pin D-Type socket
Mains Input:
2 x Filtered IEC, 85-264V
switchable, fused, 6W max
Fuse Rating:
2 x Anti-surge fuse 1A 20
x 5mm
Equipment Type
RB-DDA6A-2P:
6 way AES/EBU stereo digital
distribution amplifier with
dual power supplies
Physical Specification
Dimensions (Raw):
48cm (W) x 10.8cm (D)
x 4.2cm (H) (1U)
19” (W) x 4.3” (D) x 1.7” (H)(1U)
Dimensions (Boxed):
58.5cm (W) x 22.5cm (D)
x 7cm (H)
23” (W) x 8.9” (D) x 2.8” (H)
Weight:
Nett: 1.5kg Gross: 2.0kg
Nett: 3.3lbs Gross: 4.4lbs
103
Audio & Video Interfaces - Audio Distribution Amplifiers
RB-DDA6S 6 Way Stereo S/PDIF Digital Distribution Amplifier
Category: Audio Distribution
Amplifiers.
Product Function: 1 input, 6 output
S/PDIF digital audio distribution
amplifier.
Typical Applications: Audio distribution
at 16 bit 44.1kHz from a consumer CD
player to multiple digital recorders,
distribution of high quality 24 bit 96kHz
signals from digital mixing desks to
recorders and connection of the output
of, say, a BluRay player to multiple
studios.
Features: Fully buffered outputs,
sample rates up to 96kHz.
104
The RB-DDA6S digital distribution amplifier
is similar to the RB-DDA6A except that it is
used for distributing digital audio data in S/
PDIF format.
It has a single S/PDIF audio input which is
distributed to 6 x S/PDIF audio outputs at
the same level and condition as the input
signal. It can accept input sample rates in
the range of 30kHz - 100kHz, and bit rates of
16, 20 and 24 bit.
Uses include audio distribution at 16 bit
44.1kHz from a consumer CD player to
multiple digital recorders, distribution of
high quality 24 bit 96kHz signals from digital
mixing desks to recorders and connection
of the output of, say, a BluRay player to
multiple studios.
Specification For RB-DDA6S
Audio Specification
Equipment Type
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Impedance:
ϳϱɏцϱйƵŶďĂůĂŶĐĞĚ
RB-DDA6S:
KƵƚƉƵƚ
Impedance:
ϳϱɏцϱйƵŶďĂůĂŶĐĞĚ
Physical Specification
Sample
Freq. Range:
30-100kHz (i.e. including 32kHz,
44.1kHz, 48kHz, 64kHz, 88.2kHz &
96kHz)
Signal Level:
Balanced min 0.5V ±20% peak to peak
Connections
Input:
1 x S/PDIF RCA phono (unbalanced)
Outputs:
6 x S/PDIF RCA phono (unbalanced)
Mains Input:
Filtered IEC, switchable 110-120V, or
220-240V, fused, 6W max
Fuse Rating:
Anti-surge fuse 100mA 20 x 5mm
(230VAC)
Anti-surge fuse 250mA 20 x 5mm
(115VAC)
6 way stereo S/PDIF digital distribution
amplifier
Dimensions
(Raw):
28cm (W) x 10.8cm (D) x 4.2cm (H) (1U)
11” (W) x 4.3” (D) x 1.7” (H) (1U)
Dimensions
(Boxed):
36cm (W) x 20.5cm (D) x 6cm (H)
14.2” (W) x 8” (D) x 2.4” (H)
Weight :
Nett: 0.9kg Gross: 1.35kg
Nett: 2lbs Gross: 3lbs
Audio & Video Interfaces - Audio Distribution Amplifiers
RB-DDA6W 6 Way Word Clock Distribution Amplifier
The RB-DDA6W word clock distribution
amplifier distributes a word clock BNC
input signal to 6 word clock BNC outputs
re-conditioned. It is used to distribute
reference clocks for digital audio systems.
Specification For RB-DDA6W
Audio Specification
Equipment Type
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Impedance:
ϱϬɏ
RB-DDA6W:
KƵƚƉƵƚ
Impedance:
фϱϬɏ
Dimensions
(Raw):
28cm (W) x 10.8cm (D) x 4.2cm (H) (1U)
11” (W) x 4.3” (D) x 1.7” (H) (1U)
1 x BNC female
Dimensions:
(Boxed):
36cm (W) x 20.5cm (D) x 6cm (H)
14.2” (W) x 8” (D) x 2.4” (H)
Outputs:
6 x BNC female
Weight:
Mains Input:
Filtered IEC, switchable 110-120V, or
220-240V, fused, 6W max
Nett: 0.9kg Gross: 1.35kg
Nett: 2lbs
Gross: 3lbs
Fuse Rating:
Anti-surge fuse 100mA 20 x 5mm
(230VAC)
Anti-surge fuse 250mA 20 x 5mm
(115VAC)
Connections
Category: Audio Distribution Amplifiers.
Product Function: 1 input, 6 output TTL
word clock distribution amplifier.
Typical Applications: Distribution
of word clock signal from a master
generator to other studio equipment to
keep them synchronised.
Features: Fully buffered outputs.
It has a single female BNC audio input which
is distributed to 6 female BNC outputs.
The units primary application is to distribute
a master TTL word clock source to multiple
pieces of equipment that need to be
synchronised from the master.
6 way word clock distribution amplifier
Physical Specification
Input:
105
Audio & Video Interfaces - Audio Distribution Amplifiers
RB-DDA6W-2P 6 Way Word Clock Distribution Amplifier with Dual Power Supplies
Category: Audio Distribution Amplifiers.
Product Function: 1 input, 6 output TTL
The RB-DDA6W-2P word clock distribution
amplifier distributes a word clock BNC
input signal to 6 word clock BNC outputs
re-conditioned. It is used to distribute
reference clocks for digital audio systems.
word clock distribution amplifier.
Typical Applications: Distribution of word
clock signal from a master generator to
other studio equipment to keep them
synchronised.
Features: Fully buffered outputs, dual IEC
mains inputs.
It has a single female BNC audio input which
is distributed to 6 female BNC outputs. The
units primary application is to distribute a
master TTL word clock source to multiple
pieces of equipment that need to be
synchronised from the master.
Specification For RB-DDA6W-2P
The unit provides redundancy protection
Audio Specification
by using two power supplies that can be
Input impedance: 50O
supplied from separate mains feeds. The unit
Output impedance: <50O
automatically load shares between the two
Connections
supplies so that if either supply should fail, it
Audio Inputs:
1 x BNC female
will continue to work correctly. If both supplies Audio Outputs:
6 x BNC female
2 x Relay output indicators on
fail, the input is connected directly to output 1. Alarm Outputs:
1 x 9 pin D-Type socket
The condition of the two supplies is indicated
on the front panel by two red LEDs which
illuminate to indicate the correct function of
the supply. On the alarm output connector
there is a changeover relay to indicate the
status of each supply. This is normally closed
to indicate power fail and once the supply is
working correctly the relay activates to make a
normally open contact.
Mains Input:
2 x Filtered IEC, 85-264V switchable,
fused, 6W max
Fuse Rating:
2 x Anti-surge fuse 1A 20 x 5mm
Equipment Type
RB-DDA6W-2P:
Physical Specification
Dimensions (Raw):
48cm (W) x 10.8cm (D) x 4.2cm (H)
(1U)
19” (W) x 4.3” (D) x 1.7” (H) (1U)
Dimensions (Boxed): 58.5cm (W) x 22.5cm (D) x 7cm (H)
23” (W) x 8.9” (D) x 2.8” (H)
Weight:
106
6 way word clock distribution
amplifier with dual power supplies
Nett: 1.5kg Gross: 2.0kg
Nett: 3.3lbs Gross: 4.4lbs
Audio & Video Interfaces - Audio Distribution Amplifiers
RB-DDA6A3 6 Way Stereo AES3ID Digital Audio Distribution Amplifier
Category: Audio Distribution Amplifiers.
Product Function: 1 input, 6 output
AES3id digital audio distribution
amplifier.
Typical Applications: Distribution of
audio from a BluRay player to recorders,
or feeding multiple studios with an
output from a master clock.
Features: Fully buffered outputs, sample
rates up to 96kHz.
The RB-DDA6A3 digital distribution
amplifier is used for distributing AES3id
digital audio, repeating both the audio data
and the status information of the input
whilst re-normalising to standard digital
audio levels. It has a single BNC AES3id
audio input which is distributed to 6 BNC
AES3id audio outputs.
Specification For RB-DDA6A3
Technical Specification
Audio Specification
Input impedance:
Equipment Type
RB-DDA6A3:
75:
KƵƚƉƵƚŝŵƉĞĚĂŶĐĞ͗
ϳϱɏ
Physical Specification
Sample Freq Range:
30-100kHz (i.e. including 32kHz,
44.1kHz, 48kHz, 64kHz, 88.2kHz
& 96kHz)
Dimensions (Raw):
28cm (W) x 10.8cm (D) x 4.2cm
(H) (1U)
11” (W) x 4.3” (D) x 1.7” (H) (1U)
Signal Level:
1V ±20% peak to peak (output
1.17V to drive long cable lengths)
Dimensions (Boxed):
36cm (W) x 20.5cm (D) x 6cm (H)
14.2” (W) x 8” (D) x 2.4” (H)
Weight:
Nett: 0.95kg Gross: 1.4kg
Nett: 2.1lbs Gross: 3.2lbs
Connections
Applications include distributing audio from
a BluRay player to recorders, or feeding
multiple studios with an output from a
master clock.
It can accept input sample rates in the range
of 30kHz - 100kHz, and bit rates of 16, 20
and 24 bit.
6 way stereo AES3ID digital audio
distribution amplifier
Inputs:
1 x BNC female
Outputs:
6 x BNC female
Mains Input:
Filtered IEC, switchable 110-120V,
or 220-240V, fused, 6W max
Fuse Rating:
Anti-surge fuse 100mA
20 x 5mm (230VAC)
Anti-surge fuse 250mA
20 x 5mm (115VAC)
107
Audio & Video Interfaces - Audio Distribution Amplifiers
RB-DDA22 Digital Audio Distribution Amplifier With Multiple Outputs
The RB-DDA22 is a multiple input and
multiple output digital audio distribution
amplifier designed to accept one of 5
different digital inputs and output to
22 digital audio outputs of 5 different
connector types.
It is used for distributing digital audio
data in many formats, repeating both the
audio data and the status information of
the input whilst re-clocking the data and
re-normalising to the appropriate standard
digital audio levels.
Input Signals:
• 1 x XLR 3 female AES/EBU digital audio
input
• 1 x BNC AES-3id digital audio input
• 1 x RJ45 AES/EBU digital audio input
• 1 x RCA Phono S/PDIF input
• 1 x TOSLink digital audio optical input
Output Signals:
• 6 x XLR-3 male AES/EBU digital audio
outputs
• 6 x BNC AES-3id digital audio outputs
• 6 x RJ45 AES/EBU digital audio outputs
• 2 x RCA Phono S/PDIF outputs
• 2 x TOSLink optical outputs
The RJ-45 connectors are wired to be
StudioHub+ compatible, a format defined by
Radio Systems Inc.
Applications include distributing audio from
a digital mixing desk to multiple digital
Category: Audio Distribution Amplifiers.
Product Function: Multi input, multi
output format, digital audio distribution
amplifier.
108
Typical Applications: Distribution of an
AES/EBU output to many recorders with
different digital inputs. General tool for
studio digital audio distribution.
Features:
• Multiple input formats.
• Multiple output formats.
• Sample rates up to 192kHz.
• Front panel input selection.
recorders, or feeding multiple studios with
an output from a USB audio player.
or above 192kHz then the appropriate LED
will flash.
It can accept input sample rates in the range
of 30kHz - 200kHz, and bit rates of 16, 20
and 24 bit. So, it can be used for standard
CD signal distribution at 16 bit 44.1kHz,
as well as for high quality 24 bit 96kHz or
192kHz recording.
Pressing the INPUT SELECT central button
illuminates it red and allows you to select
the required input. Pressing the button will
step the selected input to the next input
selection including auto mode.
The front panel has an INPUT SELECT button
and 7 indicator LEDS. The button is used to
select the input to be used and shows the
mode of operation of the unit. The LEDs
will indicate, depending on the mode of
operation, the input selected, valid signal
presence, or the operational frequency.
In ‘operating mode’, the central button LED
is unlit. The unit selects this mode when
a valid input signal has been present and
no button presses have been made for
8 seconds. The LEDs around the button
illuminate amber to directly indicate the
following frequencies 32, 44.1, 48, 88.2,
96, 176.4 & 192 kHz. If the input is at a
non-standard frequency then the unit will
indicate both the frequencies either side
of the incoming frequency – e.g. if the
input signal has a frequency of 64kHz then
both the 48kHz and 88.2kHz LEDs will be
illuminated. If the frequency is below 32kHz
In Input Select Mode the LEDs have the
following states:
Off – Signal not selected & no valid signal.
Green – Signal not selected & valid signal.
Red – Signal selected & no valid signal.
Amber – Signal selected & valid signal.
present or Auto Mode Selected.
Note: Input Select mode is chosen
automatically when the selected input is not
present.
In this special input select mode the unit
will hunt through all the inputs until it finds
a valid signal. If multiple signals are present
you can force selection of the next input by
holding the input select button down for 2
seconds, when the next valid input will be
selected.
Audio & Video Interfaces - Audio Distribution Amplifiers
192
Specification For RB-DDA22
Audio Specification
Connections
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Digital Inputs:
Signal Level:
Balanced: 3V/10V peak to peak min/max
Unbalanced: Min 0.5V±20% peak to peak
Sample
Frequencies:
32, 44.1, 48, 88.2, 96,176.4 or 192kHz
Bit Depth:
Up to and including 24 bit
Digital Outputs:
Front Panel Operational Controls & Indicators
Equipment Type
1 x AES/EBU XLR 3 pin socket
1 x AES/EBU BNC
1 x AES/EBU RJ45 socket (StudioHub+TM)
1 x S/PDIF RCA phono socket
1 x TOSLink optical input
6 x AES/EBU XLR 3 pin plug
6 x AES/EBU BNC
6 x AES/EBU RJ45 socket (StudioHub+TM)
2 x S/PDIF RCA phono socket
2 x TOSLink optical output
Digital
Input Select:
AES/EBU (XLR), AES/EBU (BNC), AES/EBU
(RJ45), S/PDIF or TOSlink optical
Mains Input:
Universal filtered IEC, continuously
rated [email protected] 63Hz, max 10W
Indicators:
Input presence indicators via tricolour
LEDS around the input select button
Fuse Rating:
Anti-surge fuse 1A 20 x 5mm
RB-DDA22:
Digital audio distribution amplifier with
multiple outputs
Physical Specification
Dimensions
(Raw):
48cm (W) x 10.8cm (D) x 4.2cm (H) (1U)
19” (W) x 4.3” (D) x 1.7” (H) (1U)
Dimensions
(Boxed):
58.5cm (W) x 22.5cm (D) x 7cm (H)
23” (W) x 8.9” (D) x 2.8” (H)
Weight:
Nett: 1.4kg Gross: 2.0kg
Nett: 3.0lbs Gross: 4.4lbs
Accessories
RB-RK3:
1U Rear panel rack kit for large
Redboxes
* StudioHub+TM is a registered trademark of Radio
Systems Inc
109
Audio & Video Interfaces - Audio Distribution Amplifiers
RB-DA24MD 24 Way Mono Audio Distribution Amplifier
Category: Audio Distribution Amplifiers.
Product Function: Mixing & distribution
of 2 mono analogue audio inputs to 24
mono outputs.
Typical Applications: Distribute mono
emergency audio signals to multiple
powered monitors, to distribute a mono
110
source (e.g. mic input converted to line) to
many outputs, e.g. at a news conference.
Features:
• 2 input, 24 output distribution amplifier.
• Input gain adjustment.
• Balanced I/O that can be wired
unbalanced.
• Use as a stereo pair, or 2 x mono inputs.
• High pass roll-off filter.
• Output protection against PoE &
phantom power.
• Mixed output level adjustment.
The RB-DA24MD is a 1U rack mount high
performance 24 way audio distribution
amplifier. It has 2 inputs which can be each
individually routed to 12 outputs, or mixed
and routed to all 24 outputs.
The inputs can be configured as either dual
mono, input 1 routed to outputs 1-12 and
input 2 routed to outputs 13-24, or mixed-
mono, inputs 1 and 2 mixed at a pre-set
level and routed to all 24 outputs.
When set to mixed-mono operation there
are 3 different output mix level modes,
enabled by altering internal jumpers,
providing -6dB, -3dB or 0dB adjustment of
the output.
Audio & Video Interfaces - Audio Distribution Amplifiers
The XLR inputs and D-type outputs are
electronically balanced and can be wired
unbalanced. Each output is individually
buffered so that a short circuit on one won’t
affect the others.
Each output is also protected against
connection to both POE (power over
Ethernet) and phantom power circuits.
The RB-DA24MD has master gain controls
for both input 1 and input 2 which are preset potentiometers accessible through the
rear panel. These controls allow the gain
to be adjusted from -15dB to +15dB, useful
for normalising consumer to professional
signals and vice versa.
A 125Hz 6dB per octave roll off filter is
activated by a push switch on the rear panel.
When selected the filter is the applied to
both inputs.
An LED power indicator on the front panel
shows that the unit is powered.
Specification For RB-DA24MD
Connections
Audio Specification
Inputs:
2 x XLR 3 pin female (balanced)
Outputs:
2 x 25 way D-type plug (balanced,
can be unbalanced)
Mains Input:
Filtered IEC, switchable 110-120V,
or 220-240V, fused, 12W maximum
Fuse Rating:
Anti-surge fuse 100mA
20 x 5mm (230VAC)
Anti-surge fuse 250mA
20 x 5mm (115VAC)
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фϱϬɏ
Maximum Input
Level:
+28dBu
Maximum Output
Level:
+28dBu
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Response:
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load, ref 1kHz)
Gain Range:
Adjustable -15dB loss to
+15dB gain
Common Mode
Rejection:
>66dB typically
LF Roll Off Filter:
125Hz at 6dB/octave
Distortion:
0.01%THD @ 1kHz, ref
+8dBu output
Noise:
-100dB unity gain, ref
+8dBu output
Equipment Type
RB-DA24MD:
24 way mono distribution amplifier
Physical Specification
Dimensions (Raw):
48cm (W) x 15.8cm (D) x 4.2 (H) (1U)
19” (W) x 6.2” (D) x 1.7” (H) (1U)
Dimensions
(Boxed):
58.5cm (W) x 27.4cm (D) x 10.8cm (H)
23” (W) x 10.9” (D) x 4.3” (H)
Weight:
Nett: 1.6kg Gross: 2.2kg
Nett: 3.5lbs Gross: 4.8lbs
111
Audio & Video Interfaces - Audio Distribution Amplifiers
RB-AES4B3 Quad 3 Way Passive AES3ID Splitter BNC Connectors
The RB-AES4B3 is a passive, quad “oneto-three” splitter housed in a 19” rack.
Each bank is designed to split a single
AES3ID digital audio source to up to three
destinations, using female BNC connectors.
Category: Audio Distribution Amplifiers.
Product Function: 4 sets of: 1 input, 3
output AES3id distribution amplifier.
Typical Applications: Distribution of
AES3id source to 3 mixing consoles.
Features: No power needed, can apply
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112
Particularly useful in a video production
and broadcast environment, the RB-AES4B3
splits the input signal through high quality
transformers.
Specification For RB-AES4B3
Audio Specification
Cable Drive Capability
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cable at sample rates up to and including 96kHz.
Connections
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to unconnected outputs to maintain
optimum carrier parameters.
The RB-AES4B3 requires no power to
operate, ensuring your audio remains
connected from source to destination(s)
without interruption from power failures.
Inputs:
4 x BNC female
Outputs:
12 x BNC female
Equipment Type
RB-AES4B3
Quad 3 way passive AES3ID splitter
with BNC connectors.
Physical Specification
Dimensions
(Raw):
48cm (W) x 10.8cm (D) x 4.2cm (H) (1U)
19” (W) x 4.3” (D) x 1.7” (H) (1U)
Dimensions
(Boxed):
58.5cm (W) x 22.5cm (D) x 7cm (H)
23” (W) x 8.9” (D) x 2.8” (H)
Weight:
Nett: 1.0kg Gross: 1.4kg
Nett: 2.25lbs Gross: 3.0lbs
Audio & Video Interfaces - Audio Distribution Amplifiers
RB-AES4X3 Quad 3 Way AES/EBU Splitter
XLR Connectors
The RB-AES4X3 is a passive, quad “oneto-three” splitter housed in a 19” rack.
Each bank is designed to split a single
AES3 digital audio source to up to three
destinations, using Neutrik XLR connectors.
The RB-AES4X3 provides solutions to a
range of digital signal distribution problems,
where correct termination is essential to
maintain signal integrity. The input signal is
split through high quality transformers, and
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to unconnected outputs.
The RB-AES4X3 requires no power to
operate, ensuring your audio remains
connected from source to destination(s)
without interruption from power failures.
Category: Audio Distribution Amplifiers.
Product Function: 4 sets of: 1 input, 3
Specification For RB-AES4X3
Audio Specification
Cable Drive Capability
The table below sets out the minimum signal amplitude
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cable, based on the sample rate of the digital audio:
Sample Rate
Minimum Signal Amplitude
32kHz
*2Vpk-pk
44.1kHz
*2Vpk-pk
48kHz
*2Vpk-pk
output AES3 distribution amplifier.
Typical Applications: Distribution of
AES3 source to 3 mixing consoles.
Features: No power needed, can apply
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outputs, transformer based solution
provides reliability.
The table below sets out the minimum signal amplitude
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cable, based on the sample rate of the digital audio:
Sample Rate
Minimum Signal Amplitude
176.4kHz
3Vpk-pk
192kHz
3Vpk-pk
Connections
Inputs:
4 x XLR 3 pin female
Outputs:
12 x XLR 3 pin male
88.2kHz
5Vpk-pk
Equipment Type
96kHz
5Vpk-pk
RB-AES4X3
* Minimum of 2Vpk-pk is defined by the AES3 format
specification.
Quad 3 way AES/EBU splitter with
XLR connectors
Physical Specification
Dimensions
(Raw):
48cm (W) x 10.8cm (D) x 4.2cm (H)(1U)
19” (W) x 4.3” (D) x 1.7” (H) (1U)
Dimensions
(Boxed):
58.5cm (W) x 22.5cm (D) x 7cm (H)
23” (W) x 8.9” (D) x 2.8” (H)
Weight:
Nett: 1.0kg
Gross: 1.4kg
Nett: 2.25lbs Gross: 3.0lbs
113
Audio & Video Interfaces - Audio Distribution Amplifiers
RB-MS4X3 Quad 3 Way Passive Microphone Splitter
Category: Audio Distribution Amplifiers.
Product Function: To split up to 4
microphone or line level signals to 3
independent outputs each.
Typical Applications:
To share microphones between radio/TV/
production studios.
Features:
• Passive transformer design.
• Ground lift switch.
• External or looped phantom power
inputs.
• Line level -30dB pad.
The RB-MS4X3 is a passive, quad, “oneto-three” splitter housed in a 19” rack.
Each bank is designed to split a single
microphone or line source to up to three
destinations, using Neutrik XLR connectors.
Cable connections are located on the
rear panel, with recessed controls and
indicators available to the user on the front
panel, allowing quick and easy access to
setup parameters.
Wide Signal Range Capability
The RB-MS4X3 uses high quality audio
transformers that are capable of accepting
input levels of up to +18dBu, making the
splitter useful in both microphone and line
level splitting applications. A 30dB pad can
be applied to the input, allowing a line level
signal to be interfaced into a microphone
input with suitable levels and the correct
termination.
Versatile Phantom Powering Options
The RB-MS4X3 offers three methods of
providing phantom power to a microphone
connected to its input:
114
Audio & Video Interfaces - Audio Distribution Amplifiers
• From the external +48VDC power
connector.
• From a microphone amplifier connected
to output 1 (direct).
• From a microphone amplifier connected
to output 2 (phantom loopback).
Controlling which method is used to provide
phantom power is easy with front panel
switches and indicators. It is possible to
concurrently power a microphone using any
two of the above methods without degrading
audio performance, thus providing power
supply redundancy.
Specification For RB-MS4X3
Audio Specification
Test
Conditions
Frequency
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• Ref. -6dBu, 1kHz
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Results
Total Harmonic
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• Ref. +3dBu, 50Hz
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Total Harmonic
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• Ref. 0.1%THD+N, 50Hz
+13dBu
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Total Harmonic
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• Ref. 1%THD+N, 50Hz
+18dBu
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Common Mode
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• Ref. 20kHz
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0.02%
>60dB
Connections
Ground Loop Hum Elimination
Ground loop hum problems can be
quickly eliminated using the front panel
toggle switches to lift pin 1 of the output
connectors (output 2 and 3 only).
Inputs:
4 x XLR 3 pin female
Outputs:
12 x XLR 3 pin male
DC Phantom
Power Input:
+48V DC, 2.5mm socket
Equipment Type
RB-MS4X3
Quad 3 way microphone splitter
Physical Specification
Dimensions (Raw):
48cm (W) x 10.8cm (D) x 4.2cm (H)
(1U)
19” (W) x 4.3” (D) x 1.7” (H) (1U)
Dimensions (Boxed): 58.5cm (W) x 22.5cm (D) x 7cm (H)
23” (W) x 8.9” (D) x 2.8” (H)
Weight:
Nett: 1.5kg Gross: 1.9kg
Nett: 3.3lbs Gross: 4.2lbs
115
Audio & Video Interfaces - Phantom Power Supply
RB-MSP6 6 Way +48V Phantom Power Supply
Category: Phantom Power Supplies.
Product Function: Providing multiple
+48V power outputs.
The RB-MSP6 is a 1U rack-mount phantom
power supply, designed to be used in
conjunction with up to six RB-MS4X3s,
Typical Applications: Connecting to
multiple RB-MSP6 microphone splitters to
ensure phantom power redundancy for
connected microphones.
Features:
• Dual redundant IEC power inlets.
• 6 x locking DC output connectors for
+48V phantom power.
Two mains AC inlets power separate 48V
supplies that are diode fed to six locking DC
connector outputs. Each output connector
is individually protected with a resettable
fuse rated at 50mA, so a fault on one output
will not affect the others.
Should either of the 48V supplies fail, or
if their AC power input is removed, the
corresponding red front panel LED will
providing mains input redundancy to ensure
extinguish, and a relay alarm output is
the target microphones remain powered in
provided on the rear panel D-Type.
the event of a single mains failure.
Specification For RB-MSP6
Mains Inputs:
2 x Filtered IEC, 90V-264V,
47-63Hz, fused, 20W maximum
Fuse Ratings:
2 x Anti-surge fuse 1A 20 x 5mm
Phantom Power
Outputs:
6 x locking DC connectors,
2.5mm pin, 48V DC, 50mA
maximum current, protected by
resettable fuse.
Equipment Type:
RB-MSP6:
6 way +48V phantom power supply
Physical Specification:
Dimensions (Raw):
28cm (W) x 10.8cm (D) x 4.2cm (H) (1U)
11” (W) x 4.3” (D) x 1.7” (H) (1U)
Dimensions (Boxed): 36cm (W) x 20.5cm (D) x 6cm (H)
14.2” (W) x 8” (D) x 2.4” (H)
Weight:
Nett: 1.2kg Gross: 1.5kg
Nett: 2.6lbs Gross: 3.3lbs
Accessories:
RB-RK2:
116
1U rear panel rack kit for small
Redbox range
Audio & Video Interfaces - Headphone Distribution Amplifiers
RB-HD1 Stereo Headphone Amplifier With
VCA Volume Control
The RB-HD1 is a 1U rack-mount stereo
headphone amplifier for driving up to two
pairs of professional stereo headphones
from a single stereo or mono input. One
headphone socket is on the front panel with
one on the rear.
A mono input can be mixed into the main
headphone feed, for example, for mixing
in talkback to the headphones. This has an
input level control via a recessed adjustable
potentiometer. The mono mix input can also
be controlled remotely.
The main stereo input uses electronically
balanced XLR-3 connectors on the rear panel,
which can be wired un-balanced. The output
volume for the headphones can be controlled
either by a pot situated on the front panel
or a VCA signal supplied externally via the
remote connector.
A stereo/mono switch is recessed on the rear
panel to prevent accidental knocking. With
mono selected, audio is sent to both left and
right ear pieces.
A LED power indicator on the front panel
displays the power supply connection.
Category: Headphone Distribution
Amplifiers.
Product Function: Provides a headphone
output from a line level signal.
Specification For RB-HD1
Audio Specification
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Impedance:
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ϲϬϬɏŚĞĂĚƉŚŽŶĞƐ
Mono Mix
Input Gain
Range:
22dBu
Max Input Level:
+28dBu
Volume Control:
-80dB to +11dB gain
Connections
Main Stereo
Inputs:
2 x XLR 3 pin female (balanced, can be
unbalanced)
Mono Mix
Input:
1 x XLR 3 pin female (balanced, can be
unbalanced)
Typical Applications: Headphone
monitoring for voice over booths, adding
headphone output to mixer programme
output.
Features:
• Front and rear headphone outputs.
• VCA remote level control.
• Stereo/mono selection.
• Mono talkback input.
Outputs:
2 x ¼” (6.35mm) A-gauge 3-pole
stereo jack sockets
Remote Control:
9-pin D-type socket
Mains Input:
Filtered IEC, switchable 110-120V, or
220-240V, fused, 9W max
Fuse Rating:
Anti-surge fuse 100mA 20 x 5mm
(230VAC)
Anti-surge fuse 250mA 20 x 5mm
(115VAC)
Equipment Type
RB-HD1:
Stereo headphone amplifier
Physical Specification
Dimensions
(Raw):
48cm (W) x 10.8cm (D) x 4.2cm (H) (1U)
19” (W) x 4.3” (D) x 1.7” (H) (1U)
Dimensions
(Boxed):
58.5cm (W) x 22.5cm (D) x 7cm (H)
23” (W) x 8.9” (D) x 2.8” (H)
Weight:
Nett: 1.35kg Gross: 2.0kg
Nett: 3lbs
Gross: 4.4lbs
117
Audio & Video Interfaces - Headphone Distribution Amplifiers
RB-HD2 Dual Stereo Headphone Amplifier
The RB-HD2 is a high performance 2-way
stereo headphone distribution amplifier
for driving up to 2 pairs of professional
stereo headphones from a single stereo or
mono input.
A switch on the rear panel enables the
distribution of a mono signal to all four
outputs (i.e. both earpieces of a pair of
stereo headphones) via the left channel
input. The stereo/mono switch is located
on the rear panel to prevent accidental
knocking.
118
The XLR-3 inputs are electronically balanced
and can be wired unbalanced. There are
two pre-set controls on the rear panel that
adjust the level of the master input signal to
the outputs.
Each output is on a ¼” stereo jack socket
and is designed to drive 150mW into 32
ohm to 600 ohm stereo headphones. The
outputs are individually buffered with their
own front panel volume control. A LED
power indicator on the front panel displays
the power supply connection.
Category: Headphone Distribution
Amplifiers.
Specification For RB-HD2
Audio Specification
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Impedance:
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Product Function: Provides 2 headphone
outputs from a line level signal.
Typical Applications: Adding headphone
outputs to mixer programme output.
Features: Stereo/mono selection and input
gain controls.
Outputs:
2 x ¼” (6.35mm) A-gauge 3-pole
stereo jack sockets
Mains Input:
Filtered IEC, switchable 110-120V,
or 220-240V, fused, 9W max
Fuse Rating:
Anti-surge fuse 100mA 20 x 5mm
(230VAC)
Anti-surge fuse 250mA 20 x 5mm
(115VAC)
Max Input Level:
+28dBu
Input Gain
Range:
-12dB to +20dB gain (pre-set pots)
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headphones
Equipment Type
Volume
Controls:
-80dB to +11dB gain
Physical Specification
Connections
Main Stereo
Input:
2 x XLR 3 pin female (balanced, can
be unbalanced)
RB-HD2:
Dual stereo headphone amplifier
Dimensions
(Raw):
48cm (W) x 10.8cm (D) x 4.2cm (H) (1U)
19” (W) x 4.3” (D) x 1.7” (H) (1U)
Dimensions
(Boxed):
58.5cm (W) x 22.5cm (D) x 7cm (H)
23” (W) x 8.9” (D) x 2.8” (H)
Weight:
Nett: 1.35kg
Nett: 3lbs
Gross: 2.0kg
Gross: 4.4lbs
Audio & Video Interfaces - Headphone Distribution Amplifiers
RB-HD6 6 Way Stereo Headphone
Distribution Amplifier
The RB-HD6 headphone distribution
amplifier is a 1U rack-mount which
distributes stereo audio to up to 6 different
sets of headphones, or can be used as 6
independent headphone amplifiers, each
with their own input and volume control.
A typical application might be to provide
common headphone feeds for guests in
a radio studio, with a separately derived
feed, perhaps including talk-back, for
the presenter.
The main stereo input is on XLR-3 connectors
on the rear panel which are electronically
balanced and can be wired unbalanced. A
stereo/mono input select switch on the rear
panel sums left and right outputs to provide
a mono feed to the headphones. The unit
can receive an override audio signal via a jack
socket for each output channel, allowing the
unit to act as 6 single independent headphone
amps. Plugging in the jack plug will divert
the headphone output from the master
audio signal to the audio present on the jack
plug. The override audio inputs can also be
individually configured as parallel outputs of
Category: Headphone Distribution
Amplifiers.
the front sockets, by setting internal jumpers.
The master volume control adjusts overall
level of the 6 outputs and does not affect
the level of channels using the override
inputs. The master volume can be disabled by
internal jumpers.
Specification For RB-HD6
Product Function: Distributes a stereo line
level signal to 6 x headphone outputs.
Typical Applications: Provides multiple
guest headphone outputs in a radio studio.
Features: Individual & master volume
controls, insert jacks on each output,
paralleled outputs and mono/stereo switch.
Connections
Main Stereo
Inputs:
2 x XLR 3 pin female (balanced, can be
unbalanced)
Insert Inputs/
Parallel Outputs:
6 x ¼” (6.35mm) A-gauge 3-pole stereo
jack sockets (unbalanced)
Outputs:
6 x ¼” (6.35mm) A-gauge 3-pole stereo
jack sockets
Mains Input:
Filtered IEC, switchable 110-120V, or
220-240V, fused, 9W max
Fuse Rating:
Anti-surge fuse 100mA 20 x 5mm
(230VAC)
Anti-surge fuse 250mA 20 x 5mm
(115VAC)
Audio Specification
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Impedance:
хϮϬŬɏďĂůĂŶĐĞĚďƌŝĚŐŝŶŐ͕хϭϬŬɏ
unbalanced override
KƵƚƉƵƚ>ĞǀĞů͗
ƌŝǀĞƐϭϱϬŵtŝŶƚŽϯϮɏƚŽϲϬϬɏ
headphones
RB-HD6:
Individual
Volume Control:
-60dB to +18dB gain
Physical Specification
Max Input Level:
+28dBu
Dimensions
(Raw):
48cm (W) x 10.8cm (D) x 4.2cm (H) (1U)
19” (W) x 4.3” (D) x 1.7” (H) (1U)
Dimensions
(Boxed):
58.5cm (W) x 22.5cm (D) x 7cm (H)
23” (W) x 8.9” (D) x 2.8” (H)
Weight:
Nett: 1.35kg
Nett: 3lbs
Override Inputs:
+3dBu for full volume at +18dB gain
Master Vol
Control:
±10dB gain
Equipment Type
6 way stereo headphone distribution
amplifier
Gross: 2.0kg
Gross: 4.4lbs
119
Audio & Video Interfaces - Headphone Distribution Amplifiers
RB-DHD6 Digital 6 Way Stereo Headphone Distribution Amplifier
Category: Headphone Distribution
Amplifiers.
120
Product Function: Distributes a stereo
digital line level signal to 6 x headphone
outputs.
Typical Applications: Provides multiple
guest headphone outputs from the digital
output of a radio mixer or router matrix.
Features: Individual volume controls,
AES/EBU or S/PDIF input switch.
The RB-DHD6 digital 6 way headphone
distribution amplifier is a 1U rack-mount
which receives a digital input signal, as
either AES/EBU or S/PDIF and converts
it to 6 individually buffered, jack-plug,
headphone outputs, each with their own
volume control.
Audio & Video Interfaces - Headphone Distribution Amplifiers
Useful for connection to digital mixing
desks, digital routers and matrices, the RBDHD6 connects directly to an AES/EBU or
S/PDIF output to provide the highest quality
audio directly to the headphones.
The input connectors consist of a single
balanced XLR-3 for the AES/EBU input and a
single unbalanced phono connector for the
S/PDIF input.
A button located on the rear panel is used
to select either the AES/EBU, or S/PDIF,
input and de-emphasis on the output can be
controlled via dipswitch.
If de-emphasis is selected, the RB-DHD6 will
decode 50/15μs emphasis when indicated
by certain channel status bits in the
incoming digital audio data.
Specification For RB-DHD6
Fuse Rating:
Audio Specification
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ƌŝǀĞƐϭϱϬŵtŝŶƚŽϯϮɏƚŽϲϬϬɏ
headphones
Dynamic
Range:
>100dB
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ϭϭϬɏцϮϬй^ͬh
ϳϱɏцϭϱй^ͬW/&
Maximum
Output Level:
+12dBu unbalanced
Headphone
Gain Range:
-80dBu to +12dBu
Sample
30kHz-100kHz
Frequency Range:
Connections
Digital Inputs:
1 x AES/EBU XLR 3 pin female,
1 x S/PDIF RCA phono
Headphone
Outputs:
6 x ¼” (6.35mm) A/B-gauge 3-pole
stereo jack sockets
Mains Input:
Filtered IEC, 110V-120V, or 220-240V,
fused, 10W max
Anti-surge fuse 100mA 20 x 5mm
(230VAC)
Anti-surge fuse 250mA 20 x 5mm
(115VAC)
Operational Controls
De-emphasis
On/Off:
DIP switch selection
Digital Input
Select:
AES/EBU or S/PDIF, via push-switch
Equipment Type
RB-DHD6:
Digital 6 way stereo headphone
distribution amplifier
Physical Specification
Dimensions
(Raw):
48cm (W) x 10.8cm (D) x 4.2cm (H) (1U)
19” (W) x 4.3” (D) x 1.7” (H) (1U)
Dimensions
(Boxed):
58.5cm (W) x 22.5cm (D) x 7cm (H)
23” (W) x 8.9” (D) x 2.8” (H)
Weight:
Nett: 1.6kg Gross: 2.2kg
Nett: 3.5lbs Gross: 4.8lbs
When operating, the front panel power LED
flashes red and amber whenever the unit
is not synchronised to the incoming digital
signal.
121
Audio & Video Interfaces - Mixers, Source Selectors & Switchers
RB-SS10 10 Way Stereo Analogue Source Selector/Mixer
Category: Mixers, Source Selectors &
Switchers.
Product Function: Mixes or routes
up to 10 stereo inputs to one stereo
output with front panel & remote
button control.
122
Typical Applications: For monitoring
multiple stereo audio sources, as a studio
on-air switcher.
Features:
• Individual channel gain controls.
• Mix mode available.
• Optional channel selection through
remote control.
• Front panel selection Inhibit.
• Headphone monitoring of output.
• Simultaneous routing of GPIO with audio.
The RB-SS10 10 way stereo analogue
source selector/mixer is a 1U rack-mount
unit that produces a stereo analogue audio
output from 10 selectable stereo analogue
sources. There are 10 illuminated front
panel push-buttons, which select and
indicate the current channel selection. The
selection and indication is also available
through a remote connector on the rear
panel. To stop accidental front panel
selection there is a remote input to inhibit
the front panel buttons.
Two of the stereo inputs are on XLR so
that you can hot-plug audio sources,
e.g. portable recorders.
As well as being able to act as a source
select module, the RB-SS10 can act as a
mixer, by enabling the mix mode (using
the remote input).
The gain for left and right inputs can be
individually adjusted by using the pre-set
potentiometers on the front panel.
Audio & Video Interfaces - Mixers, Source Selectors & Switchers
As well as routing the selected audio signal,
the unit will also route a remote signal
input through the remote connector to the
selected input source, for starting external
audio equipment such as a CD player.
The front panel headphone output has its
own volume control, which is independent
of the level adjustment for the main
outputs, and has a maximum output level
of +12dBu.
The volume control can be made to also
alter the output level of the main XLR
outputs by using a switch on the rear panel
to enable/disable this feature.
There is a designation strip on the front
panel, useful for giving the buttons a
meaningful description. The strip covers the
input gain controls so that once configured,
they can’t easily be altered - ideal for
installation work.
The LED on the front panel is used to
indicate that power is present on the unit.
RB-SS10 Flowchart.
Specification For RB-SS10
Audio Specification
Connections
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Impedance:
ϮϬŬɏďƌŝĚŐŝŶŐ
Inputs:
KƵƚƉƵƚ
Impedance:
фϱϬɏ
4 x XLR 3 pin female (2 x stereo)
(balanced, can be unbalanced)
2 x 25 way D-type socket (female)
(4 stereo balanced channels on each)
Equipment Type
2 x XLR 3 pin male (stereo balanced,
can be unbalanced)
Maximum
Input Level:
+28dBu
Outputs:
Maximum
Output Level:
+28dBu
Remote
Start I/O:
25 way D-type plug (male)
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Response:
ϮϬ,njƚŽϮϬŬ,njцϬ͘ϭĚ;ϲϬϬɏůŽĂĚ͕
ref 1kHz)
Remote Select/
Switch Inputs:
25 way D-type socket (female)
Input Gain
Range:
Adjustable 8dB loss to 20dB gain
(L & R adjust).
Status Outputs:
25 way D-type socket (female)
Mains Input:
Filtered IEC, 110V-120V, or 220-240V
switchable, fused, 6W maximum.
Fuse Rating:
Anti-surge fuse 160mA 20 x 5mm
(230VAC)
Anti-surge fuse 315mA 20 x 5mm
(115VAC)
Common
>66dB typically
Mode Rejection:
Noise:
-96dB unity gain ref +8dBu
Max Headphone
Output Level:
+12dBu
RB-SS10:
10 way stereo analogue source
selector/mixer
Physical Specifications
Dimensions
(Raw):
48cm (W) x 10.8cm (D) x 4.2cm (H) (1U)
19” (W) x 4.3” (D) x 1.7” (H) (1U)
Dimensions
(Boxed):
58.5cm (W) x 22.5cm (D) x 7cm (H)
23” (W) x 8.9” (D) x 2.8” (H)
Weight:
Nett: 1.5kg Gross: 2.0kg
Nett: 3.3lbs Gross: 4.4lbs
123
Audio & Video Interfaces - Mixers, Source Selectors & Switchers
RB-DSS10 10 Way Stereo Digital Source Selector
Category: Mixers, Source Selectors &
Switchers.
Product Function: 10 stereo digital input
source selection unit to 1 stereo output.
Typical Applications: Switching the
broadcast output from digital mixers in
multiple studios.
Features:
• Both AES/EBU & S/PDIF I/O.
• Individual channel gain controls.
• Optional channel selection via remote
control.
• Front panel selection inhibit.
• Headphone monitoring of output.
• Routing of remote inputs with
selected audio channel.
The RB-DSS10 digital source selector is a
1U rack-mount which produces an AES/EBU
and S/PDIF level digital audio output from
10 selectable AES/EBU or S/PDIF digital
input signals. There are 10 illuminated
front panel push-buttons, which select and
indicate the current channel selection. The
selection and indication is also available
through a remote connector on the rear
panel. To stop accidental front panel
selection there is a remote input to inhibit
the front panel buttons.
The digital receivers in this unit are fully
24 bit, 96kHz capable. When an input
is selected from the front panel, or
remotely, the unit will attempt to capture
the incoming signal on either the AES/
EBU or the S/PDIF signal inputs, with
priority given to the AES/EBU input. If the
AES/EBU signal becomes locked while
the S/PDIF signal is routed, the unit will
automatically switch to the incoming
AES/EBU signal.
Once the receiver has successfully locked
to a digital input, the LED illuminates,
the tally is made, and the audio is routed
simultaneously to both the digital audio
outputs and converted to analogue
audio for monitoring on the front panel
headphone socket. If the incoming audio
signal is not present, the push-button LED
and remote tally flash to indicate that the
incoming digital signal is missing.
The headphone output has its own volume
control, which is independent of the level
adjustment for the main outputs, and has a
maximum output level of +12dBu.
As well as routing the selected audio signal,
the unit will also route a remote signal
input through the remote connector to the
selected input source, for starting external
audio equipment, such as a CD player.
There is a designation strip on the front
panel, useful for giving the buttons a
meaningful description.
RB-DSS10 Flowchart.
124
Audio & Video Interfaces - Mixers, Source Selectors & Switchers
The LED on the front panel is used to
indicate that power is present on the unit.
However, it also has a secondary role to
indicate whether the selected channel is
routing the AES/EBU (red LED) or S/PDIF
input (amber LED).
Specification For RB-DSS10
Audio Specification
Connections
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ϳϱɏцϱйƵŶďĂůĂŶĐĞĚ;^ͬW/&Ϳ
Signal Level:
3V/10V peak to peak min/max
(AES/EBU)
0.5V ±20% peak to peak (S/PDIF)
Audio Outputs:
Sample Freq
Range:
30-100kHz (i.e. including 32kHz,
44.1kHz, 48kHz, 64kHz, 88.2kHz and
96kHz), following input signal
Remote
Start I/O:
Bit Depth:
Up to and including 24 bits, following
input signal
Remote
1 x 25 way D-type socket (female)
Input Select & Switch Inputs:
Max Headphone
Output Level:
+12dBu
Status Outputs:
1 x 25 way D-type socket (female)
Headphone
Outputs:
6 x ¼” (6.35mm) A-gauge 3-pole
stereo jack sockets
Audio Inputs:
8 x AES/EBU XLR 3 pin female
2 x AES/EBU (part of 1 x 25 way
D-type plug)
10 x S/PDIF (part of 1 x 25 way
D-type plug)
Mains Input:
Filtered IEC, continuously rated
85-264VAC @ 47-63Hz, max 10W
Fuse Rating:
Anti-surge fuse 1A 20 x 5mm
Equipment Type
RB-DSS10
10 way stereo digital source selector
1 x AES/EBU XLR 3 pin male
1 x S/PDIF (part of 1 x 25 way D-type
plug)
Physical Specifications
Dimensions
(Raw):
48cm (W) x 10.8cm (D) x 4.2cm (H) (1U)
19” (W) x 4.3” (D) x 1.7” (H) (1U)
1 x 25 way D-type plug (male)
Dimensions
(Boxed):
58.5cm (W) x 22.5cm (D) x 7cm (H)
23” (W) x 8.9” (D) x 2.8” (H)
Weight:
Nett: 1.6kg Gross: 2.2kg
Nett: 3.5lbs Gross: 4.8lbs
125
Audio & Video Interfaces - Mixers, Source Selectors & Switchers
RB-PMX4 10 Input, 4 Output Analogue Pre-set Mixer
Category: Mixers, Source Selectors &
Switchers.
Product Function: 10 mono input to 4
mono output mixer/router.
126
Typical Applications: Four bus mini-mixer,
a multiple clean-feed generator, a 4 zone
mixer for pubs and clubs and a quad stereo
to mono converter.
Features:
• Neutrik XLR I/O.
• Individual input channel gain controls.
• Total flexibility for routing/mixing signals.
• Simple to use and configure.
• Front panel cover for secure applications.
The RB-PMX4 is a high performance 10
mono input to 4 mono output pre-set
mixer. Each of the four outputs has a 10
way DIP switch associated with it to select
which of the 10 inputs are routed to it. So,
by altering the DIP switches, any of the
input sources can be mixed to any of the
outputs. The DIP switches are enclosed
by a screw-on cover on the front panel so
that the settings can not be accidentally
changed for secure applications.
Audio & Video Interfaces - Mixers, Source Selectors & Switchers
The RB-PMX4 has been designed for
situations where a small mixer is needed
for installations where it will be configured
and then only altered occasionally, or never
altered at all.
Uses for this product are numerous
including a four bus mini-mixer, a multiple
clean-feed generator, a 4 zone mixer for
pubs and clubs and a quad stereo to mono
converter to name a few.
The XLR-3 inputs and outputs are
electronically balanced and can be wired
unbalanced. Each output is individually
buffered so that a short circuit on one won’t
affect the others.
Each input has its own gain control which
is a pre-set potentiometer accessible
through the front panel. This provides gain
adjustment of -8dB to +18dB. This is useful
for normalising consumer and professional
signals to give outputs of -15dBu and 0dBu
respectively.
Specification For RB-PMX4
Audio Specification
/ŶƉƵƚ
Impedance:
хϮϬŬɏďĂůĂŶĐĞĚďƌŝĚŐŝŶŐ
Maximum
Input Level:
+36dBu
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Response:
ϮϬ,njƚŽϮϬŬ,njцϬ͘ϭĚƵ;ϲϬϬɏůŽĂĚ͕
@ 1kHz)
Common
>60dBu typically
Mode Rejection:
Noise:
-86dBu RMS 22Hz-22kHz unity gain,
ref +8dBu
KƵƚƉƵƚ
Impedance:
фϱϬɏ
Maximum
Output Level:
+28dBu
Gain Range:
Adjustable -8dBu to +18dBu gain
Off-isolation/
Crosstalk:
>90dBu @ 1kHz
Distortion:
<0.01% @ 1kHz, 0dBu to +26dBu
Connections
Inputs:
10 x XLR 3 pin female (Balanced, can
be unbalanced)
Outputs:
4 x XLR 3 pin male (Balanced, can be
unbalanced)
Mains Input:
Filtered IEC, switchable 110-120V,
or 220-240V, fused, 6W maximum
Fuse Rating:
Anti-surge fuse 100mA 20 x 5mm
(230VAC)
Anti-surge fuse 250mA 20 x 5mm
(115VAC)
Equipment Type
RB-PMX4:
10 input, 4 output analogue pre-set
Mixer
Physical Specification
Dimensions
(Raw):
48cm (W) x 10.8cm (D) x 4.2cm (H) (1U)
19” (W) x 4.3” (D) x 1.7” (H) (1U)
Dimensions
(Boxed):
58.5cm (W) x 22.5cm (D) x 7cm (H)
23” (W) x 8.9” (D) x 2.8” (H)
Weight:
Nett: 1.5kg Gross: 2.2kg
Nett: 3.3lbs Gross: 4.8lbs
127
Audio & Video Interfaces - Mixers, Source Selectors & Switchers
RB-DMX4 4 x 4 Channel Digital Audio Mixer/Router
192
RB-DMX4 4 x 4 Channel Digital Audio Mixer/Router
(With RB-SYD Video Sync Board).
Category: Mixers, Source Selectors &
Switchers.
Product Function: 4 mono digital inputs
to 4 mono digital outputs mixer/router.
Typical Applications: Four bus digital
mini-mixer, multiple digital clean-feed
generator, a 4 zone mixer for pubs and
128
clubs and a dual digital stereo to mono
converter.
Features:
• Passive signal path.
• AES/EBU, S/PDIF & TOSlink I/O.
• Headphone output with input selector.
• Input & output presence indicators.
• Input channel gain controls.
• TTL wordclock & AES/EBU sync inputs.
• Optional sync cards can be added.
• Mono or stereo operation.
• Total flexibility for routing/mixing signals.
• Front panel cover for secure applications.
The RB-DMX4 is a digital mixer capable of
mixing or routing 4 mono input channels
into 4 mono outputs, or 2 stereo inputs
into 2 stereo outputs. The inputs are
sample rate converted to allow sources
of different sample rates to be mixed.
The flexible Mix Matrix allows for a wide
variety of mixing options and creativity,
using 4 blocks of 4 way DIPSwitches to
select which inputs are mixed or routed to
which outputs.
The RB-DMX4 has 4 x digital mono audio
inputs, selectable in pairs via front panel
INPUTS 1 & 2 and INPUTS 3 & 4 push
buttons, from either AES/EBU balanced
XLRs, S/PDIF unbalanced phonos or TOSlink
unbalanced optical inputs. Sample rate
converters on each input mean that sources
of different sample rates can be used with
the output sample rate being defined
independently. The colour of the INPUTS 1
& 2 and INPUTS 3 & 4 push-buttons indicate
Audio & Video Interfaces - Mixers, Source Selectors & Switchers
whether the input source is synchronised
(no colour) or not (flashing green and red).
locked to, the internal on-board clock
generator. No sync signal is used or required.
Each input has a trim pot, which can be
used to attenuate the input signal. This
allows for a perfect mix of channels at
different audio levels. Audio presence LEDS
around each input button give an indication
of input audio level. There is one LED for
each channel. There are also 4 presence
LEDS around the MONITOR button which
give an indication of output level. Additional
gain can be added by accessing the OUTPUT
GAIN mode.
Auto Sync Mode - In this mode the digital
output sample rate follows the selected sync
input. When the sync signal is not present
the output sample rate will be set by, and
locked to, the internal on-board clock
generator at the selected output frequency.
There are 2 stereo outputs which are
available as simultaneous AES/EBU balanced
XLRs, S/PDIF unbalanced phonos or TOSlink
unbalanced optical outputs. The output
sample rates are selectable via rear panel
DIPSwitches from one of 32kHz, 44.1kHz,
48kHz, 88.2kHz, 96kHz, 176.4kHz or 192kHz.
The unit has TTL wordclock BNC and AES/
EBU XLR synchronising inputs as standard
and optionally, the RB-SYA and RB-SYD
synchronisation boards can be fitted to
synchronise the unit to analogue or digital
video signals. A rear panel DIPswitch
block is used to decide whether the unit
is synchronised to Input 1 & 2, Input 3 &
4, the AES/EBU sync input, the wordclock
sync input or an optional video sync
board. The DIPswitch block also selects the
synchronisation mode of the unit and the
MONITOR button flashes whenever the unit
is not synchronised to an incoming sync
signal. Selectable sync modes are as follows:
Master Mode - In this mode the digital
output sample rate is simply set by, and
Auto Lock Mode - The digital output
sample rate follows the sync input. If the
sync signal is removed then the output
sample rate will be set by, and locked to,
the internal on-board clock generator at the
closest frequency available to the previous
sync input.
Slave Mode - In this mode the digital
output sample rate follows the sync input.
When the sync signal is not present the
digital output is turned off.
There is a monitor socket on the front panel
with a gain pot to allow you to monitor the
output of each channel. The monitored
channel can be selected via a push button
on the front panel which, when held, can
also supply up to 12dB of gain. If the level
that is being monitored is close to full scale,
a 12dB attenuation can be added to the
monitor output via a DIPswitch on the rear
panel.
The unit can be placed in mono or stereo
mode via rear panel DIPswitch. Stereo mode
allows you to monitor the two input pairs
as stereo channels as well as controlling the
input gain as a pair, giving tied audio levels.
The RB-DMX4 has been designed to have a
passive signal path through the main input,
so if power to the unit fails, signal inputs 1
& 2 are routed to outputs 1 & 2 and signal
inputs 3 & 4 are routed to outputs 3 & 4.
Rear Panel Operational Controls
This is essential for applications such as
installation at transmitter sites, where a
power failure to the unit should not prevent
the audio input signal from being output
to the transmitter. Please note that this is
not true for the TOSLink outputs which are
muted.
Master Select:
32, 44.1, 48, 88.2, 96,176.4 or 192kHz
Frequency via rear panel DIPSwitches
Sync Source
Select:
INPUTS 1&2, INPUTS 3&4, AES Sync,
Word Clock, Video Sync via rear
panel DIPSwitches
Sync Mode
Select:
Master, Auto Sync, Auto Lock, Slave via
rear panel DIPSwitches
Stereo Features: Stereo gain control and monitor
outputs via rear panel DIPSwitches
The RB-DMX4 can be controlled using
Sonifex free software, SCi. Contact Sonifex
for further information if you have a
particular requirement that isn’t catered for
by the RB-DMX4 as standard.
Monitor
Attenuation:
12dB Monitor attenuation via rear
panel DIPSwitches
Serial Mode:
Enter serial control mode via rear panel
DIPSwitches
Boot Mode:
Boot up base code or firmware via
rear panel DIPSwitches
Connections
Specification For RB-DMX4
Digital Inputs:
2 x AES/EBU XLR 3 pin female
2 x S/PDIF RCA phono
2 x TOSLink optical input
Digital Outputs:
2 x AES/EBU XLR 3 pin plug
2 x S/PDIF RCA phono socket
2 x TOSLink optical output
Sync Inputs:
1 x AES/EBU XLR 3 pin female
1 x Word Clock BNC
1 x Video Input (optional)
Audio Specification
Dynamic Range:
>138dB
Remote I/O Port: 15 way D-type plug
Distortion
and Noise:
<-137dB THD + N at 1kHz, ref 0dB FS
Serial Port:
RS232, 9 pin D-type socket
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Mains Input:
Universal filtered IEC, continuously
Signal Level:
Balanced: 3V/10V peak to peak
min/max
Unbalanced: Min 0.5V±20% peak
to peak
rated [email protected] 63Hz, max 10W
Fuse Rating:
Anti-surge fuse 2A 20 x 5mm
Equipment Type
RB-DMX4:
4 x 4 channel digital audio mixer/router
Physical Specifications
Sample
Frequencies:
32, 44.1, 48, 88.2, 96,176.4 or 192kHz
Dimensions
(Raw):
48cm (W) x 10.8cm (D*) x 4.2cm (H) (1U)
19” (W) x 4.3” (D*) x 1.7” (H) (1U)
Bit Depth:
Up to and including 24 bit
Dimensions
(Boxed):
59cm (W) x 27.5cm (D*) x 11cm (H)
23.2” (W) x 10.8” (D*) x 4.3” (H)
Front Panel Operational Controls & Indicators
Weight:
Digital
Input Select:
AES/EBU, S/PDIF or TOSlink optical
via INPUT 1 & 2 or INPUT 3 & 4
push-buttons
Nett: 1.4kg Gross: 2.0kg
Nett: 3.1lb Gross: 4.4lb
Accessories
Mix Control:
Output mix selection system via
front panel DIPSwitches
Input Gain:
Input gain control for four INPUT
channels via potentiometers, from
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Output Gain:
Adjustable via front panel push switch to
0dB, 3dB, 6dB or 12dB
Monitor
Select Control:
Headphone monitor channel select
and output gain via push button
Indicators:
Input and output presence indicators
via bicolour LEDS around each
push button
RB-SYA:
Analogue video sync board
(NTSC, PAL & SECAM)
RB-SYD:
Digital video sync board (SD-SDI
& HD-SDI)
RB-RK3:
1U Rear panel rack kit for large
Redboxes
* Note that this product is deeper than standard Redboxes
129
Audio & Video Interfaces - Mixers, Source Selectors & Switchers
RB-SSML1 Mic/Line Source Selector with Compressor/Limiter
Category: Mixers, Source Selectors &
Switchers.
Product Function: Compressing or
limiting an incoming mic or stereo line
signal, mixing with a stereo monitor
input for metering and mixing to two
headphone outputs.
130
Typical Applications: Designed for voiceover & over-dubbing applications in voiceover booths/suites, use with audio codecs.
Features:
• Two headphone outputs.
• Mic high pass filter & phonatom power.
• Wide mic gain range adjustment.
• Line inputs & outputs can be balanced or
wired unbalanced.
• Compression ratio & threshold limits
adjustment.
• Bright LED output level metering.
The RB-SSML1 is a 1U rack-mountable
source selector for compressing or limiting
an incoming microphone or stereo line
signal and mixing this signal with a stereo
monitor input, which can then be metered
and mixed to two headphone outputs.
The unit is mainly used in situations where
level control is required, for example in
voiceover applications.
The mic input consists of an independent
low-noise microphone pre-amplifier for
converting microphone level signals to a line
level. There are independent switches to
control a high pass filter (low frequency rolloff at 125Hz) and to provide phantom power
at +48V to the connected microphone.
A preset pot on the rear panel allows
adjustment of the mic gain from 36dB to
75dB.
Audio & Video Interfaces - Mixers, Source Selectors & Switchers
The compression ratio and threshold limits
of the compressor/limiter section are
fully adjustable via linear pots situated on
the front panel. The threshold can be set
between -30dBu and +20dBu. When the
input signal rises above the threshold level
a soft-knee compression is applied at the
selected ratio. The compressor has an attack
time of approximately 20ms and a release
time of approx 400ms, and can operate
at ratios of 1:1 (no compression) to :1
(limiting). A front panel BYPASS button can
be used, where no compression is applied.
A rear-panel DIPswitch allows compression
only to be applied to the mic input and not
the stereo line input.
The metering is carried out after the
compressor/limiter section and consists of
two rows of 12 round LEDs showing levels
The XLR-3 stereo line output can be fed
from either the mic or line input, but not
the stereo monitor input. The line output
is electronically balanced and can be wired
unbalanced by grounding the non-phase
signal, allowing you to feed both balanced
and unbalanced equipment. A pushbutton
switch is provided to reduce the output by
10dB for this purpose.
An LED power indicator on the front panel
displays the power supply connection.
Specification For RB-SSML1
Connections
Technical Specification
Mic Input:
Maximum
Input Level:
-10dBu (mic), +28dBu (line),
electronically balanced
Line Input:
2 x XLR 3 pin female (Balanced)
Monitor Input:
2 x XLR 3 pin female (Balanced)
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Impedance:
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Output:
2 x XLR 3 pin male (Balanced, can be
unbalanced)
Maximum
Output Level:
+28dBu
Headphone
Outputs:
2 x ¼” (6.35mm) A-gauge 3-pole
stereo jack sockets
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Impedance:
фϱϬɏ Mains Input:
Filtered IEC, 110V-120V, or 220-240V
switchable, fused, 9W maximum
Low Frequency
125Hz @ 6dB/octave
Fuse Rating:
Gain Range
(mic):
Adjustable 36dB to 75dB gain
(-80dB volume min.)
Anti-surge fuse 100mA 20 x 5mm (230VAC)
Anti-surge fuse 250mA 20 x 5mm (115VAC)
Volume Control
(line):
-80dB to +6dB gain (-70dB +16dB with
additional input gain)
RB-SSML1:
E.I.N:
130dB
Physical Specification
Distortion:
<0.02% THD @ 1kHz, ref +8dBu output
Common Mode
Rejection:
>66dB typically
Dimensions
(Raw):
48cm (W) x 10.8cm (D) x 4.2cm (H) (1U)
19” (W) x 4.3” (D) x 1.7” (H) (1U)
Dimensions
(Boxed):
58.5cm (W) x 22.5cm (D) x 7cm (H)
23” (W) x 8.9” (D) x 2.8” (H)
Weight
Nett: 1.3kg Gross: 2.0kg
Nett: 2.9lbs Gross: 4.4lbs
Frequency
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Mic L.F. Filter Mic.
Volume
20Hz to 20kHz ±0.3dB
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Threshold
Compression
Fixed
Gain
LED Level
Jumper Meter Drivers
Compressor
Controls
R
Stereo
Output
Fixed
Gain
Line Volume L
Fixed Gain
Stereo
Line
Mic/Line source selector with
compressor/limiter
Analogue
Switch
Gain L.F. Filter
Phantom In/Out
On/Off
L
1 x XLR 3 pin female (Balanced)
Equipment Type
Phantom Power: 48V
RB-SSML1 Flowchart.
Phantom
Power (48V)
Roll-Off:
Line Volume R
Jumper
Peak
Hold
On/Off
Fixed Gain
L
Volume
Stereo Jack
Output
Stereo Fixed Gain
Monitor
R
Fixed Gain
Line Volume L
M
The XLR-3 stereo monitor input has an
adjustable volume control via a back panel
recessed pot, and has an additional 10dB
gain increase via a switch on the rear panel,
for use with unbalanced equipment. The
audio on this input is only present on
the headphone outputs, and is therefore
suitable as a return feed from a codec, a PC
audio output, or similar equipment.
between -17dB and +11dB. An internal
jumper allows the metering to follow
either the stereo output, or the headphone
monitor outputs and a rear panel DIPswitch
can disable the peak hold display.
Line Volume R
M
A front panel switch selects between mic &
line inputs. Both mic and line inputs have
fully adjustable volume control via front
panel pots, with an additional 10dB gain
increase for the line input via a rear panel
switch, for use with unbalanced equipment.
The mono mic input is converted to a
stereo signal before being passed to the
compressor/limiter.
Audio Output
Amplifier
Stereo Jack
Output
131
Audio & Video Interfaces - Mixers, Source Selectors & Switchers
RB-OA3 3 Studio On-Air Switcher
Category: Mixers, Source Selectors &
Switchers.
Product Function: To switch the on-air
transmission output between multiple
radio studios.
132
Typical Applications: Designed for multistudio switching & routing the studio
output to the transmitter.
Features:
• Controls the offer/accept switching
for 3 studios.
• Switches 3 stereo bidirectional channels
so allows sharing of equipment
between studios, e.g. studio output,
codec clean-feeds, telco feeds & other
equipment.
• Has a stereo transformer mix input, e.g.
for playout system.
• Continuity mode allows switching to/
from a PC playout system, so a 3rd
studio can be used as sustaining service
allowing other studios to be powered
down for maintenance or power saving.
• Latching relay switching means
transmission path remains intact on
power-fail.
• Expandable to switch 6 stereo channels
between 5 studios.
• Last studio to offer bus.
The RB-OA3 is a 1U rack-mount, unity
gain on-air switcher, capable of switching
three stereo pairs between three studios.
Each studio can control the transmission
path together with two peripheral paths
for equipment such as a codec or hybrid
and there is also a “Last studio to offer”
bus, allowing for seamless and continuous
broadcast from any multi-studio radio
network. A “sustain” mode allows for a
sustaining system, such as a PC automation
system, to control the broadcast.
Audio & Video Interfaces - Mixers, Source Selectors & Switchers
Multiple RB-OA3 units can be connected
together to switch more studios or more
stereo pairs.
The switching is achieved using relays,
except the “last studio to offer” which
is switched by an analogue switch. The
transmission path is switched using latching
relays. This means that if there is a power
failure to the unit, the transmission path will
remain selected.
All studios are connected using 25 way
D-types for electronically balanced audio
signals and control is achieved using 15
way D-types, connecting to an external
control unit such as the Sonifex S2-MTBS
mixer control panel. A transmission mix
connection is included to mix audio which is
generic to all studios into the transmission
path. This could be used for jingles or
adverts for example. The RB-OA3 also allows
for the control of a profanity delay to be
shared by all connected studios.
Each studio has the ability to offer the
transmission. Once offered, the transmission
is fed to the other studios via the “last
studio to offer” bus. The next scheduled
station
can then fade in the transmission and accept
at the appropriate time meaning
the transmission can be continuous.
The “sustain” mode can be used to control
an automated studio, such as an overnight
music system. In this case, station control
will switch to the automated system
by holding the Offer button down for a
number of seconds, pre-determined by
a calibration routine. This will select the
sustaining studio, set as studio three, which
will immediately offer control back to all
remaining studios. Any studio can then
accept to resume orthodox broadcasting.
The RB-OA3 can be expanded to switch
between up to 5 studios or up to 6 stereo
pairs by connecting a multiple of units
together via RJ45 serial connections. With
the addition of a single unit, expansion
in “studio” mode allows for 2 additional
studios to take control of the transmission
path and additional equipment. If the
expansion is made in “bus” mode, then
three additional stereo channels can be
added. 4 x RB-OA3 units can be connected
together to switch 6 stereo pairs between
up to 5 studios.
Specification For RB-OA3
Common Mode
Rejection:
< -64dB @ 10kHz
The modes are configured by dip switch
configurations, on the rear of each unit. Two
DIP switches control the unit ID and there
are two switches which decide between
“studio” and “bus” modes. A master
unit, defined by a preset ID, conducts all
communication between all units.
Distortion:
0.5% THD ref 17dBu @ 40Hz
Bandwidth:
±0.5dBu 10Hz to 36kHz
Audio Specification
Transmission and Peripheral Path:
Relays are used for switching these specific paths leading to
a passive, transparent audio path.
LSO Path:
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Gain Range:
Unity gain
Frequency Response: 20Hz to 20kHz ±0.1dB
Common Mode
Rejection:
< -66dB typically
Distortion:
0.01%THD @ 1kHz
Noise:
-100dB unity gain, ref +8dBu
Mix Audio Transformer Specifications:
Connections
Studio I/O:
3 x 25 way D type socket (female)
Transmission I/O:
1 x 25 way D type socket (female)
Dump/Delay Control: 1 x 9 way D type plug (male)
Studio Control:
3 x 15 way D type sockets (female)
LSO Expansion Port:
1 x 9 way D type socket (female)
Mix Input:
1 x 9 way D type socket (female)
Serial Ports:
2 x RJ45
Mains Input:
Filtered IEC, 110V-120V, or
220-240V switchable, fused,
9W maximum
Fuse Rating:
Anti-surge fuse 1A 20 x 5mm
Equipment Type
RB-OA3:
RB-0A3
RB-0A3b
RB-0A3a
3 Studio on-air switcher
Physical Specification
Dimensions
(Raw):
48cm (W) x 15.8cm (D ) x 4.2cm
(H) (1U)
19” (W) x 6.2” (D) x 1.7” (H) (1U)
Dimensions
(Boxed):
59cm (W) x 27.5cm (D*) x 11cm (H)
23.2” (W) x 10.8” (D*) x 4.3” (H)
Weight:
Nett: 1.9kg Gross: 2.5kg
Nett: 4.2lbs Gross: 5.5lbs
RB-0A3c
Adding RB-OA3a allows 3 stereo channels to be switched between 5 studios.
Adding RB-OA3b allows 6 stereo channels to be switched between 3 studios.
Adding RB-OA3a , RB-OA3b and RB-OA3c allows 6 stereo channels to be switched between 5 studios.
133
Audio & Video Interfaces - Mixers, Source Selectors & Switchers
RB-OA3R Remote Switcher Panel For RB-OA3
The RB-OA3 has been designed for
installation in a central technical area with
cabling to custom panels, or selection
switches, in each studio. If you don’t have
any custom switch panels available, you
can use an RB-OA3R in each studio.
front panel push button is illuminated by
coloured LEDs and controls the OFFER,
ACCEPT, DELAY and DUMP functions
The RB-OA3R is a 1U rack mount switch
unit for use in each studio that needs to
be connected to the STUDIO 1-3 CONTROL
connectors on an RB-OA3. It takes it’s power
from the RB-OA3 unit so needs no power
supply itself.
Studio Control:
It contains four buttons which are used to
control the functions of the RB-OA3. Each
134
Category: Mixers, Source Selectors &
Switchers.
Product Function: Remote switch panel
for RB-OA3 unit.
Typical Applications:
Allows remote control of the RB-OA3 unit.
Features:
• Powered from the RB-OA3 unit.
• Allows the RB-OA3 unit to be situated in
the main control room.
• Illumination of selected button showing
status.
Specification For RB-OA3R
Audio Specification
Connections
1 x 15 way D type plug (male)
Equipment Type
RB-OA3R:
1 Studio on-air switcher
Physical Specification
Dimensions
(Raw):
48cm (W) x 10.8cm (D ) x 4.2cm (H) (1U)
19” (W) x 4.3” (D) x 1.7” (H) (1U)
Dimensions
(Boxed):
58.5cm (W) x 22.5cm (D) x 7cm (H)
23” (W) x 8.9” (D) x 2.8” (H)
Weight:
Nett: 1.2kg Gross: 1.8kg
Nett: 2.6lbs Gross: 4lbs
RB-OA3C Expansion Unit Cable For RB-OA3
If you add another RB-OA3 to an existing
unit to expand either the number of buses
or studios, you need additional cables to
connect it which are contained in this kit:
1 x Transmission output expansion cable,
25 pin D-type male to 25 pin D-type male,
30cm lead, wired pin to pin.
1 x LSO expansion cable, 9 pin D-type male
to 9 pin D-type male, 30cm lead, wired pin
to pin.
1 x RS232 expansion cable, RJ45 to RJ45
standard wiring, 30cm lead.
1 x kit should be used for each expansion
RB-OA3 being used.
Audio & Video Interfaces - Microphone Amplifiers & Limiters
RB-MA1 Single Microphone Amplifier
The RB-MA1 consists of a low noise
microphone pre-amplifier for converting
mic level signals to line level, or for driving
long lines from microphones to mixing
equipment.
Category: Microphone Amplifiers &
Limiters.
Product Function: Amplification of mic
level signals to line level.
Typical Applications: To amplifiy
additional microphones in a studio
talks area, court room microphone
amplification
Features:
• Neutrik XLR connectors.
• Mic input gain adjustment.
• +48V phantom power.
• High pass (rumble) filter.
The connections and controls are on the
rear panel. The microphone input is an
XLR-3 type and is electronically balanced.
The gain for the input can be adjusted by
a recessed pre-set potentiometer which
allows for the use of both dynamic and
powered microphones.
The line output is of an XLR-3 type and is
electronically balanced. It can be wired
unbalanced by grounding the non-phase
signal, allowing you to feed both balanced
and unbalanced equipment.
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Impedance:
Gain Range:
Adjustable 36dB to 75dB gain
There is a switch to control a high pass
filter (low frequency roll-off at 125Hz) and
to provide phantom power at +48V to the
connected microphone.
Distortion:
0.01% THD @ 1kHz, ref +8dBu output
Specification For RB-MA1
Audio Specification
Maximum
Input Level:
-10dBu
Maximum
Output Level:
+28dBu
Low Frequency
Roll-Off:
125Hz @ 6dB/octave
E.I.N.:
130dB
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Phantom Power: 48V
Connections
Input:
1 x XLR 3 pin female (Balanced)
Output:
1 x XLR 3 pin male (Balanced, can be
unbalanced)
Mains Input:
Filtered IEC, switchable 110-120V, or
220-240V, fused, 6W max
Fuse Rating:
Anti-surge fuse 100mA 20 x 5mm
(230VAC)
Anti-surge fuse 250mA 20 x 5mm
(115VAC)
Equipment Type
RB-MA1:
Single microphone amplifier
Physical Specification
Common
>60dB typically
Mode Rejection:
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Response:
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ref 1kHz)
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Impedance:
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Dimensions
(Raw):
28cm (W) x 10.8cm (D) x 4.2cm (H) (1U)
11” (W) x 4.3” (D) x 1.7” (H) (1U)
Dimensions
(Boxed):
36cm (W) x 20.5cm (D) x 6cm (H)
14.2” (W) x 8” (D) x 2.4” (H)
Weight:
Nett: 0.90kg Gross: 1.35kg
Nett: 2lbs
Gross: 3lbs
135
Audio & Video Interfaces - Microphone Amplifiers & Limiters
RB-MA2 Dual Microphone Amplifier
Category: Microphone Amplifiers &
Limiters.
Product Function: Amplification of two
mic level signals to line level.
Typical Applications: To amplifiy
additional microphones in a studio
talks area, court room microphone
amplification, converting mic signal
where mic and amp are a long distance
apart.
Features:
• Neutrik XLR connectors.
• Mic input gain adjustment.
• +48V phantom power.
• High pass (rumble) filter.
136
The RB-MA2 consists of two independent
low-noise microphone pre-amplifiers for
converting mic level signals to line level, or
for driving long lines from microphones to
processing equipment.
For each channel there are independent
switches to control a high pass filter (low
frequency roll-off at 125Hz) and to provide
phantom power at +48V to the connected
microphones.
All connections and controls are on the
rear panel. The microphone inputs are
XLR-3 type and are electronically balanced.
The input gain for each input can be
adjusted individually by a recessed pre-set
potentiometer which allows for the use of
both dynamic and powered microphones.
Specification For RB-MA2
The XLR-3 line outputs are electronically
balanced and can be wired unbalanced
by grounding the non-phase signal,
allowing you to feed both balanced and
unbalanced equipment.
Audio Specification
Gain Range:
Adjustable 36dB to 75dB gain
(each input)
Distortion:
0.01% THD @ 1kHz, ref +8dBu output
Phantom Power: 48V
Connections
Inputs:
2 x XLR 3 pin female (Balanced)
Outputs:
2 x XLR 3 pin male (Balanced, can be
unbalanced)
Mains Input:
Filtered IEC, switchable 110-120V, or
220-240V, fused, 6W max
Fuse Rating:
Anti-surge fuse 100mA 20 x 5mm
(230VAC)
Anti-surge fuse 250mA 20 x 5mm
(115VAC)
Maximum
Input Level:
-10dBu
Maximum
Output Level:
+28dBu
Low Frequency
Roll-Off:
125Hz @ 6dB/octave
E.I.N.:
130dB
Physical Specification
Common Mode
Rejection:
>60dB typically
Dimensions
(Raw):
28cm (W) x 10.8cm (D) x 4.2cm (H) (1U)
11” (W) x 4.3” (D) x 1.7” (H) (1U)
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Response:
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ref 1kHz)
Dimensions
(Boxed):
36cm (W) x 20.5cm (D) x 6cm (H)
14.2” (W) x 8” (D) x 2.4” (H)
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Impedance:
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Weight RB-MA2: Nett: 1.00kg Gross: 1.45kg
Nett: 2.2lbs Gross: 3.2lbs
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Impedance:
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Equipment Type
RB-MA2:
Dual microphone amplifier
Audio & Video Interfaces - Microphone Amplifiers & Limiters
RB-MA2G Dual Microphone Amplifier
Gain Controls
Category: Microphone Amplifiers &
Limiters.
Product Function: Amplification of two
mic level signals to line level.
Typical Applications: To amplifiy
additional microphones in a studio
talks area, court room microphone
amplification, converting a mic to a line
signal where mic and amp are a long
distance apart.
Features:
• Front panel level adjustment.
• Front panel bright LED metering.
• Neutrik XLR connectors.
• Coarse mic input gain adjustment.
• +48V phantom power.
• High pass (rumble) filter.
The RB-MA2G consists of two independent
low-noise microphone pre-amplifiers with
front panel level controls and metering,
for converting microphone level signals
to line level, or for driving long lines from
microphones to mixing equipment.
All connections are on the rear panel. The
microphone input is XLR-3 type and is
electronically balanced. The input gain can
be adjusted individually by a recessed pre-set
potentiometer. A front panel potentiometer
gives ±12dB level adjustment and a 12 segment
LED meter shows the audio level for each
channel. The XLR-3 line output is electronically
balanced and can be wired unbalanced by
grounding the non-phase signal, allowing you to
feed both balanced and unbalanced equipment.
For each channel there are independent
switches to control a high pass filter (LF roll-off
at 125Hz) and to provide phantom power at
+48V to the connected microphone. An LED
power indicator on the front panel displays the
power supply connection.
Specification For RB-MA2G
Audio Specifications
Maximum Input Level:
+4dBu
Maximum Output Level: +28dBu
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Low Frequency Roll-Off: 125Hz @ 6dB/octave
Gain Range:
Adjustable 12dB to 80dB gain
E.I.N:
-128dBu, 20kHz BW, max gain,
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Distortion:
<0.01%, +8dBu, 20Hz – 20kHz,
40dB gain, 20kHz bandwidth
Common Mode Rejection: >66dB @ 1kHz
Phantom Power:
48V
Frequency Response:
20Hz to 20kHz +0/-0.1dB
Connections
Input:
2 x XLR 3 pin female (Balanced)
Output:
2 x XLR 3 pin male (Balanced, can
be unbalanced)
Mains Input:
Filtered IEC, 110V-120V, or
220-240V switchable, fused, 9W
maximum
Fuse Rating:
Anti-surge fuse 100mA 20 x 5mm
(230VAC)
Anti-surge fuse 250mA 20 x 5mm
(115VAC)
Equipment Type
RB-MA2G:
Dual microphone amplifier
with gain
Physical Specifications
Dimensions (Raw):
48cm (W) x 10.8cm (D) x 4.2cm (H)
(1U)
19” (W) x 4.3” (D) x 1.7” (H) (1U)
Dimensions (Boxed):
58.5cm (W) x 22.5cm (D) x 7cm (H)
Weight:
Nett: 1.35kg Gross: 2.0kg
Nett: 3.0lbs Gross: 4.4lbs
23” (W) x 8.9” (D) x 2.8” (H)
137
Audio & Video Interfaces - Microphone Amplifiers & Limiters
RB-DMA2 Dual Digital Microphone Amplifier
Category: Microphone Amplifiers &
Limiters.
Product Function: Amplification of two
138
mic level signals to line level & AES/EBU
outputs
Typical Applications: To amplifiy
additional microphones in a studio
talks area, court room microphone
amplification, converting mic signal where
mic and amp are a long distance apart.
Features: See RB-ADDA & RB-MA2.
The RB-DMA2 consists of two independent
low-noise microphone pre-amplifiers
for converting microphone level signals
to a digital AES/EBU or S/PDIF output.
Individual analogue balanced line level
outputs are also produced for use, for
example, to feed talkback systems. The
unit can either be used as two independent
microphone amplifiers, or one mic input
can be copied to both channels of the
digital output.
Audio & Video Interfaces - Microphone Amplifiers & Limiters
The microphone inputs are XLR-3 type and
are electronically balanced. The input gain
for each input can be adjusted individually
by a volume control on the front panel
enabling the use of dynamic and powered
microphones and each has a LED level
indicator. For each channel there are
independent switches to control a high pass
filter (low frequency roll-off at 125Hz) and
to provide phantom power at +48V to the
connected microphones.
The RB-DMA2 has AES/EBU, S/PDIF and TTL
word clock sync inputs and has the same
sync modes, bit depth selection, channel
status bit adjustment, front panel LED
synchronisation and calibration routine as
the RB-ADDA. Please refer to that product
for further information.
Specification For RB-DMA2
Audio Specification
Input Level:
Max -25dBu, Min -62dBu to give FSD
/ŶƉƵƚ/ŵƉĞĚĂŶĐĞ͗ ϮŬɏŶŽŵŝŶĂů
Analogue
Outputs:
2 x XLR 3 pin male (balanced)
Digital Outputs:
1 x AES/EBU XLR 3 pin male (balanced),
1 x S/PDIF RCA phono socket
Mains Input:
Filtered IEC, switchable 110-120V,
or 220-240V, fused, 10W max
Fuse Rating:
Anti-surge fuse 100mA 20 x 5mm
(230VAC)
Anti-surge fuse 250mA 20 x 5mm
(115VAC)
Input Gain Range: 37dB
Low
125Hz @ 6dB/octave
Frequency Roll-off:
Signal To Noise:
130dB EIN
Dynamic Range:
>110dB
Equipment Type
Distortion
And Noise:
<0.01% THD + N absolute @ 1kHz
RB-DMA2:
Phantom Power: 48V
Connections
Dual digital microphone amplifier
Physical Specification
Dimensions
(Raw):
48cm (W) x 10.8cm (D) x 4.2cm (H) (1U)
19” (W) x 4.3” (D) x 1.7” (H) (1U)
Microphone
Inputs:
2 x XLR 3 pin female (balanced)
Dimensions
(Boxed):
58.5cm (W) x 22.5cm (D) x 7cm (H)
23” (W) x 8.9” (D) x 2.8” (H)
Sync Inputs:
1 x AES/EBU XLR 3 pin female
(balanced)
1 x S/PDIF RCA phono socket,
1 x TTL BNC female
Weight:
Nett: 1.6kg
Nett: 3.5lbs
Gross: 2.2kg
Gross: 4.8lbs
139
Audio & Video Interfaces - Microphone Amplifiers & Limiters
RB-ML2 Stereo Microphone & Line Level
Limiter
The RB-ML2 is a stereo microphone and line
Category: Microphone Amplifiers &
Limiters.
Product Function: To limit the maximum
output level for 2 x mic/line audio signals.
Typical Applications: To protect mixer
inputs, for mic inputs to a PC workstation,
to amplifiy additional microphones in a
studio talks area.
Features:
• Mic/line input & mic/line output levels.
• Mic input gain adjustment.
• +48V phantom power.
• High pass (rumble) filter.
level limiter. The unit is mainly used where
assistance with level control is required, for
protection of mixer inputs and to prevent
distortion. Ideal for news-booths, and the
input to PC workstations, it provides an
economical level control solution.
The RB-ML2 has two electronically balanced
XLR-3 inputs, which are routed to a line
amplifier, or microphone amplifier, via a
rear push-button. The microphone amplifiers
have independent pre-set gain controls,
and DIP switches for a high pass filter (low
frequency roll-off at 125kHz) and phantom
power to provide +48V to the connected
microphones.
The outputs of these amplifiers are passed
through a VCA limiter circuit that can operate
jointly on the signals in stereo mode, or
independently in dual mono mode. The
rear panel mode switch changes the unit
from dual mono to stereo. Stereo limiting
operates by limiting both left and right
outputs if either left or right input needs to
be limited. Dual mono limiting operates by
limiting left and right signals individually,
so you can use the RB-ML2 as two separate
mono mic/line limiters.
/ŶƉƵƚ
Impedance:
хϮϬŬɏďĂůĂŶĐĞĚďƌŝĚŐŝŶŐ
Input Gain
Range:
Adjustable +22dB to +67dB gain (mic)
via 2 x pre-set potentiometers (L & R),
0dB gain (line)
Maximum
Output Level:
-18dBu (mic), +28dBu (line),
electronically balanced
KƵƚƉƵƚ
Impedance:
фϱϬɏďĂůĂŶĐĞĚ
Output
Gain Range:
Adjustable -54dB to -24dB gain (mic),
-8dB to +22dB gain (line), ref 0dBu
line input, gain via 2 x pre-set pots
The characteristics of the limiter can
be set via level threshold pre-sets. For
each channel there is an input gain and
a threshold level control. With the limit
threshold set to maximum, the input
through to output can be normalised using
the input potentiometers. Once the unit
is acting as a buffer with gain/attenuation,
the limit threshold level can be set, with the
recovery adjusted for the application. The
power LED indicates limiting by flashing.
Limit Threshold: Adjustable -8dBu to +28dBu
Analogue
Inputs:
2 x XLR 3 pin female mic or line
switchable (balanced) (L & R)
The two XLR-3 electronically balanced
outputs can be set to either line or mic
output levels via a push-button. This allows
the RB-ML2 to be used in line with a line or
mic input on a mixer, or similar equipment.
Mic/Line
Control:
2 x push-buttons for mic/line inputs Input
Analogue
Outputs:
2 x XLR 3 pin male mic or line
switchable (balanced) (L & R)
Mains Input:
Filtered IEC, switchable 110-120V, or
220-240V, fused, 6W max.
Fuse Rating:
Anti-surge fuse 100mA 20 x 5mm (230VAC)
Anti-surge fuse 250mA 20 x 5mm (115VAC)
Specification For RB-ML2
Mic/Line
Output Control:
2 x push-button for mic/line outputs
Audio Specification
Phantom
1 x 4-way DIP switch
Power & LF Filter:
Maximum
Input Level:
-6dBu (mic), +28dBu (line),
electronically balanced
&ƌĞƋƵĞŶĐLJ
Response:
ϮϬ,njƚŽϮϬŬ,njцϬ͘ϭĚ;ϲϬϬɏůŽĂĚ͕
ref 1kHz)
Noise (RMS):
<-70dB unity gain, ref +8dBu output
Distortion:
< 0.02% THD + N @ 1kHz, ref +8dBu
output, threshold set at +10dBu
Common
>66dB typically
Mode Rejection (Line):
Common
>86dB typically
Mode Rejection (Mic):
Phantom Power: +48V
LF Filter:
125Hz @ 6dB/octave
Connections & Controls
Mono/Stereo
Mode Select:
1 x push-button
Limit Level
Threshold Set:
2 x pre-set potentiometers
Equipment Type
RB-ML2:
Stereo microphone limiter
Physical Specification
Dimensions
(Raw):
28cm (W) x 10.8cm (D) x 4.2cm (H) (1U)
11” (W) x 4.3” (D) x 1.7” (H) (1U)
Dimensions
(Boxed):
36cm (W) x 20.5cm (D) x 6cm (H
14.2” (W) x 8” (D) x 2.4” (H)
Weight:
140
Nett: 0.90kg
Nett: 2lbs
Gross: 1.35kg
Gross: 3lbs
Audio & Video Interfaces - Microphone Amplifiers & Limiters
RB-SL2 Twin Mono, Or Stereo, Limiter
Category: Microphone Amplifiers &
Limiters.
Product Function: To limit the maximum
output level for a stereo or twin mono
line level audio signal.
Typical Applications: Limiting program
output from a radio studio mixer, school
radio audio output limiter.
Features:
• Stereo or mono operation, can be used
as two mono limiters.
• Input gain & threshold limit controls.
The RB-SL2 is a stereo, or twin
independent mono, VCA limiter for use in
news-rooms and other locations where
the correct level into recording equipment
is required, but not necessarily under the
control of an engineer, for example, for
overload protection.
It can also be used as an inexpensive
main output limiter for small scale radio
stations, hospital radio and student radio.
The XLR-3 electronically balanced inputs
and outputs can be wired unbalanced to
accept an output from domestic equipment.
For each channel there is an input gain
and a threshold level control. With the
limit threshold set to maximum, the input
through to output can be normalised using
the input potentiometers. Once the unit
is acting as a buffer with gain/attenuation,
the limit threshold level can be set, with the
recovery adjusted for the application. The
power LED indicates limiting by flashing.
Specification For RB-SL2
The rear panel mode switch changes the
unit from dual mono to stereo, when only
the pre-sets for channel 1 (left) are active
and apply to both channels. Stereo limiting
operates by limiting both left and right
outputs if either left or right input needs to
be limited. Dual mono limiting operates by
limiting left and right signals individually,
so you can use the RB-SL2 as two separate
mono limiters.
Limit Threshold: Adjustable -8dBu to +28dBu
Audio Specification
Maximum
Input Level:
+28dBu
Maximum
Output Level:
+28dBu
Input Gain:
Adjustable -8dB to +18dB gain
&ƌĞƋƵĞŶĐLJ
Response:
ϮϬ,njƚŽϮϬŬ,njцϬ͘ϭĚ;ϲϬϬɏůŽĂĚ͕
ref 1kHz)
Noise:
-100dB unity gain, ref +8dBu output
Distortion:
0.01% THD @ 1kHz, ref +8dBu output,
threshold set at +10dBu
Common
>66dB typically
Mode Rejection:
/ŶƉƵƚ
impedance:
хϮϬŬɏďĂůĂŶĐĞĚďƌŝĚŐŝŶŐ
KƵƚƉƵƚ
impedance:
фϱϬɏďĂůĂŶĐĞĚ
Connections
Inputs:
2 x XLR 3 pin female (Balanced, can be
unbalanced)
Outputs:
2 x XLR 3 pin male (Balanced, can be
unbalanced)
Mains Input:
Filtered IEC, switchable 110-120V, or
220-240V, fused, 6W max
Fuse Rating:
Anti-surge fuse 100mA 20 x 5mm
(230VAC)
Anti-surge fuse 250mA 20 x 5mm
(115VAC)
Equipment Type
RB-SL2:
Twin mono, or stereo, limiter
Physical Specification
Dimensions
(Raw):
28cm (W) x 10.8cm (D) x 4.2cm (H) (1U)
11” (W) x 4.3” (D) x 1.7” (H) (1U)
Dimensions
(Boxed):
36cm (W) x 20.5cm (D) x 6cm (H
14.2” (W) x 8” (D) x 2.4” (H)
Weight:
Nett: 1.0kg Gross: 1.45kg
Nett: 2.2lbs Gross: 3.2lbs
141
Audio & Video Interfaces - Stereo To Mono Converters
RB-SM1 Single Stereo To Mono Converter
Category: Stereo to Mono Converters.
Product Function: To create a mono
output from a stereo source.
Typical Applications: To supply cue in
broadcast environments, to make a mono
feed for AM transmission, or to feed a
mono PA system.
Features:
• Neutrik XLR connectors.
• Balanced audio but can be wired
unbalanced.
• Output gain adjustment.
142
The RB-SM1 converts a stereo input to
a fully buffered and balanced mono line
output, e.g. to supply cue in broadcast
environments or to feed a mono PA
system.
Output gain adjustment using a pre-set
potentiometer for the converter allows a
normalised mono output from domestic
stereo equipment. This potentiometer is
accessible through the rear panel.
Distortion:
0.01% THD @ 1kHz, ref +8dBu output
Noise:
-100dB, unity gain, ref +8dBu output
The connections, which are on the rear
panel, are of an XLR-3 type. The input is
electronically balanced with an impedance
ŽĨϮϬŬɏďƌŝĚŐŝŶŐ͘dŚŝƐĐĂŶďĞǁŝƌĞĚ
unbalanced to accept an output from
domestic equipment.
The output is electronically balanced with
ĂŶŽƵƚƉƵƚŝŵƉĞĚĂŶĐĞŽĨфϱϬɏ͘dŚĞŽƵƚƉƵƚ
can be wired unbalanced by grounding the
non-phase signal, allowing you to feed both
balanced and unbalanced equipment.
Connections
Inputs:
2 x XLR 3 pin female (Balanced, can
be unbalanced)
Output:
1 x XLR 3 pin male (Balanced, can
be unbalanced)
Specification For RB-SM1
Mains Input:
Filtered IEC, switchable 110-120V,
or 220-240V, fused, 6W max
Audio Specification
Fuse Rating:
Anti-surge fuse 100mA 20 x 5mm
(230VAC)
Anti-surge fuse 250mA 20 x 5mm
(115VAC)
Maximum
Input Level:
+28dBu
Maximum
Output Level:
+28dBu
/ŶƉƵƚ
Impedance:
хϮϬŬɏďĂůĂŶĐĞĚďƌŝĚŐŝŶŐ
KƵƚƉƵƚ
Impedance:
фϱϬɏďĂůĂŶĐĞĚ
Dimensions
(Raw):
28cm (W) x 10.8cm (D) x 4.2cm (H) (1U)
11” (W) x 4.3” (D) x 1.7” (H) (1U)
&ƌĞƋƵĞŶĐLJ
Response:
ϮϬ,njƚŽϮϬŬ,njцϬ͘ϭĚ;ϲϬϬɏůŽĂĚ͕
ref 1kHz)
Dimensions
(Boxed):
36cm (W) x 20.5cm (D) x 6cm (H
14.2” (W) x 8” (D) x 2.4” (H)
Gain Range:
Adjust -8dB to +18dB gain, ref. 0dB
input on L and R
Weight:
Nett: 1.00kg Gross: 1.45kg
Nett: 2.2lbs Gross: 3.2lbs
Common
Mode Rejection:
>66dB typically
Equipment Type
RB-SM1:
Single stereo to mono converter
Physical Specification
Audio & Video Interfaces - Stereo To Mono Converters
RB-SM2 Dual Stereo To Mono Converter
The RB-SM2 is a dual version of the
RB-SM1, consisting of two independent
converters which will produce two fully
buffered and balanced mono line outputs
from two stereo inputs.
Category: Stereo to Mono Converters.
Product Function: To create two mono
outputs from two stereo sources.
Typical Applications: To supply cue in
broadcast environments, to make a mono
feed for AM transmission, or to feed a
mono PA system.
Features:
• Neutrik XLR connectors.
• Balanced audio but can be wired
unbalanced.
• Output gain adjustment.
All connections are on the rear panel. The
XLR-3 inputs are electronically balanced
ǁŝƚŚĂŶŝŵƉĞĚĂŶĐĞŽĨϮϬŬɏďƌŝĚŐŝŶŐ͘dŚĞƐĞ
can be wired unbalanced to accept an
output from domestic equipment.
by grounding the non-phase signal, allowing
you to feed both balanced and unbalanced
equipment.
Output gain adjustment using pre-set
potentiometers for both converters allows
a normalised mono output from domestic
stereo equipment. The potentiometers are
accessible through the rear panel.
Specification For RB-SM2
Audio Specification
The XLR-3 line outputs are electronically
balanced with an output impedance of
фϱϬɏ͘dŚĞŽƵƚƉƵƚƐĐĂŶďĞǁŝƌĞĚƵŶďĂůĂŶĐĞĚ
Gain Range:
Adjust -8dB to +18dB gain, ref. 0dB
input on L and R
Common
>66dB typically
Mode Rejection:
Distortion:
0.01% THD @ 1kHz, ref +8dBu output
Noise:
-100dB, unity gain, ref +8dBu output
Connections
Inputs:
4 x XLR 3 pin female (Balanced, can be
unbalanced)
Outputs:
2 x XLR 3 pin male (Balanced, can be
unbalanced)
Mains Input:
Filtered IEC, switchable 110-120V, or
220-240V, fused, 6W max
Fuse Rating:
Anti-surge fuse 100mA 20 x 5mm
(230VAC)
Anti-surge fuse 250mA 20 x 5mm
(115VAC)
Maximum
Input Level:
+28dBu
Maximum
Output Level:
+28dBu
/ŶƉƵƚ
Impedance:
хϮϬŬɏďĂůĂŶĐĞĚďƌŝĚŐŝŶŐ
Physical Specification
KƵƚƉƵƚ
Impedance:
фϱϬɏďĂůĂŶĐĞĚ
Dimensions
(Raw):
28cm (W) x 10.8cm (D) x 4.2cm (H) (1U)
11” (W) x 4.3” (D) x 1.7” (H) (1U)
&ƌĞƋƵĞŶĐLJ
Response:
ϮϬ,njƚŽϮϬŬ,njцϬ͘ϭĚ;ϲϬϬɏůŽĂĚ͕
ref 1kHz)
Dimensions
(Boxed):
36cm (W) x 20.5cm (D) x 6cm (H)
14.2” (W) x 8” (D) x 2.4” (H)
Equipment Type
RB-SM2:
Weight:
Dual stereo to mono converter
Nett: 1.05kg Gross: 1.50kg
Nett: 2.3lbs Gross: 3.3lbs
143
Audio & Video Interfaces - Power Controllers
RB-LC3 3 Way Light/Power Controller
Typical Applications: Controlling radio
studio On-Air, Mic-Live and Telephone
signal lamps.
Features:
• GPI to control switching of 3 IEC outlets.
• Telephony input for ‘Call Ringing’
Category: Power Controllers.
displays.
Product Function: To switch mains power to
3 x IEC outlets for mains voltage signal lamps. • Flash rate control.
The RB-LC3 is a triple output switching
unit for controlling external mains
indicators, primarily studio status lights for
broadcasting applications, such as On-Air,
Mic-Live and Rehearsal/Live lights.
Each output can be individually controlled
by one of three remote inputs (pulled high,
or low), by a telephony input (when ringing,
or off-hook, or both), or a combination of
two inputs (to control two outputs, e.g.
for Rehearsal/Live situations). The type of
control is set using the 12 way DIP switch
(4 switches for each output, allowing 16
different settings).
All connections are on the rear panel.
The three IEC outputs are controlled by
zero-cross point drivers. When an output
is activated, the A.C. voltage level at that
output will be equal to the mains input
voltage used to power the unit.
External control of the switched mains
outputs is via the 15 way D-type plug
connector.
ringing and off-hook conditions can be
indicated.
A pre-set potentiometer on the rear panel
controls the flash rate of the output when
the appropriate mode is selected. Neon
indicators on each power socket show the
status of the mains output.
Specification For RB-LC3
Audio Specification
The telephone line input and handset
output are via two RJ11-4 type connectors.
The telephone connections are wired pin to
pin from line to handset except when the
remote ‘Ring Mute’ control input is asserted.
In this case the ring signal to the handset
is muted.
The status of the telephone line is
continually monitored so that handset
Connections
Mains Input:
Non-filtered IEC, 110V-240V autoadjusting, fused, 6W maximum
Mains Outputs:
3 x Non-filtered IEC plugs, 1A fused
Fuse Rating
(Mains Outputs):
3 x Anti-surge fuse 1A 20 x 5mm
Telephone:
2 x RJ11-4 sockets
Control Inputs
& Outputs:
15 way D-type plug
Inputs: 0V- 5V DC
Outputs: Open collector 20mA sink
capability
Equipment Type
RB-LC3:
3 way light/power controller
Physical Specification
Dimensions
144
(Raw):
28cm (W) x 10.8cm (D) x 4.2cm
(H) (1U)
11” (W) x 4.3” (D) x 1.7” (H) (1U)
Dimensions
(Boxed):
36cm (W) x 20.5cm (D) x 6cm (H)
14.2” (W) x 8” (D) x 2.4” (H)
Weight:
Nett: 1.00kg Gross: 1.45kg
Nett: 2.2lbs Gross: 3.2lbs
Audio & Video Interfaces - Talkback & Communications
RB-MM1 Mix-Minus Generator
Category: Talkback & Communications.
Product Function: To generate a suitable
mix to send to a telephone hybrid or
codec from a mixer.
Typical Applications: Used with mixers
without mix-minus busses.
Features:
• Neutrik input and output connectors.
• Output level adjustment.
• Cancellation null adjustment.
• Band pass filter.
Whenever programming originates from
outside of the studio, or if listeners/viewers
are calling a phone-in using telephone
hybrids or codecs, mix-minus feeds are
required. Most telephone lines incur delays
which prohibit off-air monitoring, because
the caller, or remote talent, would hear
their own voice in delay which is very
disconcerting. The solution is to feed a mix
back to the caller minus his or her own
voice. Some mixing desks do not have a
dedicated telco channel to generate a
clean-feed, or mix minus, so the RB-MM1
can be used.
The RB-MM1 is a unit for generating a
suitable mix to send to a telephone hybrid
or codec. A stereo output is taken from a
mixer, together with a post fader output
Specification For RB-MM1
from the mono telephone fader on the
mixer. The caller audio is removed from the
station output so that it can be sent to the
telephone line via the hybrid.
Analogue audio inputs and outputs are via
Neutrik XLR connectors. The output level
to the TBU can be adjusted using a rear
panel pre-set potentiometer. To control
the cancellation null (the amount of the
telephone channel which is subtracted
from the mixer output signal), 2 multiturn potentiometers are provided, one for
the LF null and the other for the full-band
null. Additionally, a band pass filter can be
switched in and out, via a rear panel switch,
to condition the signal for the telephone
hybrid. To use the RB-MM1 unit with
full-band ISDN codecs, the band pass filter
can be switched out. For stereo codecs, or
conference calls, multiple RB-MM1 units
can be used.
Audio Specification
Maximum
Input Level:
+28dBu
/ŶƉƵƚ/ŵƉĞĚĂŶĐĞ͗
хϮϬŬɏ
Maximum
Output Level:
+28dBu
KƵƚƉƵƚ/ŵƉĞĚĂŶĐĞ͗ фϱϬɏ
Output Gain Range: Adjustable -15dB to +12dB, ref 0dBu
gain via a multi-turn pot
Common
Mode Rejection:
>60dB
Band Pass
Filter Range:
200Hz to 4kHz, 12dB/octave
LF Null
Adjustment:
Better than 40dB at 100Hz
Mix-Minus Null:
Better than 40dB at 1kHz
Frequency
Response:
20Hz - 22kHz ± 0.1dB
Distortion:
0.01% THD @ 1kHz, ref +8dBu
output (C-Message weighted)
Noise:
-90dBu unity gain +6dBu
Connections
Analogue Inputs
2 x XLR 3 pin female (balanced) (L & R)
From Mixer Output:
Analogue Input
1 x XLR 3 pin female (balanced) (L & R)
From Telephone Fader:
Analogue
Output To TBU:
1 x XLR 3 pin male (balanced) (L & R)
Mains Input:
Filtered IEC, switchable 110-120V,
or 220-240V, fused, 6W max.
Fuse Rating:
Anti-surge fuse 100mA 20 x 5mm
(230VAC)
Anti-surge fuse 250mA 20 x 5mm
(115VAC)
Equipment Type
RB-MM1:
Mix-minus generator
Physical Specification
Dimensions
(Raw):
28cm (W) x 10.8cm (D) x 4.3cm (H)
11” (W) x 4.3” (D) x 1.7” (H)
Dimensions
(Boxed):
36cm (W) x 20.5cm (D) x 6cm (H)
14.2” (W) x 8” (D) x 2.4” (H)
Weight:
Nett: 1.1kg Gross: 1.5kg
Nett: 2.4lbs Gross: 3.3lbs
145
Audio & Video Interfaces - Talkback & Communications
RB-MTV1 Contribution Voiceover Monitor With Talkback
Category: Talkback & Communications.
Product Function: Mixes a mic, talkback
and two stereo inputs to headphone
output and two line outputs.
Typical Applications: In voice-over booths,
news booths, commentary locations, for
continuity announcements and for any
other similar applications where voice
needs to be added to programme content
and then monitored, with talkback.
146
Features:
• Wide mic input gain range.
• High pass filter & +48V phantom power.
• Switchable level limiter with threshold
control.
• Mic level metering.
• ‘Lazy’ talkback input/output & button.
• Two stereo inputs, which can be balanced
or unbalanced (+10dB pad).
• Front panel headphone output with a
volume control for each input.
• Headphone monitoring configured via
DIPswitches.
• Two mic/line outputs.
• Remote control of MIC and TALK buttons.
The RB-MTV1 contribution voiceover
monitor is a 1U rack-mount designed to be
used in voice-over booths, news booths,
commentary locations, for continuity
announcements and for any other similar
applications where voice needs to be
added to programme content and then
monitored. Programme feeds, auxiliary
feeds and a talkback feed can be taken and
monitored.
The RB-MTV1 has four inputs and two
outputs. It has a mono microphone input
on XLR with switched coarse gain and
variable fine gain control using a multi-turn
preset potentiometer to give an overall gain
range from +20dB to +80dB. There is also
a switched LF rumble filter, switched +48V
phantom power and switched level limiting
control.
Audio & Video Interfaces - Talkback & Communications
A rear-panel multi-turn preset
potentiometer allows adjustment of the
threshold at which the limiter begins to
operate, from -8dBu to +26dBu. There is an
indication of the limiter activity using a blue
LED and the microphone level is monitored
by a simple 5 LED meter. The meter can be
configured to either show the MIC signal
activity in normal operation, i.e. when TALK
is pressed the meter is off, or the meters
can permanently show the MIC activity
even when the TALK button is on or the MIC
button is off.
There is a mono balanced TALKBACK input
on XLR. There are two balanced XLR stereo
inputs, CUE and programme (PGM) each
with a 10dB input gain switch to facilitate
the use of unbalanced sources such as from
PC audio cards, domestic CD players, etc.
Each of these inputs, MIC, TALKBACK, CUE
and PGM can be mixed and monitored
in the front and rear headphone outputs
+48V
Mic Gain
Mic
Input
Remotes
There are two mono balanced outputs. The
processed microphone signal is fed to the
two outputs when the latching front panel
MIC button is active - the button illuminates
when active.
Lazy
Output
Limit LED
Mic Limit
LF +20dB On
Filter Gain Off
T/B
Input
with individual volume controls. Which of
the inputs is presented to the headphone
outputs can be configured using two banks
of DIPSwitches on the underside of the unit,
one bank for audio that is heard in the left
ear piece and one bank for audio monitored
in the right ear piece. For example, the left
and right PGM inputs can be sent to the
right ear-piece and all other signals to the
left ear-piece, or the TALK signal could be
sent to the right ear-piece and all other
signals to both ear-pieces. In this way, you
can configure the unit for your particular
application on installation. If a presenter
doesn’t like to hear themselves in their
headphones when using the TALK button,
there is also a DIPswitch option to mute the
mic signal to the headphones.
Mic/
Line
Mic/
Line
Mic Meter
Output 1
Output 2
Headphone
Front
L
Cue
R
+10dB
Left
Headphone
Mix
L
Headphone
Rear
PGM
R
+10dB
Right
Headphone
Mix
The two main outputs can be independently
switched to be at ‘Line’ or ‘Mic’ level
outputs using rear panel push switches.
Setting the output to a microphone level
allows the unit to be inserted into the
microphone channel of a mixer.
Specification For RB-MTV1
There is an option to permanently enable
the MIC button, even when remotely
controlled, for occasions when you
always want the MIC channel left open.
Additionally, there is an option to mix the
CUE input as a mono feed to the outputs
permanently.
Common Mode Rejection: >60dB typically
The front panel TALK button is a momentary
push switch that routes the processed
microphone signal to the “LAZY” TALK
output, whilst disconnecting it from the
main outputs, allowing the operator to talk
to a colleague. This enables the unit to be
used as a talkback intercom between two or
more studios.
There is a rear panel remotes connector
giving remote control of the two front panel
MIC and TALK buttons and opto-isolated
tallies of their status.
A red LED on the front panel of the unit
indicates when power is present.
Audio Specification
/ŶƉƵƚ/ŵƉĞĚĂŶĐĞ͗
ϮŬёŶŽŵŝŶĂůďĂůĂŶĐĞĚD/ŝŶƉƵƚ
хϮϬŬёĂůůŽƚŚĞƌŝŶƉƵƚƐ͘
Maximum Input Level:
-10dBu MIC input
+26dBu TALK, CUE & PGM inputs
Mic Gain Range:
Adjustable +20dB to +80dB
E.I.N.:
130dB
Low Frequency Roll-Off:
125Hz @ 6dB/octave
Distortion:
0.01% THD @ 1kHz, ref +8dBu o/p
Phantom Power:
48V
Limiter Threshold:
Adjustable +8dBu to +28dBu
KƵƚƉƵƚ/ŵƉĞĚĂŶĐĞ͗
фϱϬё
Maximum Output Level:
+26dBu balanced outputs
,ĞĂĚƉŚŽŶĞKƵƚƉƵƚ
Level:
ƌŝǀĞƐϭϱϬŵtŝŶƚŽϯϮёƚŽϲϬϬё
headphones
Frequency Response:
20Hz to 20KHz ±0.1dB (ref 1KHz)
Rear Panel Connections
Mic Input:
1 x XLR 3 pin female (Balanced)
TALKBACK Input:
1 x XLR 3 pin female (Balanced)
CUE Inputs:
2 x XLR 3 pin female (Balanced)
PGM Inputs:
2 x XLR 3 pin female (Balanced)
Output 1:
1 x XLR 3 pin male (Balanced)
Output 2:
1 x XLR 3 pin male (Balanced)
TALK (LAZY) Output:
1 x XLR 3 pin male (Balanced)
Headphone Outputs:
2 x ¼” (6.35mm) A-gauge 3-pole
stereo jack sockets
Remote I/O Port:
9-way ‘D’-type socket
Mains Input:
Filtered IEC, 110V-120V, or 220240V switchable, fused, 9W max
Fuse Rating:
Anti-surge fuse 100mA 20 x 5mm
(230VAC)
Anti-surge fuse 250mA 20 x 5mm
(115VAC)
Equipment Type
RB-MTV1:
Contribution voiceover unit with
talkback
Physical Specification
Dimensions
(H) (Raw):
48cm (W) x 10.8cm (D) x 4.3cm
19” (W) x 4.3” (D) x 1.7” (H) (1U)
Dimensions
(Boxed):
58.5cm (W) x 22.5cm (D) x 7cm(H)
23” (W) x 8.9” (D) x 2.8” (H)
Weight :
Nett: 1.3kg Gross: 1.9kg
Nett: 2.9lbs Gross: 4.2lbs
Accessories
RB-RK3:
1U Rear panel rack kit for large
Redboxes
RB-MTV1 Block Diagram.
147
Audio & Video Interfaces - Talkback & Communications
RB-IPE IP Extender for GPIO & Analogue Control Signals
Category: Talkback & Communications.
Product Function: Provides remote control
and telemetry of GPIO and analogue
control voltages over an Ethernet network.
Typical Applications: It allows any
tallies & control signals, together with
analogue potentiometer movements, to
be sent across a network, e.g. for remote
alarm points, to trigger failure alarms at
transmitter sites and to control remote
equipment at unmanned posts, outstations
or transmitter sites.
Features:
• 8 x isolated current sink inputs.
• 8 x pull to ground protected inputs.
• 8 x isolated relay change-over output
contacts.
• 8 x opto-isolated output contacts.
• 8 x 0 to 3.3V/5V/12V input signals.
• 8 output signals nominally at 0 to 3.3V.
• Two units can act as master and slave.
• Input & output voltages can be
mapped.
• Webserver configuration and control.
The RB-IPE is a 1U rackmount unit designed
to provide remote control of GPIO and
analogue control voltages over an Ethernet
network. Configured using a built-in web
server, two units can control each other
across an Ethernet network, or a single unit
can be controlled via Ethernet commands
and the web server interface.
The unit can be used in any position where
you need to remotely acquire GPO signals
or remotely control equipment, for example
controlling equipment at unmanned posts,
outstations or transmitter sites.
Each unit has 16 x general purpose inputs on
8 x RJ45 connectors, consisting of 8 x isolated
current sink inputs and 8 x pull to ground
148
Audio & Video Interfaces - Talkback & Communications
protected inputs; and 16 x general purpose
outputs on 8 x RJ45 connectors using 8 x
isolated relay change-over contacts and 8
x opto-isolated contacts. These rear panel
RJ45 connectors have an LED for each GPIO
which shows its state.
On another 8 x RJ45 connectors there are
also 8 x 0 to 3.3V/5V/12V input signals and 8
output signals nominally at 0 to 3.3V output,
with other output voltage configurations
possible. The outputs can all be controlled
from the inputs of another RB-IPE, or from
Ethernet commands.
This allows any tallies and control signals,
together with analogue potentiometer
movements, to be sent across a network,
e.g. for remote alarm points, to trigger
failure alarms at a transmitter site and to
control remote equipment.
When two units are connected together at
different sites, if a general purpose input
state changes at one site the unit sends
the new state to the other site and the
appropriate opto-isolator output changes on
that unit. Similarly input voltage controls are
monitored and the changing voltage is sent
to the remote unit where an output voltage
changes accordingly.
The signals can be routed and distributed
such that a single input signal to a unit on
one site can be routed to multiple outputs in
a unit on another site and/or have the logic
inverted and distributed to multiple outputs.
Also, the state of the GPOs when the unit
is powered on can be configured, allowing
more reliable recovery of external connected
equipment from a power-fail condition.
The analogue I/O control signals can be
mapped to give different ranges between the
incoming and outgoing signal, e.g. 0V to 5V
input giving a 0V to 12V output, or a linear
input mapped to a log scale output. Also,
by programming threshold values, analogue
input voltages can be mapped to GPO pins,
e.g. for sending a signal to a GPO when a
volume knob is turned too high.
The web server in the RB-IPE can be
configured with a static IP address or by
using DHCP.
The three front panel green LEDs give an
indication of Ethernet connectivity, i.e. they
show when commands are being sent/
received. The CONNECTED LED shows link
status, the GPIO LED illuminates whenever a
GPIO state changes and the ANALOGUE LED
indicates a change in state of the analogue
voltage signals.
The RB-IPE is powered from a universal
mains input between 85-264V AC at
47-63Hz.
RB-IPE Home Page.
RB-IPE Configuration of Analogue Inputs Page.
Specification For RB-IPE
Maximum collector/emitter voltage
peak: 35V DC @ 7mA
Maximum collector/emitter current:
80mA @ 2.5V DC
(Note: There is a 200 mA fused +5V
power supply available on GPI ports
1 – 8 and GPO ports 9 – 16.)
Audio Specification
Rear Panel Connections
Isolated GPI:
4 x RJ45 sockets, with LED status
indicator per input
Active Low GPI:
4 x RJ45 sockets, with LED status
indicator per input
Relay GPO:
4 x RJ45 sockets, with LED status
indicator per output
Isolated GPO:
4 x RJ45 sockets, with LED status
indicator per output
Analogue Control
Inputs:
4 x RJ45 sockets
Analogue Control
Outputs:
4 x RJ45 sockets
Ethernet Port:
RJ45 with status LEDs
Mains Input:
Filtered IEC, continuously rated
85-264VAC @ 47-63Hz, 10W max
Fuse Rating:
Anti-surge fuse 1A 20 x 5mm
Input & Output Detail
Analogue Control:
Inputs:
8 x 0V-3.3V, 5V or 12V input signals
Analogue Control
Outputs:
8 x output signals, nominally
0V-3.3V, 5V or 12V
Front Panel Indicators
Power On:
Red LED
CONNECTED:
Green link status LED
GPIO:
Green GPIO change status LED
ANALOGUE:
Green analogue control I/O
change status LED
Equipment Type
RB-IPE:
IP extender for GPIO & analogue
control signals
Physical Specification
General Purpose
Inputs:
8 x isolated current sink inputs from
3.3V to +24V (Max input range:
0V to +24V)
8 x pull to ground protected inputs
(Max input range -24V to +24V)
General Purpose
Outputs:
8 x isolated relay change-over
contacts:
Nominal switching capacity
(resistive load): 1A @ 30V DC (0.5A
@ 125V AC)
8 x opto-isolated contacts:
Dimensions (Raw):
48cm (W) x 10.8cm (D) x 4.2cm (H)
(1U)
19” (W) x 4.3” (D) x 1.7” (H) (1U)
Dimensions (Boxed):
58.5cm (W) x 22.5cm (D) x 7cm (H)
23” (W) x 8.9” (D) x 2.75” (H)
Weight:
Nett: 1.6kg Gross: 2.2kg
Nett: 3.5lbs Gross: 4.8lbs
Accessories
RB-RK3
1U Rear panel rack kit for large
Redboxes
149
Audio & Video Interfaces - Tone Generators
RB-TGHD(B or X) Multi-Channel HD Tone Generator
Category: Tone Generators.
Product Function: 8 Channel audio tone
generator that provides line identification
for multi-channel audio systems.
Typical Applications: It allows correct
channel configuration in fold-down mixes
when you need to mix several audio
channels into a stereo feed, e.g. 5.1 and 7.1
surround sound.
Features:
• 8 analogue & 8 digital output channels.
• EBU R49, GLITS and BLITS tones, with
channel identification.
•
•
•
•
2, 4, 6 and 8 channel configurations.
Wordclock sync input.
Optional sync boards available.
Audio level line-up adjustable from
0dBu to +24dBu.
• Available with BNC (B) or XLR (X) digital
audio outputs.
• Programmable audio tone sequences
using SCi serial remote.
The RB-TGHD is an 8 channel audio tone
generator that provides line identification
for multi-channel audio systems, including
5.1 and 7.1 surround sound typically used
in high definition television broadcasts. By
using a range of widely accepted industry
standard tone sequences such as the
EBU R49, GLITS and BLITS tones, channel
identification and associated levels can be
determined easily.
Correct channel configuration in folddown mixes can also be highlighted when
a broadcaster needs to mix several audio
channels into a stereo feed.
192
RB-TGHDB Multi-Channel HD Tone Generator.
150
Audio & Video Interfaces - Tone Generators
192
RB-TGHDX Multi-Channel High Definition Tone Generator.
The RB-TGHD caters for 2, 4, 6 and 8
channel configurations and all of the
available audio tone sequences for each
channel configuration can be cycled through
automatically, or selected manually, and
a loop mode allows patterns of tones to
be repeated. A bank of 4 pushbuttons on
the front panel sets these options and the
associated LEDs indicate the current setting.
A set of 8 LEDs on the front panel indicate
which channel is currently outputting audio
with the LEDs numbered as both 1-8 and as
L, R, C, LFE, LS, RS, LR and RR.
The RB-TGHD is available in two variations,
each providing both analogue and digital
audio outputs. The RB-TGHDX offers
balanced AES/EBU digital audio outputs on
3 pin XLR connectors and the RB-TGHDB has
unbalanced digital audio outputs on BNC
connectors. Both types provide 8 balanced
analogue outputs on 3 pin XLR connectors.
A word clock synchronising input on BNC
connector allows the digital outputs to
be synchronised to an external signal.
Optionally, an RB-SYA, RB-SYD or RB-SYE
sync board can be fitted, to synchronise the
unit to either an analogue or digital video
sync signal, or an AES/EBU audio input
respectively. A SYNC LED on the front panel
indicates when the unit is synchronised to
the external signal.
The audio level line-up of the RB-TGHD
can be adjusted from 0dBu to +24dBu
in 1dBu steps (ref FSD) to suit different
environments via the front panel mounted
DIPSwitches. Changing the audio line-up
does not affect the gain relationship
151
Audio & Video Interfaces - Tone Generators
Specification For RB-TGHD(B & X)
between each channel, ensuring that
correct levels through the target system can
be maintained. These switches also provide
settings for digital sample rate which ranges
from 32 kHz to 192 kHz, and digital sample
width which is either 16 bit or 24 bit.
The serial port allows the RB-TGHD to
be connected to a PC running SCi, the
Sonifex Serial Control Interface. This allows
full control of the unit and the ability
to generate a user defined audio tone
sequence.
ŶĂůŽŐƵĞKƵƚƉƵƚ Impedance:
фϱϬё ŝŐŝƚĂůKƵƚƉƵƚ
/ŵƉĞĚĂŶĐĞ͗
ϭϭϬёďĂůĂŶĐĞĚ
ϳϱёƵŶďĂůĂŶĐĞĚ
4 x AES/EBU XLR 3 pin male
(balanced) or
4 x BNC female (un-balanced)
Digital Input:
Word clock on BNC
SCi Control Page.
25-way ‘D’-type socket
17 inputs, 7 tally outputs
>100dB
Serial Comms Port:
9-way ‘D’-type socket
Max Output Level:
+24dBu
Mains Input:
Distortion & Noise:
<-85dB THD+N at 1kHz
Crosstalk:
<-110dB (20Hz to 20kHz)
for analogue outputs
Filtered IEC, continuously
rated 85-264VAC,
47-63Hz, 60W peak,
30W average
Fuse Rating:
Anti-surge fuse 1A 20 x 5mm
Equipment Type
Channels:
2, 4, 6 or 8
Sequence Mode:
Auto or Manual
Sequence:
EBU R49 stereo line-up
GLITS stereo line-up
BLITS stereo line-up
EBU R49 channel ID
BLITS channel ID
Phase
User defined (using SCi)
Sequence Loop Mode:
On or Off (enables looping of
current sequence)
Digital Sample
Frequency:
32kHz, 44.1kHz, 48kHz, 88.2kHz
96kHz, 176.4kHz or 192kHz
(via DIPSwitches)
Digital Sample Width:
16bit or 24bit (via DIPSwitches)
Audio Line-Up:
Channel Identification:
SCi Miscellaneous Page.
Analogue Outputs:
RB-TGHDB:
Multi-channel HD tone generator
with BNC digital outputs
RB-TGHDX:
Multi-channel HD tone generator
with XLR digital outputs
Physical Specification
Dimensions
(Raw):
48cm (W) x 10.8cm (D) x 4.3cm (H)
19” (W) x 4.3” (D) x 1.7” (H) (1U)
Dimensions
(Boxed):
59cm (W) x 27.5cm (D*) x 11cm (H)
23.2” (W) x 10.8” (D*) x 4.3” (H)
Weight :
Nett: 1.3kg
Nett: 2.9lbs
Gross: 1.9kg
Gross: 4.2lbs
Accessories
RB-SYA:
Analogue video sync board
(NTSC, PAL & SECAM)
0dBu to +24dBu in 1dB steps ref
FSD (via DIPSwitches)
RB-SYD:
Digital video sync board
(SD-SDI & HD-SDI)
LEDs indicating 1-8 and L, R, C,
LFE, LS, RS, LR and RR
RB-SYE:
AES/EBU sync board
RB-RK3:
1U Rear panel rack kit for large
Redboxes
Rear Panel Connections
SCi User Sequence Page.
Remote I/O Port:
Dynamic Range:
Front Panel Controls
In addition, a remote port on the rear
provides a simple interface to control the
unit and has several outputs to indicate
which tone sequence is active.
152
Digital Outputs:
Audio Specification
8 x XLR 3 pin male
(balanced)
Portable Packs - Audio Distribution Amplifiers
CM-AESX3 Single 3 Way AES/EBU Passive Splitter With XLR Connectors
Category: Audio Distribution Amplifiers.
Product Function: 1 input, 3 output AES3
distribution amplifier.
Typical Applications: Distribution of AES3
source to 3 mixing consoles.
Features: No power needed, can apply
ϭϭϬёƚĞƌŵŝŶĂƚŝŽŶƚŽƵŶĐŽŶŶĞĐƚĞĚ
outputs, transformer based solution
provides reliability.
CM-AESX3 Front & Rear View.
The CM-AESX3 is a passive “one-to-three”
splitter housed in a small but robust
aluminium box. It splits a single AES3 digital
audio source to up to three destinations,
using Neutrik XLR connectors.
The CM-AESX3 solves a range of digital
signal distribution problems, where correct
termination is essential to maintain signal
integrity. The input signal is split through high
ƋƵĂůŝƚLJƚƌĂŶƐĨŽƌŵĞƌƐĂŶĚϭϭϬёƚĞƌŵŝŶĂƚŝŽŶ
can be applied, to unconnected outputs.
The CM-AESX3 requires no power to operate,
ensuring your audio remains connected from
source to destination(s) without interruption
from power failures.
Specification For CM-AESX3
CM-AESX3 Technical Specification
Connections
CM-AESX3: Cable Drive Capability
Inputs:
1 x XLR 3 pin female
The table below sets out the minimum signal amplitude
ƌĞƋƵŝƌĞĚƚŽĚƌŝǀĞϭϬϬŵ;ĐƵŵƵůĂƚŝǀĞͿŽĨϭϭϬёƚǁŝƐƚĞĚƉĂŝƌ
cable, based on the sample rate of the digital audio:
Outputs:
3 x XLR 3 pin male
Equipment Type
CM-AESX3
Single 3 way AES/EBU passive splitter
with XLR connectors
Sample Rate
Minimum Signal Amplitude
32kHz
* 2Vpk-pk
Physical Specification
44.1kHz
* 2Vpk-pk
48kHz
* 2Vpk-pk
Dimensions
(Raw):
7.7cm (W) x 8.3cm (D) x 4.2cm (H)
3.0” (W) x 3.3” (D) x 1.7” (H)
Dimensions
(Boxed):
22.9cm (W) x 12.7cm (D) x 7.6cm (H)
9.0” (W) x 5.0” (D) x 3.0” (H)
Weight:
Nett: 0.22kg Gross: 0.33kg
Nett: 0.49lbs Gross: 0.73lbs
88.2kHz
5Vpk-pk
96kHz
5Vpk-pk
* Minimum of 2Vpk-pk is defined by the AES3 format
specification.
The table below sets out the minimum signal amplitude
ƌĞƋƵŝƌĞĚƚŽĚƌŝǀĞϯϬŵ;ĐƵŵƵůĂƚŝǀĞͿŽĨϭϭϬёƚǁŝƐƚĞĚƉĂŝƌ
cable, based on the sample rate of the digital audio:
Sample Rate
Minimum Signal Amplitude
176.4kHz
3Vpk-pk
192kHz
3Vpk-pk
153
Portable Packs - Audio Distribution Amplifiers
CM-AESB3 Single 3 Way Passive AES3ID Splitter With BNC Connectors
Category: Audio Distribution Amplifiers.
Product Function: 1 input, 3 output AES3
distribution amplifier.
Typical Applications: Distribution of AES3
source to 3 mixing consoles.
Features: No power needed, can apply
ϭϭϬёƚĞƌŵŝŶĂƚŝŽŶƚŽƵŶĐŽŶŶĞĐƚĞĚ
outputs, transformer based solution
provides reliability.
The CM-AESB3 is a passive “one-to-three”
splitter housed in a small but robust
aluminium box. It is designed to split a
single AES3ID digital audio source to up
to three destinations, using female BNC
connectors.
CM-AESB3: Cable Drive Capability
ƵŵƵůĂƚŝǀĞĐĂďůĞĚƌŝǀĞĐĂƉĂďŝůŝƚLJŽĨϭϬϬŵŽĨϳϱёĐŽĂdžŝĂů
cable at sample rates up to and includin g 96kHz.
Connections
Inputs:
1 x BNC female
Outputs:
3 x BNC female
Equipment Type
Particularly useful in a video production
and broadcast environment, the CM-AESB3
splits the input signal through high quality
ƚƌĂŶƐĨŽƌŵĞƌƐ͘ϳϱёƚĞƌŵŝŶĂƚŝŽŶĐĂŶďĞ
applied, if desired, to unconnected outputs
to maintain optimum carrier parameters.
The CM-AESB3 requires no power to
operate, ensuring your audio remains
connected from source to destination(s)
without interruption from power failures.
CM-AESB3 Front & Rear View.
154
Specification For CM-AESB3
CM-AESB3
3 Way passive AES3ID splitter with BNC
connectors.
Physical Specification
Dimensions
(Raw):
7.7cm (W) x 8.3cm (D) x 4.2cm (H)
3.0” (W) x 3.3” (D) x 1.7” (H)
Dimensions
(Boxed):
22.9cm (W) x 12.7cm (D) x 7.6cm (H)
9.0” (W) x 5.0” (D) x 3.0” (H)
Weight:
Nett: 0.22kg Gross: 0.33kg
Nett: 0.49lbs Gross: 0.73lbs
Portable Packs - Audio Distribution Amplifiers
CM-MS3 Single 3 Way Passive Microphone Splitter
Category: Audio Distribution Amplifiers.
Product Function: To split a microphone
or line level signal to 3 independent
outputs.
The CM-MS3 is a passive “one-to-three”
splitter housed in a small but robust
aluminium box. It is designed to split a
single microphone or line source to up
to three destinations, using Neutrik XLR
connectors.
Wide Signal Range Capability
The CM-MS3 uses high quality audio
transformers that are capable of accepting
input levels up to +18dBu, making the
splitter useful in both microphone and line
level splitting applications.
Versatile Phantom Powering Options
The CM-MS3 offers three methods of
providing phantom power to a microphone
connected to its input:
• From the external +48VDC power
connector.
• From a microphone amplifier connected
to output 1 (direct)
Typical Applications:
To share microphones between radio/TV/
production studios.
Features:
• Passive transformer design.
• Ground lift switch.
• External or looped phantom power
inputs.
• External +48V phantom power input.
• From a microphone amplifier
connected to output 2 (phantom
loopback)
Controlling which method is used to
provide phantom power is achieved with
simple push switches. It is possible to
concurrently power a microphone using
any two of the above methods without
degrading audio performance, thus
providing power supply redundancy.
Ground Loop Hum Elimination
Ground loop hum problems can be quickly
eliminated using the push switches to lift
pin 1 of the output connectors (output 2
and 3 only).
Specification For CM-MS3
Connections
Test
Conditions
Results
Inputs:
Frequency
Response:
Ref. -6dBu, 1kHz
Source impedance = 150?
Load impedance = 10k?
10Hz 30kHz
±0.5dB
1 x XLR 3 pin female (balanced, can
be unbalanced)
Outputs:
3 x XLR 3 pin male (balanced, can
be unbalanced)
Total Harmonic
Distortion:
Ref. +3dBu, 50Hz
Source impedance = 150?
Load impedance = 10k?
0.02%
DC Phantom
Power Input:
Total Harmonic
Distortion 0.1%:
Ref. 0.1%THD+N, 50Hz
Source impedance = 150?
Load impedance = 10k?
+13dBu
Total Harmonic
Distortion 1%:
Ref. 1%THD+N, 50Hz
Source impedance = 150?
Load impedance = 10k?
+18dBu
Common Mode
Rejection Ratio:
Ref. 20kHz
Source impedance = 600?
Load impedance = 10k?
>60dB
+48V DC, 2.5mm socket
Equipment Type
RB-MS3
Single 3 way passive microphone
splitter
Physical Specification
Dimensions
(Raw):
7.7cm (W) x 8.3cm (D) x 4.2cm (H)
3.0” (W) x 3.3” (D) x 1.7” (H)
Dimensions
(Boxed):
22.9cm (W) x 12.7cm (D) x 7.6cm (H)
9.0” (W) x 5.0” (D) x 3.0” (H)
Weight:
Nett: 0.30kg Gross: 0.40kg
Nett: 0.66lbs Gross: 0.88lbs
155
156
Professional PCIe Sound Cards &
Radio Capture Cards
Professional PCIe Sound Cards
PC-AD2 2 Stereo Analogue I/O & 2 Stereo AES-3 I/O PCIe Half Height Sound Card
158
158
PC-DIG4 Digitorc 4, 4 Stereo AES-3 I/O PCIe Sound Card
161
PC-AUR44 Auricon 4.4 PCIe Analogue Sound Card
162
Radio Capture Cards
PC-DAB1-4 Multi-Ensemble DAB+/DAB Radcap PCle Card
163
163
PC-FM6-32 FM Radcap PCle Card (6 to 32 Channels)
164
PC-AM6-32 AM Radcap PCle Card (6 to 32 Channels)
165
157
Professional Sound Cards & Radio Capture Cards
Professional Sound Cards
These professional sound cards and radio capture cards were precision engineered in Australia.
Use of the highest quality components & excellent electronic design give these cards the flatest
frequency response in the business.
PC-AD2 2 Stereo Analogue I/O & 2 Stereo AES-3 I/O PCIe Half Height Sound Card
Category: Professional Sound Cards &
Radio Capture Cards.
Product Function: Provides two AES-3
and two balanced analogue audio inputs
and outputs for use in a PC.
Typical Applications: Audio
workstations, recording studios,
automation systems, audio logging,
multi-channel playout.
Features:
• 2 independent transformer-coupled
AES-3 inputs and outputs.
• 2 independent high level balanced
stereo inputs and outputs.
158
• One of the stereo line inputs can
act as a dual mono or stereo mic
input with +48V phantom power and
limiter.
• 1 AES-11 synchronisation input.
• 2 opto isolated GPIOs.
• 24-bit audio resolution.
• Sampling rates up to 192kHz.
• Asynchronous sampling rate
converters on each input.
• Card synchronisation to any input,
AES-11 input or other PC-AD2 card.
• 32 and 64 bit drivers for
Windows7/8/10, Windows Server
2008/R2, 2012/R2.
• WDM-compliant supporting Wave,
DirectSound, DirectShow and Core
Audio APIs.
• Simultaneous record/play per
channel.
192
The PC-AD2 is a dual stereo analogue input/output and dual stereo AES-3 digital input/
output sound card in the PCIe half height format. One of the analogue inputs can be
used as a mic input, there is a dedicated AES-11 sync input and also 2 GPIOs. It is fully
compatible with the Windows™ Wave, DirectSound, DirectShow, MCI and Core Audio APIs,
supporting audio up to 24 bit, 192kHz.
The PC-AD2 is a professional-quality
half-height PCIe audio input-output card,
offering both analogue and AES-3 stereo
inputs and outputs. It is supplied with a
Windows WDM driver to provide full sound
card functionality on Windows 7, 8, 8.1, 10
and Windows Server 2008 R2, 2012 and
2012 R2. Many of the modern smaller PC
systems only have space for half height PCIe
cards – the PC-AD2 card has been designed
with this in mind allowing a high channel
count in a small format card.
The PC-AD2 has two stereo analogue inputs
(the first of which can be used as a dual
mono or stereo microphone input) and two
stereo analogue outputs, two stereo AES-3
inputs and two stereo outputs, a dedicated
AES-11 sync input and two general purpose
(GPIO) inputs and outputs.
When acting as a microphone input,
phantom power can be applied at +48V to
the microphone input connection, but is
removed if Line Input 1 becomes active. An
audio limiter on the mic input automatically
reduces the microphone gain if the
recording level approaches clipping.
The peak analogue level sets the clipping
threshold for the analogue inputs to one
of +15, +18, +20, +22 or +24dBu. With the
Windows endpoint faders set to maximum,
these levels correspond to full scale on the
software audio streams.
The card’s core sampling rate can be
synchronised to an external source
chosen from either of the AES-3 inputs,
the dedicated AES-11 sync input or by
using the inter-card cable (PC-AD2SY) for
synchronising to another PC-AD2.
Professional Sound Cards & Radio Capture Cards
The PC-AD2 Properties/Configuration Page.
Independent asynchronous sampling
rate conversion on the inputs supports
rates from 32kHz to 192kHz, while the
output rate can be configured as 192kHz,
96kHz, 88.2kHz, 48kHz, 44.1kHz or 32kHz,
either free-running or locked to an AES11
reference on any of the digital inputs.
Software sampling rate conversion is
automatically inserted by Windows when
the application rate does not match the
hardware rate, ensuring that files of any
sample rate can be played and recorded.
Extended software bit depths of 32 and 24
bits are supported as well as 16 and 8 bits
for playback and recording. For each AES-3
input, the received sampling rate, frame
lock and sampling rate converter status is
shown.
The PC-AD2, also supplied with full height brackets.
The playback and recording topology is
virtualised because Windows treats each
input source as its own ‘endpoint’. Endpoints
are the physical audio sources and
destinations, such as microphones, speakers
and line connectors. Each of the PC-AD2
card’s physical line inputs is represented by
an endpoint device, and as the hardware
has separate A/D converters or AES-3
receivers for each one, they can be used
simultaneously. A Microphone endpoint is
also created which is shared with analogue
Line Input 1 endpoint. The Windows mixer
API is virtualised for each application,
providing just a mute and volume control for
each endpoint and affecting only the audio
going to and from that application.
The PC-AD2 has two optically-isolated
general purpose inputs and outputs which
are accessed through a programming API
supplied with the driver package. A demo
program is included with the driver package
to illustrate the operation of the GPIO API
and to quickly test the GPIO functionality.
All the standard Windows audio APIs are
available including the Core Audio API,
Wave, DirectSound, DirectShow, MCI and
MIDI playback, as are a variety of audio
compression modes via the Windows Audio
Compression Manager or other software
compression systems.
The PC-AD2 Line Output 1 Properties Page.
Multiple cards may be installed in a
single PC.
159
Professional Sound Cards & Radio Capture Cards
A 44-pin high-density D-type connector to
XLRs & 9 pin D-type breakout lead is offered
as an option, the PC-AD2BC.
Specification for PC-AD2 2 Stereo
Analogue I/O & 2 Stereo AES-3 I/O
PCIe Half Height Sound Card
Operating Systems Supported
Platform:
Windows 7, 8, 8.1 and 10
(32-bit and 64-bit versions
Server 2008 R2, 2012 and
2012 R2.
Digital Audio Interface:
2 x AES-3 (formerly known as
AES/EBU) inputs/outputs
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Synchronisation Input:
1 x AES-11 input
synchronisation:
Single lane PCIe 1.1
Output Sampling Rates:
32kHz, 44.1kHz, 48kHz,
88.2kHz, 96kHz and 192kHz
32kHz, 44.1kHz, 48kHz,
88.2kHz, 96kHz or 192kHz
Input Sampling Rates:
32kHz to 192kHz
Software Sampling Rates:
Any rate from 10Hz to 384kHz
Analogue Resolution:
24 bits
Software Resolution:
32, 24, 16 or 8 bits
Dynamic Range:
>100dB typical (unweighted)
GPIO Inputs:
Analogue THD+N:
< 0.0015% (-96dB)
Optically isolated, 2.5 to 50V
on (2.5 to 50mA).
Analogue Input Interface:
2 x stereo balanced analogue
line (1 can be switched to
dual mono or stereo
microphone)
GPIO Outputs:
Optically isolated, 5mA max
on, 80V max off.
Connector:
44-pin high-density D-type
female
PCIe Specification
Card Interface:
Audio Specification
Analogue Sampling Rate:
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(microphone)
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Analogue Maximum Signal:
+15dBu, +18dBu, +20dBu,
+22dBu, +24 dBu (software
settable)
Microphone Sensitivity:
-58dBu to +2dBu for full scale
input
Microphone E.I.N.:
-119dBu (unweighted)
(Sample rate 48kHz, max
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Microphone CMRR:
77dB @ 1kHz, 77dB @
10kHz
Microphone Phantom Power: +48V (software enabled)
Frequency Response:
160
Input – 1Hz to 88kHz
Output – 1Hz to 88kHz
(192kHz hardware sampling
rate)
Equipment Type
PC-AD2
2 Stereo analogue I/O & 2
stereo AES-3 I/O PCIe half
height sound card
Physical Specification
Dimensions (Raw):
18cm (L) x 7.0cm (H) x 2cm (D)
7.1” (L) x 2.8” (H) x 0.8” (D)
Dimensions (Boxed):
27cm (L) x 22.5cm (H) x 6cm (D)
10.6” (L) x 8.9” (H) x 2.4” (D)
Weight:
Nett: 0.10kg Gross: 0.20kg
Nett: 0.2lbs Gross: 0.4lbs
Accessories
PC-AD2BC
PC-AD2 full XLR & 9 pin D-type
breakout cable
PC-AD2SY
Cable to sync 2 x PC-AD2 cards
Professional Sound Cards & Radio Capture Cards
PC-DIG4 Digitorc 4, 4 Stereo AES-3 I/O PCIe Sound Card
Category: Professional Sound Cards &
Radio Capture Cards.
Product Function: Provides four AES-3
audio inputs and outputs for use in a PC.
Typical Applications:
Audio workstations, automation
systems, audio logging, multi-channel
playout.
The Digitorc 4 has four AES-3
stereo input and output channels
on a Windows platform and is
fully compatible with the Wave,
DirectSound, DirectShow, MCI and
Core Audio APIs.
Features:
• 4 independent transformer-coupled
AES-3 inputs and outputs.
• 24-bit audio resolution.
• Sampling rates up to 96kHz.
• Asynchronous sampling rate converters
on each input.
• Card synchronisation to any input or
NTP-locked system clock.
• 32 and 64 bit drivers for Windows XP, &
Windows7/8/10.
• WDM-compliant supporting Wave,
DirectSound, DirectShow and Core
Audio APIs.
Software sampling rate conversion is
automatically inserted by Windows when
the application rate does not match the
hardware rate, ensuring that files of any
sample rate can be played. Extended
software bit depths of 32 and 24 bits are
supported as well as 16 and 8 bits for
On the card is implemented both a single lane
playback and recording.
bus-master PCIe interface and four x 24-bit
AES-3 codecs. Independent asynchronous
The playback topology consists of a master
sampling rate conversion on the inputs
output level, mute control and peak meter,
supports rates from 32kHz to 96kHz, while
a wave level and mute control, and input
the output rate can be configured as 96kHz,
monitor level and mute controls for each of
88.2kHz, 48kHz, 44.1kHz or 32kHz, either
the line inputs. The record topology consists
free-running or locked to an AES3 or AES11
of a master input level, mute control and
reference on any of the inputs. When used
peak meter, line input level controls for
with an internet time standard (e.g.ntp.org), a each of the physical inputs and a digital
very precise sampling rate can be achieved.
loopback level control and mute. The range
Specification for PC-DIG4 Digitorc 4
AES3 Sound Card
Operating Systems Supported
Platform:
Windows XP, Server 2003, Vista,
Server 2008, Windows 7, Server
2008 R2, Windows 8 and Server
2012 (32- and 64-bit versions),
Windows 10
Features
The Digitorc 4, 4 Stereo AES-3 I/O PCIe Sound Card
on the input and output master controls
is -96dB to +6dB, while the individual line
controls range from -96dB to 0dB. The mixer
functions allow inputs to be mixed back
into each output, while a digital loopback is
available from each playback channel into its
corresponding record channel.
High quality electrostatically-shielded
transformers are used on all the inputs and
outputs to give superb performance.
Multiple cards may be installed in a
single PC.
Card Interface:
Single lane PCI Express version
1.1
Line Interface:
Transformer coupled AES3
(AES/EBU)
Line Output
Sampling Rate:
96kHz, 88.2kHz, 48kHz, 44.1kHz
or 32kHz (configurable)
Line Input
Sampling Rate:
32kHz to 96kHz via independent
asynchronous sampling rate
converters
Audio Resolution:
24 bits
Sampling Rate
Accuracy:
+/- 5ppm
External:
AES11 compliant synchronisation:
Frequency Response: DC to 43.5kHz (at 96 kHz
sampling)
Connector:
25-pin D-type female
Equipment Type
PC-DIG4
Digitorc 4 4 stereo AES-3 I/O
PCIe sound card
Physical Specification
Dims (Raw):
14cm (L) x 12.5cm (H) x 2cm (D)
5.5” (L) x 4.9” (H) x 0.8” (D)
Dims (Boxed):
27cm (L) x 22.5cm (H) x 6cm (D)
10.6” (L) x 8.9” (H) x 2.4” (D)
Weight:
Nett: 0.10kg
Nett: 0.2lbs
Gross: 0.20kg
Gross: 0.4lbs
Accessories
PC-DIG4BC
Digitorc 4 XLR breakout cable
A 25-pin D-type connector to 8 x XLR
breakout lead is offered as an option.
161
Professional Sound Cards & Radio Capture Cards
PC-AUR44 Auricon 4.4 PCIe Analogue Sound Card
Category: Professional Sound Cards &
Radio Capture Cards.
Product Function: Provides four stereo
balanced analogue audio inputs and
outputs for use in a PC.
Typical Applications:
Audio workstations, automation
The Auricon 4.4 is a professional
quality 4 stereo input and 4 stereo
output analogue PCIe audio card. The
inputs and outputs can be reconfigured
as separate mono channels, giving
eight inputs and outputs.
It is supplied with a Windows WDM driver to
provide full sound card functionality under
Windows XP, Server 2003, Vista, Server 2008,
Windows 7, Server 2008R2, Windows 8,
Server 2012 & Windows 10.
The card uses 24-bit 192kHz sigma-delta
converters which pass data to and from the
PC via a single lane PCI Express interface. An
onboard FPGA provides audio buffering, level
adjustment and mixing functions. Hardware
sampling rates of 48kHz, 96kHz and 192kHz
162
systems, audio logging, multi-channel
playout.
Features:
• 4 independent high level balanced
stereo inputs and outputs.
• 24-bit audio resolution.
• Sampling rates up to 192kHz.
• 32 and 64 bit drivers for Windows XP, &
Windows 7/8/10.
• WDM-compliant supporting Wave,
DirectSound, DirectShow and Core
Audio APIs.
• Simultaneous record/play per channel.
are available, with the Windows sampling
rate converter transparently providing
support for other rates. The card supports
extended bit depths to 32 bit and the
software sampling rate and bit depth (32, 24,
16 or 8 bits PCM) can be set independently
for each input and output channel.
Windows Wave, DirectSound and DirectShow
API’s are supported, as are a variety of audio
compression modes via the Windows Audio
Compression Manager or other software
compression systems. On Windows Vista/
Server 2008 and later systems, the Core
Audio API is also fully supported.
There are four configuration settings for
the Auricon 4.4, these being Mode (stereo/
mono), H/W Sampling Rate, Input Coupling
master controls is -96dB to +6dB, while the
individual line controls range from -96dB
to 0dB.
Multiple cards may be installed in a
single PC.
A 44-pin high-density D-type connector to
XLRs breakout lead is offered as an option.
Specification for PC-AUR44 Auricon
4.4 PCIe Analogue Sound Card
The Auricon 4.4 PCIe Analogue Sound Card
and Nominal Line Level. The mode may
be configured as either stereo or mono. In
mono mode the number of input and output
channels that Windows sees is doubled.
The nominal line level can be set to +8dBu,
+4dBu or 0dBu. In each case the clipping level
is 16dB above the nominal level. The input
coupling can be set to either DC or AC (the
default is AC).
The playback topology consists of a master
output level, mute control and peak meter,
and input monitor level and mute controls for
each of the line inputs. The record topology
consists of a master input level, mute control
and peak meter, and level controls and mutes
for the physical input and digital loopback.
The digital loopback allows the output of
the card to be digitally mixed back into the
input. The range on the input and output
Operating Systems Supported
Platform:
Supports Windows - XP, Server
2003, Vista, Server 2008, Windows
7, Server 2008 R2, Windows 8,
Server 2012 (32-bit and x64
versions), Windows 10
Audio Specification
Dynamic Range:
114dB typical (unweighted)
Input Impedance:
20k (balanced)
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Maximum Signal:
+24dBu (34.6Vp-p)
Frequency Response: Input – DC to 88kHz (DC coupling)
1Hz to 88kHz (AC coupling)
Output – DC to 88kHz (192kHz
hardware sampling rate)
Connector:
44-pin high-density D-type female
Equipment Type
PC-AUR44
Auricon 4.4 PCIe analogue sound
card
Physical Specification
Dims (Raw):
14cm (L) x 12.5cm (H) x 2cm (D)
5.5” (L) x 4.9” (H) x 0.8” (D)
Dims (Boxed):
27cm (L) x 22.5cm (H) x 6cm (D)
10.6” (L) x 8.9” (H) x 2.4” (D)
Weight:
Nett: 0.10kg
Nett: 0.2lbs
Gross: 0.20kg
Gross: 0.4lbs
Accessories
PC-AUR44BC
Auricon 4.4 XLR breakout cable
Professional Sound Cards & Radio Capture Cards
Radio Capture Cards
Radio capture cards don’t have inputs and outputs, other than an aerial to capture the
required signal. They allow a PC to record or monitor a number of DAB/DAB+, FM or AM
radio channels so are useful for logging and monitoring applications.
PC-DAB1-4 Multi-Ensemble DAB+/DAB Radcap PCle Card
Category: Professional Sound Cards &
Radio Capture Cards.
Product Function: Simultaneous audio
capture of multiple DAB services for use
in a PC.
Typical Applications:
Audio logging, station monitoring, media
tracking.
Features:
• Simultaneous capture of every audio
service across multiple ensembles.
The PCIe DAB+/DAB radio capture
card receives and decodes the entire
contents of up to four DAB+/DAB
ensembles, rendering each audio
service as a virtual Windows audio
capture device for use with multichannel recording or monitoring
software.
Broadcast data services, including
DLS text and MOT slideshows, are
also available through a simple
application programming interface.
• Available in 1, 2, 3 or 4 ensemble
versions with field expansion option.
• Tunes Band III (174-240 MHz) using
standard European channel numbers.
• DAB+ and legacy DAB supported.
• Each service appears as a standard
audio input device.
• 32 and 64 bit drivers for Windows Vista,
Windows 7,8,10.
• WDM-compliant supporting Wave,
DirectSound, DirectShow and Core
Audio APIs.
• API for monitoring, control and PAD
extraction.
• Sample application for displaying DLS
text and MOT slideshow.
The card supports both legacy DAB MP2
audio coding as well as the new HE-AAC v2
encoding used with DAB+ broadcasts.
Any application that records from standard
wave input devices can be used to record
the audio streams from the DAB+ Radcap.
A recording level and mute control are
provided for each service through the
devices’ mixer ports.
A sample monitor application is included
which displays a control panel for each card
and creates buttons for each audio service.
When a button is clicked, it plays the audio
through the default output device while
Specification for PC-DAB1-4 MultiEnsemble DAB+/DAB Radcap
System Requirements
Platform:
The PC-DAB1 Multi-Ensemble DAB+/DAB
Radcap PCle.
displaying information obtained from the
service and any DLS text and MOT images
being broadcast.
The number of ensembles is factory-set as 1
(PC-DAB1), 2, 3 or 4 (PC-DAB4) but is fieldexpandable through a purchased expansion
key. Multiple cards can be installed, allowing
simultaneous monitoring or recording of
more than four ensembles.
A sample application is provided with
the card, allowing monitoring of DAB+/
DAB audio and data as well as providing
diagnostic ensemble spectrum displays,
signal quality indicators and an uncorrected
error counter. Each card panel displays the
ensemble name and identifier, along with
the phase reference correlator level and
signal spectrum.
Windows Vista, Server 2008,
Windows 7, Server 2008 R2,
Windows 8, Server 2012 (32-bit and
64-bit versions supported),
Windows 10 (Note: Windows XP
and Server 2003 are not supported)
Processor:
2GHz quad-core or better
Memory:
1GB minimum
Motherboard:
PCIe socket, single lane or greater
Other:
Sound card or motherboard sound
port for monitoring
Specifications
Tuning Range:
Band III (174-240 MHz)
DAB Format:
Mode 1
RF Input:
BNC connector
PCIe Interface:
Single lane PCIe 1.1
Number of Ensembles: Factory-configured for 1, 2, 3 or 4
ensembles (field-expandable for an
additional fee)
Total Number
of Services:
Error Correction:
128
Soft-decision Viterbi inner decoder,
Reed-Solomon outer decoder
Audio Decoding:
MP2 and HE-AAC v2
Audio Format:
48kHz 16-bit stereo (other
application sampling rates and bit
depths supported through the
Windows SRC) (24kHz and 32kHz
services are internally up-converted
to 48kHz)
Decoding Latency:
3 seconds
Equipment Type
PC-DAB1-4
Multi-ensemble DAB+/DAB radcap
PCIe card
Physical Specification
Dims (Raw):
14cm (L) x 12.5cm (H) x 2cm (D)
5.5” (L) x 4.9” (H) x 0.8” (D)
Dims (Boxed):
27cm (L) x 22.5cm (H) x 6cm (D)
10.6” (L) x 8.9” (H) x 2.4” (D)
Weight:
Nett: 0.10kg
Nett: 0.2lbs
Gross: 0.20kg
Gross: 0.4lbs
163
Professional Sound Cards & Radio Capture Cards
PC-FM6-32 FM Radcap PCle Card (6 to 32 Channels)
•
Category: Professional Sound Cards &
Radio Capture Cards.
Product Function: Simultaneous audio
capture of multiple FM radio stations for
use in a PC.
Typical Applications:
Audio logging, station monitoring, media
tracking.
Features:
• Simultaneous capture of multiple FM
The FM Radcap PCIe is a radio
capture card designed for
simultaneous recording of multiple
radio stations. The frequency of each
station is set in software and its audio
appears as a standard Windows audio
input device. RDS decoding is also
supported.
The card uses a high-speed A/D converter
to digitise the entire FM band, with up to
32 individual tuners. The Radcap achieves
exceptionally low audio distortion through
the use of linear phase filtering and
mathematically precise FM demodulation
and stereo decoding. FM demodulation
and stereo decoding is done in FPGA
164
•
•
•
•
•
stations.
Available in 6, 12, 18, 24 and 32 station
versions with field expansion option.
Tunes 87.5-108.5 MHz in 25kHz steps.
Stereo and RDS decoding.
Each station appears as a standard
audio input device.
32 and 64 bit drivers for Windows XP
and all later versions, as well as Debian
Linux.
WDM-compliant supporting Wave,
DirectSound, DirectShow and Core
Audio APIs (Windows) and ALSA (Linux).
API for monitoring and control.
Specification for PC-FM6-32
FM Radcap PCle (6 to 32 channels)
System Requirements
Platform:
The PC-FM6-32 FM Radcap PCle
Card (6 to 32 Channels).
The card can be configured to operate in stereo,
•
mono or paired mono (two mono stations
combined on a 2-channel audio stream) modes.
Multiple cards can be used in a single PC,
fabric, while RDS decoding, if enabled,
subject to available CPU bandwidth. The audio
is performed in the driver using the host
de-emphasis may also be set to either 50us or
CPU’s SSE-2 instruction set. This division of
labour between the FPGA and driver allows 75us. In Australia, New Zealand and Europe 50us
is used, while in the USA and Canada
greatest flexibility in catering for future
baseband technologies while minimising the 75us is used.
CPU overhead of the card.
A utility called FMSpectrum is supplied. This
displays the RF spectrum from 85MHz to
A WDM driver for Windows XP (SP2 or
111MHz, using data from the card’s front-end
later), Server 2003, Vista, Server 2008,
256-point FFT and may be useful in selecting
Windows 7, Server 2008 R2, Windows 8,
the best location for the antenna or resolving
Server 2012 & Windows 10 is supplied
interference problems.
as well as software for setting the tuner
frequencies and monitoring the received
The card is factory-configured for 6 (PC-FM6),
audio. A programming API and DLL for
12, 18, 24 or 32 (PC-FM32) stations, but may be
software control and monitoring are also
expanded in the field for an additional charge.
supplied.
Windows XP (SP2 or later), Server
2003, Vista, Server 2008, Windows
7, Server 2008 R2, Windows 8 and
Server 2012 (32-bit and 64-bit
versions), Windows 10
Processor:
2.5GHz Pentium 4 or better
Memory:
256 MB minimum (1GB for Vista
Windows 7/Server 2008, 2GB for
Windows 8/Server 2012)
Bus:
Single lane PCI Express v1.1
Other:
Sound card or motherboard sound
port for monitoring
Specifications
Tuning Range:
87.5MHz to 108.5MHz in 25kHz
steps
Sensitivity:
10uV for 40dB S/N
Maximum Input:
150mV RMS
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De-emphasis:
Configurable as 50us or 75us
Audio Distortion:
<0.01%
Audio Sampling Rate: 48kHz (all other rates automatically
supported via Windows sampling
rate converter)
Number of Stations:
6, 12, 18, 24 or 32 (factory
configured but end-user
expandable)
RDS Decoding:
Optionally enabled in driver
configuration
Equipment Type
PC-FM6-32
FM Radcap PCle card (6 to 32
channels)
Physical Specification
Dims (Raw):
14cm (L) x 12.5cm (H) x 2cm (D)
5.5” (L) x 4.9” (H) x 0.8” (D)
Dims (Boxed):
27cm (L) x 22.5cm (H) x 6cm (D)
10.6” (L) x 8.9” (H) x 2.4” (D)
Weight:
Nett: 0.10kg
Nett: 0.2lbs
Gross: 0.20kg
Gross: 0.4lbs
Professional Sound Cards & Radio Capture Cards
PC-AM6-32 AM Radcap PCle Card (6 to 32 Channels)
Category: Professional Sound Cards &
Radio Capture Cards.
Product Function: Simultaneous audio
capture of multiple AM radio stations for
use in a PC.
Typical Applications:
Audio logging, station monitoring, media
tracking.
Features:
The AM Radcap PCIe card is a
radio capture card designed for
simultaneous recording of up to
32 radio stations. The frequency of
each individual station may be set in
software and its audio appears as a
standard Windows audio input device.
The AM Radcap uses a high speed analogueto-digital converter to digitise the entire
AM band, with advanced digital signal
processing on a Spartan 6 FPGA used to tune
and extract the audio for each individual
station. It can be configured to either create
a separate audio stream for each station
or to pair stations together as 2-channel
streams.
• Simultaneous capture of multiple AM
stations.
• Available in 6, 12, 18, 24 and 32 station
versions with field expansion option.
• Tunes 500-1610 kHz in 1kHz steps.
• Each station appears as a standard
audio input device.
• 32 and 64 bit drivers for Windows XP
and all later versions, as well as Debian
Linux.
• WDM-compliant supporting Wave,
DirectSound, DirectShow and Core
Audio APIs (Windows) and ALSA (Linux).
• API for monitoring and control.
A WDM driver for Windows XP, Windows
7/8/10 and other versions is supplied as well
software for setting the tuner frequencies
and monitoring the received audio.
A recording level control, mute control and
peak meter are provided for each station (or
pair of stations) through the devices’ mixer
ports. The default level setting is 50%, and at
this setting 100% modulation will produce a
peak audio level 6dB below clipping.
A utility called AmSpectrum is supplied. This
displays the RF spectrum from 500kHz to
1700kHz and may be useful in selecting the
best location for the antenna or resolving
interference problems. The receiver
bandwidth can be set to wide (default) or
Specification for PC-AM6-32
AM Radcap PCle (6 to 32 channels)
System Requirements
Platform:
The PC-AM6-32 AM Radcap PCle Card
(6 to 32 Channels).
Windows XP, Server 2003, Vista,
Server 2008, Windows 7, Server
2008 R2, Windows 8, Server 2012
(32-bit and 64-bit versions),
Windows 10
Processor:
1GHz Pentium II or better
Memory:
128MB minimum (1GB for Vista and
later systems)
Bus:
Single lane PCI Express v1.1
Other:
Sound card or motherboard sound
port for monitoring
Specifications
Tuning Range:
500kHz to 1710kHz in 1kHz steps
Sensitivity:
50uV for 40dB S/N
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Filter Attenuation:
82dB at 15kHz or more from centre
frequency
narrow. The narrow setting restricts the
audio response to about 3kHz, which may
be useful in noisy environments.
Audio Bandwidth:
5kHz
Audio Distortion:
<0.1%
A utility program called Tuner is also
supplied which can be used to set the
frequency of each station and to monitor
each station through the PC’s standard
sound card or motherboard sound port. The
Tuner program also provides relative signal
strength indicator bars which may be useful
in adjusting antenna placement.
Number of Stations:
Audio Sampling Rate: 22.05kHz (other rates supported via
Windows SRC)
6, 12, 18, 24 or 32
Equipment Type
PC-AM6-32
AM Radcap PCle card (6 to 32
channels)
Physical Specification
Dims (Raw):
14cm (L) x 12.5cm (H) x 2cm (D)
5.5” (L) x 4.9” (H) x 0.8” (D)
Dims (Boxed):
27cm (L) x 22.5cm (H) x 6cm (D)
10.6” (L) x 8.9” (H) x 2.4” (D)
Weight:
Nett: 0.10kg
Nett: 0.2lbs
Gross: 0.20kg
Gross: 0.4lbs
The card is factory-configured for 6 (PCAM6), 12, 18, 24 or 32 (PC-AM32) stations,
but may be expanded in the field for an
additional charge.
165
166
Audio Logging
Net-Log Network Audio Logger
The Net-Log has been specifically designed for long-term recording and archival
by the use of a dedicated hardware recording platform with a simple network
connection for audio playback. Playback software on a network connected PC
streams the audio to the desktop where it can be played out or saved as a file.
168
Net-Log-Win Software
170
Mentor Time-Server
173
PC-FLS8 Flashlog 8 Audio & Radio Capture Logging Software
174
Flashlog 8 is configurable, ultra powerful multi-channel logging software. It makes use of
separate Sonifex hardware (sound and radio capture cards) to provide a host of logging
features on a single PC platform.
PC-FLS8LT1 Flashlog 8 Single Stereo Channel Logging Software
176
PC-FLS8LT2 Flashlog 8 Two Stereo Channel Logging Software
176
PC-RACK Industrial PC Racks for Audio Logging
177
167
Audio Logging
Net-Log Network Audio Logger
•
•
Category: Audio Logging.
Product Function: To record 4 mono
channels for long term archival and play
back over a network.
Typical Applications:
Recording radio station off air
transmissions for regulatory purposes,
recording board room proceedings,
recording courtroom trials.
Features:
• Dedicated reliable multi-channel
recording hardware.
• Easy to use Windows™ remote
operation for playback, archiving,
The Net-Log has been specifically designed
for long-term recording and archival by
the use of a dedicated hardware recording
platform that uses a simple network
connection for audio playback.
168
•
•
streaming and configuration.
Independently configurable channel
pairs to record at different sample rates,
different bit rates and in mono or stereo.
Recording duration configuration of
the channels to suit your needs, e.g. 2
channels of low quality logging for 12
months together with two channels of
high quality logging for 1 month, on one
machine.
Record inputs can be controlled remotely,
e.g. snoop recording from mic-live output
of a mixer.
Automated voice recording - each
channel can start and stop recording
automatically by setting audio threshold
levels, which means that it can be
used for voice controlled systems and
telephony recording.
• Up to 250,000 recordings per channel
can be made.
• Each Net-Log has its own IP address, so
a number of machines can be installed
on the same network.
• Fail-safe alarm indicators and remotes.
• Password protection and security
access rights to system.
• Support for hard disk sizes up to
2048GB.
• Recordings can be made using optional
G.729 algorithm, allowing longer
recordings and lower network traffic.
G.729 audio quality is comparable to
MPEG L2 at 32kbits/sec when used to
record/playback speech.
• Recording on Net-Log can be controlled
serially by Crestron and Televic board
room controllers.
Playback software on a network connected
PC streams the audio to the desktop where
it can be played out or saved as a file.
enough for continuous recording 24/7/365.
The Net-Log was designed from the ground
up to offer :
The Sonifex Net-Log is a 4 channel
audio logger which can record weeks of
programming on a large internal harddisk. The unit was designed as dedicated
hardware for reliability reasons, i.e. there’s
no PC motherboard in this machine.
Although PC based systems are great for
playback, they generally aren’t robust
• High reliability for continuous operation.
• High quality audio (mpeg compressed).
• Compatibility with existing broadcast &
Windows based systems (bwf files can be
saved).
• Automatic operation with very simple
to use software.
All audio created by Net-Log is Windows
Media PlayerTM compatible so files can be
emailed to colleagues and customers and
played out on any PC with a sound system.
There are many applications for NetLog, from radio stations recording their
broadcasts for regulatory purposes, to
small call centres, law firms and security
companies using them for monitoring.
Net-Logs have been installed in court
rooms & police interview rooms for audio
surveillance and offer a cost saving desk-top
based logging & filing system for audio.
Radio Broadcast Logging
Where logging is required by the authorities,
Net-Log provides a perfect, simple to use,
elegant solution providing playback of
requested recordings within seconds. But
even if logging isn’t regulatory, Net-Log can
still deliver audio to the desktop of anyone
at a radio station – you don’t need to go to
the racks-room, or central technical area to
play out a recording previously made. So it
has the following advantages :
• From their desktop, the sales team can
check to make sure that advertising was
played on time and can create audio files
to email to customers.
• Program controllers can snoop on their
presenters, checking performance and
Audio Logging
control and for alarm outputs (hard disk
failure and archiving warning).
checking that scheduled music was
played.
• Station managers can check up on their
competition and compare it with the
sound of their own station.
• The production team have much easier
access to recorded audio from past radio
shows for the preparation of future stings
& trailers.
Net-Log records in mpeg layer 2, the
standard for radio broadcasting, so that you
can be sure that audio files created using
Net-Log are compatible with almost every
mpeg based automation system available.
File downloads can be made in mp2, mp3
or linear .wav format, and broadcast wave
format (bwf) files are supported as standard,
ensuring tight integration with your playout
systems.
Key Features of Net-Log
Hardware Description
Net-Log is a 1U rack-mount unit which has
4 mono record inputs on 3 pin XLR sockets
with potentiometer gain controls and LED
level indicators for each input. These accept
2 stereo, or 4 mono audio streams which are
encoded in mpeg layer 2 format and written
to a large internal IDE hard disk drive.
The size of the drive is continuously being
increased as technology improves and NetLog can support sizes up to 2048GB. Contact
[email protected] for current drive sizes
installed.
A 10Mbps RJ45 network port is used to
connect Net-Log to your PC network and
audio is streamed using TCP/IP. Audio can be
streamed simultaneously to one, or many,
PCs, as well as simultaneously archiving
audio data to an IP server automatically
in the background. All the Net-Log
configurations and operational features are
controlled from a connected PC using the
Net-Log-Win software.
Net-Log has 2 serial ports on 9 pin D-types
which are used to program the IP address of
the machine and to upload new software.
Additionally, there is a 15 pin D-type
connector used for remote input recording
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Audio Logging
Net-Log-Win Software
There are 3 PC applications used to
configure and control the Net-Log :
• Net-Util - A utility mainly used to
initially configure the Net-Log and solve
connection problems.
• Net-Log-Win - This is the application used
on a day to day basis to control recording
and playback.
• Auto-Archive - This is used to
automatically archive/backup files to an
• IP server.
Net-Util Application
The Net-Util program connects to a
Net-Log unit via a serial (COM) port using
the supplied serial cable, and is used to :
• Define the IP address of the connected
Net-Log.
• Upload network settings.
• Monitor status messages.
• Upload new firmware into the Net-Log.
Net-Log-Win Application
There are 4 main screens in Net-Log-Win
which control and configure the connected
Net-Log(s) : Recording, Playback, Archiving
and Options. Access to these screens can
be password protected and users can be
configured either to View or Edit each
screen. In the top part of the screen, a drop
down list allows you to select which Net-Log
you wish to connect to and the alarm states
are also shown.
The Options screen is the place where you
configure the settings of the connected
Net-Log. The Options Archive screen in NetLog-Win is used to configure the IP address
and directory of the archiving destination
and also to define archives to be written
either as the cuts were recorded, or in files
of a certain duration, e.g. hourly.
Recording
Unlike normal hard-disk recording systems
which stop when the disk is full, on reaching
the “end” of the recording data space
Net-Log begins to overwrite the oldest
recordings, meaning that you always have
access to the last recordings made, up to
the size of the hard-disk. Additionally, it
allows simultaneous recording and playback,
so that you don’t have to stop the unit
recording in order to search and play. Each
channel can be given a service name for
reference when searching for audio.
Recording can be controlled in 5 different
ways :
• Using on-screen buttons in the Manual
Record screen.
• Using the Program Record screen to start
& stop recording at different times, using
up to 20 programs. Programs can be
saved to disk.
Net-Util Screen.
Net-Log-Win Software & Licensing
The Net-Log is initially configured and
controlled by Net-Log-Win software
which is a suite of three applications
which run under MS Windows™ 7/8/10.
The software is sold either as a two
stream, or a five stream licence. This
means that each Net-Log on a network
can simultaneously support either 2, or 5
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simultaneous connections to it for playback
or archiving. Note that you don’t need to
buy a licence for each Net-Log if they are on
the same network - if you have a number
of Net-Logs on a single network, you only
need to purchase a single set of licences,
e.g. if you have 3 Net-Logs and buy a single
Net-Log-Win05 5 stream licence, up to 5
different people can stream from each of
the 3 Net-Logs, i.e. 15 separate streams,
provided that your network infrastructure
can allow for this. Additionally, this is a buy
once licence - we don’t ask you to re-buy it
each year !
Net-Log licensing operates by you providing
us with a licence code which appears in a
licence window within Net-Log-Win. Once
we’ve verified the authenticity of your
order, we send you an authentication
code which you enter into the licence
window to licence your Net-Log. If you
don’t have a licence code, the Net-Log
will continue recording but access to the
recordings will be denied after 3 days.
Audio Logging
Manual Record Screen.
• Using a remote input on the 15 way
D-type connector.
• Automatically on voice/sound level, ideal
for recording telephone conversations,
configured in the Options Record screen.
• By Crestron and Televic board room
controllers, via an RS232 serial
connection. The controllers are mainly
used in meeting rooms and allowing NetLog to more easily be used for recording
in this environment.
In the event of needing to record directly
from the Net-Log, there is an emergency
Record Control on the front panel of
the machine.
Program Record Screen.
Net-Log-Win Playback
Net-Log-Win software for MS-Windows™
7/8/10 is used to listen to audio recorded
at a previous time and date, or for copying
audio to a file, e.g. to email to a client or
colleague.
duration and a file is downloaded from
the Net-Log and created on your local PC.
You can then use separate applications to
email the audio file, edit it, or save it to
CD, or DVD. Files can be saved as .mp2,
.mp3 or linear .wav.
• Using the Play Program screen, you can
select a number of recordings, made at
the same time over a number of days of
the week and save them off to a single
directory, e.g. for storing commercials
from a particular customer. Files can be
saved as .mp2, .mp3 or linear .wav.
• Using the Play File screen, you can
open any previously saved audio file for
playback. You can also play .mp3 and
linear .wav files this way.
Play Stream Screen.
Additionally, a separate free player is
available within Net-Log-Win to play any
audio file created from a Net-Log.
Play Download Screen.
In the Play screens, simply select a Net-Log
and the required channel service name
and a list of all the recordings made are
displayed, sorted by recorded date and
time. There are 4 ways to play audio :
Options Record Screen.
• Using the Play Stream screen, selected
audio is played out on your PC sound
system within seconds.
• Using the Play Download screen, you
can select a recording’s date, time and
Play Program Screen.
Play File Screen.
171
Audio Logging
Archiving
You can use the Archiving function to
automatically create long-term backups
of the recordings by streaming audio to a
separate server disk for archival purposes.
Net-Log-Win Free File Player.
For each recorded channel, you can define
an IP address and directory to save files
to and Net-Log can create files on that
directory either in the same way that they
are recorded, or in files of a certain length,
e.g. of 1 hour duration.
The Net-Log Auto-Archive utility is run on
the server that you are archiving to. It has
the following features :
• Simultaneous connection to multiple
Net-Logs.
• Automatic log-in, archive and log-out
process.
• Status display of Net-Logs on-screen in an
easy to understand format.
• Status reports, error reports and archive
alarm information can be emailed
automatically at regular intervals.
• Can be used to check the status of
• Net-Logs even when archiving is not used
or required.
Note that Archiving uses a licence stream.
172
Specification For Net-Log
of disk capacities, the hard-disk
size you buy may be larger
than this.
Audio Specification
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хϭϬŬɏďĂůĂŶĐĞĚ͘
Maximum Input Level: +20dB.
PC Soundcard Needed: Any windows compatible soundcard
that can play PCM (.wav) audio files
Distortion:
<0.1% @ 1kHz + 16dBu.
Operating System:
Signal to Noise Ratio:
90 dB RMS A wtd. 22kHz
bandwidth.
Equipment Type
Wow and Flutter:
Unmeasurable.
Net-Log-01
Net-Log 4 channel audio logger,
with large hard disk (contact sales
@sonifex.co.uk for current sizes
available)
Net-Log-Win01
Net-Log Windows™ software - 2
stream licence
Net-Log-Win05
Net-Log Windows™ software - 5
stream licence
Net-Log-UPS01
1U rack-mount UPS for use with
Net-Log
Net-Log-G729
G.729 software licence for one
Net-Log (up to 4 mono channels).
Phase Error at 10kHz: Unmeasurable.
Please note that the audio specification of the play output
depends on the soundcard that you have installed on your
PC. Net-Log is a recording device and has no audio output
of its own.
Rear Panel Connections, Controls and Indicators
Analogue Inputs:
4 x XLR 3 pin socket.
Input Level Controls:
4 x Rotary pre-set potentiometers.
Network:
RJ45 10Mbps ethernet.
Windows™ 7,8 or 10
Network Status LEDs: Rx, Tx, link and collision.
Physical Specification
RS232:
2 x 9 way D-type plug.
Dimensions (Raw):
Alarm Outputs and
Remote Inputs:
15 way D-type plug.
1U 19” rack x 220mm deep 48cm
(W) x 22cm (D) x 4.2cm (H) (1U)
1U 19” rack x 8.7” deep 19” (W)
x 8.7” (D) x 1.7” (H) (1U)
Power:
IEC Power plug, 95-265 VAC,
47-63Hz.
Dimensions (Boxed):
56cm (W) x 30cm (D) x 18cm (H)
22” (W) x 11.8” (D) x 7.09” (H)
Weight:
2.5kg - 5.5lbs
Controls and Indicators
Front Panel Controls:
Record control to instantly start
recording
Front Panel Power
Indicator:
Power LED (blue)
Front Panel Alarm
Indicators:
2 x Alarm LEDs, disk drive alarm
LED (red), archive alarm LED (red)
Front Panel Level
Indicators:
4 x Analogue input level LED, 1
for each channel (green, orange,
red)
Rear Panel Network
Indicators:
4 x LEDs, receive (RX), transmit
(TX), link and collision LEDs
Additional Information
Sample Rates:
16kHz, 22.05kHz, 24kHz, 32kHz,
44.1kHz and 48kHz
Audio Format:
Mpeg layer 2 with bit-rates as
below and G.729 low bit rate
Recording Types:
Mono, dual mono, stereo
Hard Disk Capacity:
A minimum hard-disk size of 1TB
is fitted.
Due to the rapidly changing nature
Audio Logging
Record Duration
The table below shows the approximate
record capacity at different bit-rates for one
mono channel on a 1TB hard disk. For 4
mono channels, divide the figures by 4.
Note: Each pair of channels can be given a
different bit-rate and * indicates bit-rates for
stereo channels only.
(kbit/sec)
Days
(kbit/sec)
Days
8
11572
112
826
16
5786
128
723
24
3857
*144
642
32
2893
160
578
40
2314
192
482
48
1928
*224
413
56
1653
*256
361
64
1446
*320
289
80
1157
*384
241
96
964
Specification For Net-Log UPS01
Batteries & Runtime
Output
Battery Type:
280 Watts/450 VA
Nominal Output Voltage: 230V
Output Voltage
Distortion:
Less than 5% at full load
Output Frequency
(sync to mains):
47-53Hz for 50 Hz nominal, 57
63Hz nominal
Crest Factor:
Up to 5 : 1
Waveform Type:
Stepped approximation to a
sinewave
Output Connections:
(4) IEC 320 C13
(2) IEC Jumpers
RBC™ Quantity:
1
Typical Backup Time at
half load:
19.4 minutes (140 Watts)
Typical Backup Time at
full load:
5.9 minutes (280 Watts)
Runtime Chart:
Smart-UPS SC
Dimensions (Raw):
1U 48.3cm rack x 43.2cm (W) x
38.5cm (D) x 4.4cm (H) (1U)
1U 19”rack x 17” (W) x 15” (D) x
1.7” (H) (1U)
Clock Sources
Mentor Server can be synchronised to NTP
(Time servers on the Internet).
Dimensions (Boxed):
47.6cm (W) x 58.9cm (D) x 16.4cm
(H)(1U)
18.74” (W) x 23.19” (D) x 6.45”
(H)
For pricing information, please contact the
Sales Department.
Typical recharge time:
5 hours
Replacement Battery:
RBC18
Physical Specification
Input
Nominal Input Voltage: 230V
Maintenance-free sealed Lead
Acid battery with suspended
electrolyte: leakproof
The Mentor Time Server is software
which runs on a MS-Windows Windows™
7/8/10 PC and synchronizes a number of
MS-Windows Windows™ 7/8/10 PC client
machines or Sonifex Net-Log units to its
time. The client PCs or Net-Logs read the
server time and have their time set using
TCP/IP at regular intervals.
Output Power Capacity: 280 Watts/450 VA
Max Configurable
Power:
Mentor Time-Server
Input Frequency:
50/60Hz +/-3Hz (auto sensing)
Input Connections:
IEC-320 C14
Input voltage range for
main operations:
160-286V
Weight:
Nett: 10.2kg
Nett: 22.4lbs
Input voltage adjustable 151-302V
range for mains operation:
Colour:
Grey
Net-Log UPS01.
Gross: 13.0kg
Gross: 28.6lbs
Sonifex Mentor Client Installation
The Mentor Client software can be installed
on any number of PCs connected to network
where a Mentor Server is located, but clock
updates will only be performed up to the
number of client licences installed on the
server.
Mentor Server
The Mentor Server to connect to is
identified by its TCP/IP address.
Equipment Type Mentor
MTS-5
Mentor Time-Server 5 client licence
MTS-25
Mentor Time-Server 25 client licence
MTS-100
Mentor Time-Server 100 client licence
MTS-500
Mentor Time-Server 500 client licence
173
Audio Logging
PC-FLS8 Flashlog 8 Audio & Radio Capture Logging Software
Category: Audio Logging.
Product Function: Continuous audio
logging software for analogue, AES-3, AM,
FM, DAB+, IP audio and internet streams
for long term archival and retrieval.
Typical Applications:
Compliance logging, air checking, content
reuse, monitoring of competitors.
Features:
• Continuous recording from 7 days to
up to 2 years, user selectable on each
channel or ensemble, dependent on
hardware & HDD size.
Flashlog 8 is configurable, ultra powerful
multi-channel logging software. It makes
use of separate Sonifex hardware (sound
and radio capture cards) to provide a host
of logging features on a single PC platform.
It can record from analogue and digital
line inputs, IP based line inputs (Axia™,
Wheatstone™ or Dante™), from the
Sonifex radio capture cards and internet
radio streams.
• Based on 64-bit Windows 7/8.1/10.
• Up to 64 analogue, AES-3 or IP-based (Axia
or Wheatstone) line inputs supported.
• Up to 32 AM stations & 32 FM stations
supported.
• Up to 4 DAB/DAB+ ensembles supported.
• Up to 32 internet radio streams supported.
• Contact-closure driven skimming option on
line inputs (separate hardware required).
• Supports archiving - optional parallel
recording of selected channels to an
archive location on the network.
• Simultaneous network playback on
multiple client PCs.
• Easy-to-use interface with rapid date/time
navigation and waveform display.
• Playback metadata display of FM RDS, DAB
DLS/Slideshow/EPG & internet stream
metadata.
But DAB+ isn’t all you get with Flashlog 8.
By using suitable hardware, Flashlog 8 has
the ability to record up to 64 stereo line
channels, 32 stereo FM stations, 32 AM
stations, 4 DAB+ ensembles and 32 internet
radio streams, limited only by the number of
PCIe slots on the motherboard that you use.
(See later for the PC-RACK units that can be
used with Flashlog 8 software).
The number of HDD logging days can be
independently set for each line, AM or FM
channel, and for each DAB+ ensemble.
Archiving also allows parallel recording of
selected channels to a permanent archive
location on the network, either continuously
or on selected days and time ranges.
All text fields are Unicode-based, allowing
multilingual display and entry.
Playback on both the logger and through
Flashback 8 supports saving log extracts as
MP3 as well as WAV, WMA and AVI files.
The Flashlog 8 recorder runs as a Windows
service and is controlled from under the
Up to 2 years of radio broadcasts or audio
can be logged, depending on available
hardware.
When also using the Sonifex PC-DAB DAB+
multi-ensemble PCIe Radcap card, the
software records the entire contents of up
to four ensembles. Decoding occurs during
playback, showing DLS text, MOT slide
shows and electronic programme guides
along with the audio.
Flashback 8 Playback Page.
174
Audio Logging
Sources tab in the Flashlog application.
Each line source can be configured as Stereo,
2 x Mono, 1 x Mono or Hidden. When set to
Hidden, the input is still being recorded, but
the source doesn’t appear on the playback
screen. The number of recording days dropdown list is dynamically populated to ensure
the hard drive space won’t be exceeded.
If line channel subcategories have been
created, the subcategory can also be selected
from a dropdown list. Each input can have a
12-character name, which is what appears
on the playback button for that channel, and
when 2 x Mono mode is selected, separate
names are given to each channel. The audio
compression rate can be set to 80kbps,
128kbps, 256kbps or lossless, and audio
failure alarms can optionally be enabled.
When used with the PC-AM and PC-FM radio
capure cards, AM and FM radio capture
sources are shown next, with edit fields for
the frequency and an optional descriptive
name. If RDS is present and no descriptive
name has been entered, the transmitted
station name is used and is dynamically
updated should it be changed by the station.
RF and audio signal levels are shown, along
with stereo and RDS indicator lights for FM
channels. Alarms (audio or carrier fail) can
be optionally enabled. A drop-down list
allows the number of logging days to be set,
with the range determined dynamically from
the available drive space.
When using the PC-DAB radio capture card,
each DAB ensemble has a spectrum display,
an RF level indicator, a phase reference
correlator level, an uncorrected error count
and an alarm enable checkbox. When alarms
are enabled, the Configure button opens a
dialog box allowing alarms to be enabled or
disabled on individual services. The number
of logging days for each ensemble is set
from a drop-down list.
Flashlog 8 AM & FM Radio Capture Source Page.
Finally shown are the internet radio streams
displaying the URL, stream status, bit-rate,
audio coding and audio levels, with dropdown lists for setting logging days and alarm
enable checkboxes. Audio level meters,
based on analysing the compressed audio
data, are also presented.
Additional buttons at the bottom of the
Sources window provide further source and
SNMP configuration.
The playback screen, available under
the Player tab on the Flashlog unit and
in the free Flashback network playback
application, provides easy selection of audio
source and playback date and time.
Across the top, tabs for each source
category or subcategory select a group of
up to 32 buttons, identified by the channel
name. In the case of DAB sources,
the broadcast short service name is used.
The waveform display can span one minute,
ten minutes, an hour or a day, by using
‘span’ buttons. The date is selected from a
calendar control, with the time set either
by using the up and down buttons or by
directly typing into the hours, minutes and
seconds fields. The playback position can
also be adjusted by dragging on the ruler
bar below the waveform.
A section of audio can be marked either
with the mouse in the waveform window
or by navigating to the start and end points
and clicking on the Mark button. Once
highlighted, the section can be saved to the
Windows clipboard or to a file in WAV, WMA
or MP3 format. For DAB channels, an AVI
video file incorporating the slideshow and
DLS text along with the audio can also be
created.
Flashlog 8 Source Page.
175
Audio Logging
PC-FLS8LT1 Flashlog 8 Single
Stereo Channel Logging Software
PC-FLS8LT2 Flashlog 8 Two
Stereo Channel Logging Software
New licences of the Flashlog 8
software are now available to allow
logging of either one or two stereo
channels, together with IP audio
streams.
Flashlog 8 DAB Inputs Page.
Aimed at the community radio
market, these new licences offer full
Flashlog functionality for logging
radio output.
A great value for money solution
using your own PC hardware.
Flashback 8 Playback Page.
Flashlog 8 Internet Radio Streams Page.
The display window on the lower
left shows DLS text for DAB services,
RDS radiotext for FM stations and
artist/title metadata for internet
streams. The window on the lower
right shows skimming events for line
channels, slideshows for DAB audio
services, a program guide tree view
for DAB EPG services and header
information for internet streams.
Flashback can be simultaneously
used on multiple workstations,
up to the limit imposed by the
recorder’s operating system (20
users for Windows 7/8/10 or the
number of installed CALs for server
versions).
176
Features
• Analogue, AES-3, Axia Livewire™,
Wheatstone Wheatnet™ & Audinate
Dante™ IP line inputs supported (Note
separate proprietary drivers needed from
the IP audio suppliers).
• Up to 64 stereo inputs supported, each
reconfigurable as a pair of mono inputs.
• Up to four assignable line sub-categories
for grouping channels.
• Selectable audio compression rates
ranging from 80kbps to lossless.
• Adjustable no. of logging days per input.
• Recording time of up to two years, settable
on each channel or ensemble (dependent
on no. of channels, audio compression rate
& HDD size).
• Operates on 64-bit Microsoft Windows™
7/8.1/10 Professional operating system.
• Network playback using free Flashback 8
DAB+ Radio Capture Using PC-DAB Card
• Up to four DAB+ ensembles.
• Adjustable no. of logging days per ensemble.
•
• Entire MSC (main service channel) contents
recorded.
•
• Receiver spectrum display, signal strength
and quality indicators, and uncorrected error
•
counter.
•
• Decodes legacy MP2 as well as HE-AAC v2 audio
AM & FM Radio Capture Using
compression.
PC-AM & PC-FM Cards
• Displays DLS text, MOT slides and electronic
• Up to 32 AM stations and 32 FM stations.
programme guide.
• Adjustable number of logging days on
• Live monitoring of audio and data.
each station.
• Selectable ensemble, stream and audio failure
• All FM logging in stereo.
alarms.
• Received signal strength, FM stereo
Internet Stream Capture
pilot and RDS indicators.
• Up to 32 internet radio streams.
• Live monitoring on logger unit.
• HLS, Shoutcast, Icecast & RTMP protocols.
• FM RDS radiotext logging.
• Selectable carrier & audio failure alarms. • MP3 and HE-AAC audio encoding.
• Metadata recording.
application (also compatible with
Flashlog 5, 6 and 7 recorders).
SNMP monitoring and alarms,
including RDS / DLS text.
Optional skimming control inputs
supported.
Live input monitoring on logger unit.
Selectable audio failure alarms.
Audio Logging
PC-RACK Industrial PC Racks for Audio Logging
Specification for Flashlog 8 Software
Recording Time: 7, 14, 28, 42, 60, 90, 100, 120, 180 days,
1 year or 2 years (individually settable on
each channel or ensemble)
Analogue Line
Inputs:
80kbps / 128kbps / 256kbps /
lossless compression (selectable per
channel).
Sonifex can supply industrial rackmount PCs to run the PC-FLS8 Flashlog logging software.
PC-RACK4D 4U Rackmount PC With
Front Panel Display
AES-3 Line Inputs: 32kHz – 96kHz 24-bit sampling
80kbps / 128kbps /256kbps / lossless
compression (selectable per channel)
Axia Livewire:
Optional software driver supporting up
to sixteen stereo inputs and playback
output.
Wheatstone
WheatNet:
Optional software driver supporting up
to sixteen stereo inputs and playback
output.
Internet Streams: RTMP, Shoutcast, Icecast, HLS stream
logging. MP3 and HE-AAC encoding.
Maximum 32 streams.
Skimming Option: Daily event list display linked to audio
log. Sequential playing and archiving
of multiple events. (Needs separate
hardware).
AM Capture:
500kHz – 1710kHz tuning range (1kHz
steps). 22.05kHz sampling. 32kbps
compression. Maximum 32 stations
FM Capture:
87.5 – 108.5MHz tuning range (50kHz
steps). 48kHz sampling. 64kbps
compression. Maximum 32 stations. RDS
radiotext logging.
DAB+ Capture:
Band III (174 – 240MHz) DAB Mode 1.
Maximum 4 ensembles. Maximum 32
services per ensemble.
Dynamic reconfiguration.
DAB+ and Legacy DAB supported.
Decoding for MP2 / HE-AAC v2
audio, DLS, JPEG / PNG images, EPG.
Internet Streams: Shoutcast, Icecast & RTMP protocols.
SNMP:
Input audio / carrier / DAB status
/RDS-DLS text monitoring.
Selectable trap generation.
Axia Audio® & Livewire® are trademarks of TLS Corporation.
WheatNet® & WheatNet-IP® are registered trademarks
of Wheatstone Corporation, Dante® is a trademark of
Audinate.
The PC-RACK4D
This is a rugged industrial 4U rackmount
PC to run the PC-FLS8 Flashlog 8 software
which has a front panel LCD display with
a pull-out drawer for the keyboard and
mouse. So, it’s completely self-contained
and useful for secure logging environments.
It has dual power supplies, 4 x PCI slots
and 8 x PCIe slots. It uses an Intel i7
processor, has 8GB RAM, runs Windows 10
Professional and uses a reliable batterybacked Adaptec SATA RAID with 3 x 2TB
Enterprise HDD. Finally there is also a
DVD-RW drive.
The 8 x PCIe slots allow this PC to be fitted
with multiple PC-AD2, PC-DIG4, PC-AUR44,
PC-FM, PC-AM and PC-DAB cards to
produce a very powerful audio and radio
capture logger.
PC-RACK4 4U Rackmount PC
This is similar to the PC-RACK4D but
doesn’t contain the integrated display
and has a separate USB keyboard and
mouse. It has dual power supplies, 7 x PCI
slots and 5 x PCIe slots. It uses an Intel i7
processor, has 8GB RAM, runs Windows 10
Professional and uses a reliable batterybacked Adaptec SATA RAID with 3 x 2TB
Enterprise HDD. There is also a DVD-RW
drive.
On this PC, there are 5 x PCIe slots which
can be fitted with multiple PC-AD2,
PC-DIG4, PC-AUR44, PC-FM, PC-AM and
PC-DAB cards.
PC-RACK1 1U Rackmount PC
This is a 1U industrial rackmount PC to
run the PC-FLS8 Flashlog 8 software. It
has a single PCIe slot so can only be used
with one PCIe audio or radio capture
card, ideal for logging up to 32 FM, or
32 AM channels or 4 DAB ensembles. It
uses an Intel i5 processor, has 8GB RAM,
runs Windows 10 Professional and uses
SATA RAID with 2 x 2TB HDD. It also has
a DVD-RW drive and separate keyboard
and mouse.
The PC-RACK1
The PC-RACK4
Please note that the PC-FLS8 Flashlog 8 software is not automatically included with these systems and must be ordered separately.
Also, as is common with PC systems, the specification can change so contact Sonifex to check the latest product specs.
177
Sonifex S2 Digital I/O Analogue Radio
Broadcast Mixer. Photo courtesy of Radio 10 Magic FM & Triple Audio.
178
Radio Broadcast Mixing Consoles
S0 Radio Broadcast On-Air Mixer
The S0 is a high quality yet simple to operate radio broadcast mixer ideally
suited to community radio stations, for educational purposes and for internet radio.
180
S1 Radio Broadcast On-Air Mixer
The Sonifex S1 mixer is a high performance compact, low cost, fixed format mixing
console designed for on-air radio use.
184
S2 Digital I/O Analogue Radio Broadcast Mixer
The S2 is a fully modular radio broadcast mixer offering digital audio quality with
analogue reliability, using the latest technology components.
190
S2 Input Channels
191
S2 Output Channels
201
S2 Meterbridge Modules
205
S2 Chassis Sizes
209
S2 Script Spaces
210
S2-PSU Power Supply
212
S2-PSUS Dual Power Supply Switcher
213
Sonifex S2 Digital I/O Analogue
Radio Broadcast Mixer. Photo
courtesy of www.jingle24.com.
179
Radio Broadcast Mixing Consoles - S0
S0 Radio Broadcasting On-Air Mixer
Category: Radio Broadcast Mixing
Consoles - S0 Radio Broadcast Mixer.
Product Function: 9 Channel analogue
fixed format on-air broadcast mixer.
Typical Applications: Community,
student, hospital and small scale radio
broadcast mixer, secondary or backup
on-air mixer.
Features:
• User friendly broadcast mixer.
• Clear, simple layout with no jargon.
• Nine multi-function channel mixer.
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
Built in telephone interface.
Controllable by automation system.
Skype™ integration for voice calls.
Built in headphone volume limiter.
Large, simple LED volume display.
Remote output for fader starts.
Speaker muting when ‘Mics’ are on.
External mic-light switching output.
‘Programme’ and ‘record’ outputs.
‘Aux’ input for iPod or MP3 players.
Four microphone/line channels.
Four stereo line channels.
Switchable telephone/AUX channel.
Stereo USB audio to and from a PC.
Guest headphone ‘talkback’.
Reliable, low cost mixing solution.
A simple radio mixer for novice
and professional users…
The S0 is a high quality yet simple to
operate radio broadcast mixer ideally
suited to community radio stations, for
educational purposes and for internet
radio.
Easy to understand, the S0 includes a
telephone hybrid for making and recording
telephone calls and a 3.5mm stereo jack
for plugging in an mp3 player. Skype calls
can be cued via USB and the S0 can be
controlled by automation systems.
The addition of a USB port allows for
recording to a PC and for playing a PC
automation system directly through the
mixer. The headphone outputs have a built
in limiter to offer hearing protection and the
studio speakers mute when a microphone
fader is open, with automatic mic live sign
switching. The S0 allows presenters and
DJs to be up and running quickly with a fully
featured radio studio mixer.
The Sonifex S0 mixer is a compact, low
cost, fixed format broadcast mixing console
designed for on-air radio use. It uses the
same high quality circuitry and components
as the Sonifex S2 and S1 mixers to provide
an audio experience second to none. The S0
can be fitted flush into a desk-top or can be
rack mounted directly using the front panel
mounting holes.
S0 Radio Broadcast Mixer Rear.
180
Radio Broadcast Mixing Consoles - S0
8. Alternatively, this signal can be routed to
channels 5, 6 or 7 if required.
The uncomplicated and intuitive front panel
layout ensures that the unit appeals to
novices and broadcast professionals alike,
whilst a range of user configurable options
allows for flexible operation.
Output Channels
The S0 has separate stereo PGM and REC
bus outputs. The PGM bus is output on
balanced stereo XLRs and the REC bus is
output on unbalanced stereo RCA phono
connectors. There are monitor outputs for
presenter headphones, guest headphones
and loudspeakers.
The console consists of nine input channels:
• 4 x mono balanced XLR mic/stereo
unbalanced RCA phono line inputs.
• 4 x dual stereo unbalanced RCA phono
line inputs.
• 1 x telephone balance unit (with line and
handset ports)/stereo unbalanced RCA
phono auxiliary input with a parallel
3.5mm stereo jack input on the front
panel.
Providing in total 4 mic, 12 stereo line, 1
TBU & 1 stereo auxiliary inputs which you
can switch between.
Input Channels
Input source buttons at the top of each
channel strip are used to select the required
mono or stereo source. The mixer has two
main stereo buses, PGM (Program) and REC
(Record), so each channel also has PGM and
REC buttons to independently select which
mixer bus the selected input is routed to.
The XLR microphone inputs on channels 1 to
4 have individually selectable +48V phantom
power and a gain calibration potentiometer
providing up to 65dB of gain for the pre-
S0 Radio Broadcast Mixer Iso View.
amp. Input channel 1 also serves as the
microphone input for a dedicated talkback
channel.
Gain for each channel is trimmed by the
front panel TRIM control providing ±15dB of
gain. A PAN/BAL(ance) control is available to
facilitate stereo imaging.
The stereo RCA phono line inputs on channels
The use of VCAs controlled by the ALPS
1 to 8 have 10dB of gain at the input to
compensate for unbalanced consumer inputs. long throw 100mm faders gives a smooth,
repeatable response and ensures tight
Input channel 9 is a TBU and stereo auxiliary
stereo tracking while eliminating mechanical
input channel. The TBU allows direct
and electronic noise.
connection to a telephone line and allows
USB Audio for Playback and
calls to be made and received through the
Recording
mixer using the handset connection. The
The S0 has the option to send and receive
auxiliary channel can switch between 2
audio over USB. This allows the audio on
independent inputs, one on the rear panel
the REC bus to be sent to a PC for recording
and one on the front panel.
or monitoring purposes. Also, the S0 can
Any channel which has the fader up is routed
receive a USB audio stream from a PC and
to the selected output, either PGM or REC
route it to the auxiliary inputs on channel
or both.
Monitoring & Headphone Limiter
The monitor loudspeakers, presenter
headphones and guest headphones are
on 6.35mm stereo jack sockets and the
headphones can be plugged in to the
front and rear of the mixer. The monitor
loudspeaker and headphone levels are
independently variable between 0 (cut off)
and 10 (max).
With the concerns over listening levels
being too high in headphones, the addition
of an adjustable limit level potentiometer
on the rear panel of the mixer is a great
idea which limits the maximum level of the
audio routed to the presenter and guest
headphone outputs. An illuminated MUTE
LED shows when a live microphone channel
has muted the loudspeakers and there is a
MUTE contact output available to illuminate
a ‘MIC LIVE’ light via a 6.35mm stereo jack
socket on the rear panel.
181
Radio Broadcast Mixing Consoles - S0
Each channel input has its own START
and STOP remote controls which can be
triggered when the channel is routed
to the PGM or REC bus and the fader is
opened or closed. The remotes can be
set-up to be either pulsed or continuous
latched outputs, and if in pulsed mode
the START button has the capability to
produce repeated start pulses.
S0 Radio Broadcast Mixer Top View.
A three way electronically interlocking
illuminated switch bank selects the source
routed to the loudspeaker and headphone
outputs from either PGM, REC or from an
additional unbalanced stereo RCA phono
input EXT 1. This external input is used for
monitoring an off air signal or studio output.
Green illuminated AUTO CUE/PFL (pre fade
listen) buttons adjacent to each level control
allow the automatic monitoring of any
channel that has been selected to pre-fade,
either to the monitors or headphones.
For the presenter headphones, SPLIT CUE/
PFL can be selected which places the
selected source in mono in the left ear, and
pre-fade in mono in the right ear.
182
Metering
A pair of bright 21 segment LED meters can
be configured to show either VU or PPM
metering. The meters follow the selection
of the presenter headphones including any
pre-fade or split pre-fade function.
Talkback
A separate TALKBACK button is provided
to allow the presenter to talk to a guest on
the guest headphones. The S0 uses input
channel 1 as the talkback source.
Channel Remotes
The remote outputs for each channel
are highly configurable providing a
comprehensive range of options to
interface external equipment to the mixer.
Configuration Settings
The S0 can be configured using the
Sonifex SCi software which uses the USB
serial connection. SCi offers a full range
of software configurable options which
can be used to customize the operation
of the mixer. It is possible to enable or
disable specific inputs, enable phantom
power for the microphone channels and
limit which buses each channel can be
routed to. Other settings control auto
cancelling of PRE FADE when the channel
fader is opened and all the remote output
configurations.
The S0 has an integral universal switch
mode power supply, which uses an IEC
mains inlet.
Radio Automation Software
Control of the Sonifex S0 Mixer
The S0 firmware has been updated to
include a new protocol which enables
radio automation software from a PC to
control the mixer.
SCi Configuration Page.
SCi Info & Updates Page.
More protocols were added which give 3rd
party remote software access to the unit,
offering the facility to radio automation
systems of control of every function on
the S0.
Radio Broadcast Mixing Consoles - S0
Skype™ Compatibility
The S0 now mixer allows you to operate
Skype™ audio on channel 9, with
the only interconnection being a USB
cable. The S0’s USB audio feature has
previously allowed either PROGRAMME
or RECORD bus audio (selected by
jumper) to be routed to a PC via USB
interface for the purpose of audio
recording, with a return path useful for
providing a pre-fade listen (PFL) for a
playout system.
A third option allows you to route
the cleanfeed output from the telco
channel to a PC via USB. Setting both
the “microphone” and “speakers”
within Skype to S0 then allows full
duplex communication between the S0
presenter and a caller, enabling the user
to audition the caller on PFL to set levels
before either going to air or recording a
feature.
Specification For S0
Input / Output Impedances
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Inputs:
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Input / Output Gain Range
Mic Input:
Preset pot +24dB to +67dB ref
-50dBu,
TRIM pot ± 15dB
Line Input:
+10dB ref 0dBu at PGM output,
TRIM pot ± 15dB
Frequency Response
Mic Input:
40Hz to 20kHz –1dB, +0dB
Line Input:
20Hz to 20kHz –0.5dB, +0dB
Noise (20Hz to 20kHz)
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Stereo Inputs:
-92dBu ref 0dB (fader down,
no routing)
Distortion
Total Harmonic
Distortion:
0.015% at 1kHz, 0.015% at
10kHz ref +8dBu
Range
Pan Range:
Off/-3dB Centre/Off
Balance Range:
± 6dB
Common Mode Rejection Ratio
Mic Input:
> 60dB typically
439
Output
Maximum PGM
Output:
+26dBu balanced
Maximum REC
Output:
+16dBu unbalanced
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Input & Output Connections
Audio Inputs:
11
4 x Microphone XLR-3 pin sockets
12 x Pair stereo line RCA phono sockets
1 x Pair stereo aux RCA phono sockets
1 x Stereo aux 3.5mm jack socket
Audio Outputs:
PGM:
2 x XLR-3 pin plug (balanced)
REC:
1 x Stereo pair RCA phono sockets
Monitor Inputs:
1 x Stereo pair RCA phono sockets
Monitor Outputs: 5 x 6.35mm (1/4”) stereo jack sockets
(2 x presenter, 2 x guest, 1 x
loudspeaker)
335.5
355
Remote Outputs: 9 x 6.35mm (1/4”) stereo jack
sockets (one per channel)
1 x 6.35mm (1/4”) stereo jack
socket for light control
Telephone:
2 x RJ11 6/4 (1 x line, 1 x handset)
USB Audio:
1 x Type-B receptacle
USB Serial:
1 x Type-B receptacle
Mains Input:
Filtered IEC, continuously rated
85-264VAC, 47-63Hz, 45W nominal,
>50W peak
Fuse Rating:
Anti-surge fuse 2A 20 x 5mm
Equipment Type
S0:
482.6
2
105
9
S0 Radio Broadcast Mixer Dimensions (in mm).
S0 radio broadcast mixer
Physical Specification
Dimensions
(Raw):
48.3cm (W) x 35.6cm (D) x 12.5cm (H)
19” (W) x 14” (D) x 4.9” (H)
Dimensions
(Boxed):
67cm (W) x 44cm (D) x 25cm (H
26.4” (W) x 17.3” (D) x 9.84” (H)
Cut-Out
Dimensions:
44cm (W) x 34.7cm (D)
17.32” (W) x 13.66” (D)
Weight:
Nett: 8.5kg Gross: 10.12kg
Nett: 18.7lb Gross: 22.26lb
183
Radio Broadcast Mixing Consoles - S1
S1 Radio Broadcast On-Air Mixer
The small radio mixer that’s big on
features.
The S1 has an amazing feature set and
offers a proper broadcast solution to
community, educational and student radio
stations that need a versatile, reliable radio
broadcast mixer. It has a ‘no compromise
on quality’ approach with excellent mic
amps, comprehensive facilities & simple,
reliable operation.
S1 Digital/Analogue Radio Broadcast Mixer.
Category: Radio Broadcast Mixing
Consoles - S1 Radio Broadcast Mixer.
Product Function: 9 Channel analogue
fixed format on-air broadcast mixer.
Typical Applications: Community,
student, hospital and small scale radio
184
broadcast mixer, secondary or backup
on-air mixer.
Features:
• High performance, ultra-reliable on-air
radio mixer.
• Low cost and easy to use.
• Small footprint, flush-mounting or
rack-mounting.
• Analogue & digital inputs & outputs.
• Main stereo output bus plus 2 Aux
output buses, pre or post fader
selectable.
• Rotating rear panel for ease of
installation.
• Integrated power supply.
• Remote outputs per channel.
• Fader start of ancilliary equipment such
as CD players.
• Programmable general purpose I/O.
• Monitor muting on mic live.
• Mic-live/on-air sign switching.
• Monitor outputs for presenter & guest
headphones, & speakers.
• Bright LED meter display PPM/VU
selectable with separate meterbridge
output.
• 100mm ALPS long-throw faders with
control via VCAs.
• Separate telco & stereo cleanfeed
o/ps for use with a telephone hybrid &
codec respectively.
• Advanced automation facilities.
• Remote control via free SCi software.
Radio Broadcast Mixing Consoles - S1
S1 Digital/Analogue Radio Broadcast Mixer, Rear View.
The Sonifex S1 mixer is a high performance
compact, low cost, fixed format mixing
console designed for on-air radio use. It
can be fitted flush into a desktop or can be
rack-mounted by the addition of optional
rack ears. The rear connectivity panel can
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providing easier connection to the unit
when it is rack-mounted. It has an integral
universal switch mode power supply, which
uses an IEC inlet.
It maintains the same front panel ease of
use found on the Sonifex S2 whilst providing
advanced automated features not seen on
any other product currently available on
the market.
The console consists of ten input channels:
• 2 x mono XLR mic/mono XLR line inputs.
• 1 x mono XLR mic/mono XLR telco input.
• 2 x mono XLR mic/mono XLR line inputs/
stereo ¼” jack unbalanced line input.
• 1 x stereo XLR line/stereo XLR cleanfeed
input.
• 2 x stereo XLR line/stereo RCA inputs, one
with a 3.5mm stereo aux input.
• 2 x stereo XLR line/stereo SPDIF & Toslink
digital inputs.
Providing in total 5 mic, 4 mono line, 10
stereo analogue, 2 stereo digital, 1 telco &
1 stereo cleanfeed inputs which you can
switch between.
Wide-Ranging Input Channels
Input source buttons at the top of each
channel strip are used to route the connected
source to the mixer buses.
All of the balanced XLR input connectors have
a 10dB gain switch on the rear panel for use
with unbalanced or low level input sources.
The XLR microphone inputs have individual
+48V phantom power switches and a gain
calibration potentiometer providing up to
65dB of gain for the preamp.
185
Radio Broadcast Mixing Consoles - S1
Input channels 1 & 2 have 2 separate
XLRs for microphone & balanced line
input sources.
Input channels 3 and 4 use an XLR and a
combination XLR/stereo jack connector
to allow a microphone input and both a
balanced mono or an unbalanced stereo line
input to be used.
Input channel 5 acts as microphone input or
as a telco input with a dedicated cleanfeed
output, with available line hold remote for
connection to a telephone hybrid.
Input channel 6 is a balanced stereo
line input on XLR together with a stereo
cleanfeed input on balanced XLR for use
with ISDN codecs.
Input channels 7 & 8 have stereo balanced
XLRs and stereo unbalanced phonos on the
rear and channel 7 also has a 3.5mm insert
jack socket on the front panel for use with
an mp3 player/iPod.
S1 Digital/Analogue Radio Broadcast Mixer, Top View.
Input channels 9 and 10 provide a stereo
analogue XLR input and stereo digital S/PDIF
and TOSLink optical inputs.
Any channel which has the fader up and the
ON button selected is routed to the main
PGM output(s).
Gain for each channel is trimmed by the
front panel TRIM control providing ± 12dB
of gain.
186
Equalisation is provided on every input
channel, allowing up to 7dB cut and boost
at HF (6.5 kHz) and LF (100Hz). A PAN/
BAL(ance) control is available to facilitate
stereo imaging.
The use of VCAs controlled by the ALPS
long throw 100mm faders give a smooth,
repeatable response and ensures tight stereo
tracking while eliminating mechanical and
electronic noise.
Analogue & Digital Outputs
The S1 has a multitude of output connectivity:
one balanced stereo XLR PGM output, two
balanced stereo AUX outputs on a 9-way
D-type socket, a mono clean feed output
on balanced XLR (for use with a telephone
hybrid), a stereo cleanfeed output on stereo
balanced XLRs (for use with an ISDN/IP
codec) and monitor outputs for presenter
headphones, guest headphones and
loudspeakers. Additional outputs are provided
for talkback on a 9-way D-type socket and
metering using a 15-way D-type socket.
A digital stereo PGM output is provided in
XLR AES/EBU or phono S/PDIF formats. The
output can be referenced to an internal clock
source between 32 kHz and 96 kHz or external
synchronisation sources can be connected
via S/PDIF, AES/EBU or BNC wordclock. The
illuminated DIGITAL SYNC LED shows when
the digital output is locked to either an
internal or external clock.
Radio Broadcast Mixing Consoles - S1
Two Auxiliary Buses
The S1 has two Auxiliary buses that can be
configured in either pre-fade or post-fade
mode. Each channel can be routed to these
buses using the potentiometers provided on
each channel strip. Each AUX output has a
trim control knob operating between fully
attenuated and unity gain. Aux buses can be
used for a wide range of purposes including
recording telephone conversations,
generating different mixes for output,
connecting to audio processing equipment
and much more.
Comprehensive Monitoring
The control room loudspeakers, presenter’s
headphones, and guest headphones are
on 6.35mm stereo jack sockets. Front
panel control knobs are used to vary the
monitor and headphone levels and the
presenter’s headphones can be plugged
in to the front and rear of the mixer. The
control room monitor loudspeaker and
headphone levels are variable between 0
(cut off) and 10 (max). An illuminated MUTE
LED shows when a live microphone channel
has muted the speakers and there are two
pairs of MUTE contact outputs available to
illuminate a ‘MIC LIVE’ light.
A five way electronically interlocking switch
bank selects the source routed to the
speakers and presenter’s headphones from
either of two external inputs (EXT 1, EXT
2), the PGM, or AUX Outputs. The buttons
illuminate to show the selected source,
PGM, AUX 1 and AUX 2 in green, EXT 1
and EXT 2 in red. The external inputs can
be used for monitoring an off air signal or
another studio output.
Green illuminated AUTO CUE/PFL buttons
adjacent to each level control allow the
monitoring of PFL when an input channel
has been selected to CUE/PFL, either to the
monitors or headphones.
For the Presenter headphones, SPLIT CUE/
PFL can be selected which places the
selected source in mono in one ear and PFL
in mono in the other.
A pair of bright 21 segment LED meters
can be configured to show either VU or
PPM metering. The meters show either
the PGM output or the selection heard in
the presenter’s headphones, selectable by
PGM/MFM (Meter Follows Monitor) button.
Talkback
A separate TALKBACK button is provided
to allow the Presenter to talk to a Guest.
The S1 uses input channel one as the
talkback source. Global talkback can also
be configured (see the Programmable GPIO
section) to allow every mic channel to
communicate with the Presenter and Guest
headphones.
A separate TALKBACK 9-way D-type socket
on the rear panel can be used to connect
the talkback output and input to and from
other studios.
Configurable Remotes Per Channel
The remote outputs for each channel are
highly configurable providing every possible
option required to interface your external
equipment to the mixer. Each channel
input has its own START and STOP remote
controls, except the microphone inputs
which have only START outputs. (If you
wanted a MIC stop control signal, you could
configure this using a GPIO). Each remote
can be configured to be triggered from a
variety of sources, including the ON button
and the CUE/PFL Button. The remotes can
be set-up to be either pulsed or continuous
latched outputs, and if in pulsed mode the
ON button has the capability to produce
repeated start pulses on additional ON
button presses, when the channel is ON
with the fader up.
In the case of the Telco & Stereo Cleanfeed
outputs you may want to generate start and
stop remotes independently of the position
of the fader, so that the hybrid or codec can
be triggered to connect before the channel
is opened. This is possible in ‘Telco Mode’
where a ‘start’ is sent when the ON button
is initially pressed & a ‘stop’ is sent when
the ON button is pressed again, turning
the channel selection off. In addition the
channel can be configured to show the state
of the external equipment, e.g. using a tally
back GPI signal from an automation system,
the ON button can flash green when the end
of a song is being reached.
Programmable GPIOs
The S1 also has five General Purpose
Inputs/Outputs (GPIO’s) available. Each
input/output has been designed to be
very versatile & almost any operation with
the mixer can be achieved when utilising
them. A GPI can be configured using the
SCi software to perform up to 10 different
operations from a single input.
For example, in an automated playout
situation, you may want to:
• Force 4 channels ON with the faders
forced up.
• Force 4 channels to stereo Line Input.
• Mute the Control Room Monitors.
S1-RCK Optional S1 9U 19” Rack Ears.
187
Radio Broadcast Mixing Consoles - S1
Specification For S1
Crosstalk
Input / Output Impedances
Inter-Channel:
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Equalisation
The GPOs are equally powerful. The
outputs can be configured to be pulsed
or continuous, and active high or low. An
output can be generated based on actions
required. For example, a GPO can be
configured to generate a global talkback
function with a set-up so that:
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LF:
Shelving at 100Hz ± 7dB
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HF:
Shelving at 6.5kHz ± 7dB
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Range
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• When Mic 2, Mic 3 & Mic 4 are all not
live, generate a GPO continuous output.
Input / Output Gain Range
This configuration would use 9 of the 10
operations available for an individual GPI.
The GPO could then be linked back
into one of the GPI’s with the input
configured to:
This would allow the guests to freely
talk to each other until one of their
microphones goes live. This is just an
example of the potential uses of the
programmable GPIOs.
Remote Control & Configuration
Using SCi Software
The S1 mixer is highly configurable and
has numerous options allowing the mixer
configuration to be tailored to your
requirements. Using the freely available
Sonifex SCi software, every parameter of
the mixer can be set up with ease.
188
Pan Range:
Off/-3dB centre/off
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Balance Range:
± 6dB
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Output:
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Common Mode Rejection Ratio
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Input:
ϱϬё
Mic Input:
>60dB typically
Digital I/O
Sync Input Sample
Rate Range:
32kHz – 96kHz
Mic Input:
Preset pot +21dB to +64dB ref
-50dBu, TRIM pot ± 12dB
Output Sample
Rates:
32kHz, 44.1kHz, 48kHz, & 96kHz
(Using Onboard Clock)
Mono Line
Switch 0dB to +10dB ref 0dBu, Input:
TRIM pot ± 12dB
Output Sample
Width:
24 bit
Stereo
Line Input:
Switch 0dB to +10dB ref 0dBu,
TRIM pot ± 12dB
Output
Telco Input:
Switch 0dB to +10dB ref 0dBu,
TRIM pot ± 12dB
Stereo Cleanfeed
Input:
Switch 0dB to +10dB ref 0dBu,
TRIM pot ± 12dB
Digital Input:
0dBFS = +18dBu on input;
TRIM pot ± 12dB allowing 0dBu
to +24dBu
SCi Channel Configuration Page.
• Force PFL on Mics 2, 3 & 4.
• Enable Auto CUE/PFL on Guest
H/phones.
< -80dBu typically
Digital Output:
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Output Load:
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Maximum Output
(Analogue):
+26dBu balanced into 2k or greater
Rear Panel Switches
Phantom Power:
+48V individual per mic input
Gain Switch:
+10dB per XLR balanced mono &
stereo input & EXT 1, EXT 2 inputs
0dBFS = +18dBu
Frequency Response
Input & Output Connections
Mic Input:
40Hz to 20kHz –1dB, +0dB
Analogue Inputs:
Line Inputs:
20Hz to 20kHz –0.5dB, +0dB
RCA Inputs:
20Hz to 20kHz –0.5dB, +0dB
5 x Microphone XLR 3 pin sockets
2 x Mono line XLR 3 pin sockets
2 x Mono/stereo line combi XLR 3 pin
sockets & 6.35mm stereo jack sockets
1 x Telco XLR 3 pin socket
5 x Stereo line pairs XLR 3 pin sockets
2 x Stereo line pairs phono sockets
1 x Stereo cleanfeed pair XLR 3 pin
sockets
1 x Stereo insert, 3.5mm jack socket
Digital Inputs:
2 x S/PDIF phono sockets,
2 x TOSLink optical connectors
Noise (20Hz to 20kHz)
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Stereo Inputs
SCi System Configuration Page.
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-92dBu ref 0dB (fader down, no
routing)
-91dBu (fader down, one channel
routed)
-86 dBu (unity gain, one channel
routed)
-83 dBu (unity gain, two channels
routed)
Distortion
Total Harmonic
Distortion:
0.015% at 1kHz, 0dBu
0.015% at 10kHz, 0dBu
Analogue Outputs:
PGM:
AUX:
TELCO:
STEREO CLEANFD:
2 x XLR 3 pin plug (balanced)
9-way ‘D’-type socket (balanced)
XLR 3 pin plug (balanced)
2 x XLR 3 pin plug (balanced)
Radio Broadcast Mixing Consoles - S1
156mm
Monitor Outputs:
6.35mm stereo jack sockets (2 x
Presenter, 1 x Guest, 1 x Monitor)
Digital Outputs:
AES/EBU XLR 3 pin plug (balanced) or
S/PDIF phono socket.
Digital Sync Inputs: Wordclock on BNC, AES/EBU on XLR
socket and S/PDIF on phono socket.
EXT Inputs:
9-way ‘D’-type socket
Mono Remotes:
25-way ‘D’-type plug
Stereo Remotes:
25-way ‘D’-type plug
Meterbridge Port:
15-way ‘D’-type socket
Talkback Interface: 9-way ‘D’-type socket
RS232 Serial Port:
9-way ‘D’-type socket
Mains Input:
Filtered IEC, continuously rated
85-264VAC, 47-63Hz, 45w nominal,
50w peak
Fuse Rating:
380mm
400mm
Anti-surge fuse 2A 20 x 5mm
Equipment Type
S1:
S1 radio broadcast mixer
Physical Specification
Dimensions
(Raw):
44.5cm (W) x 40.0cm (D) x 15.6cm (H)
17.5” (W) x 15.75” (D) x 6.1” (H)
Dimensions
(Boxed):
62cm (W) x 58cm (D) x 36cm (H)
24.40” (W) x 22.83” (D) x 14.17” (H)
Cut-Out
Dimensions:
44.0cm, 17.3” (W) x 38.1cm, 15” (D)
both dims +0.2cm, -0cm
Weight:
Nett: 12.0kg Gross: 13.6kg
Nett: 26.4lb Gross: 29.9lb
340mm
Accessories
S1-RCK:
Optional S1 9U rack ears.
10mm
439mm
445mm
2mm
59mm
S1 Digital/Analogue Radio Broadcast Mixer Dimensions.
189
Radio Broadcast Mixing Consoles - S2
S2 Digital I/O Analogue Radio Broadcast Mixer
Category: Radio Broadcast Mixing
Consoles - S2 Digital I/O Analogue Radio
Broadcast Mixer.
Product Function: Modular multichannel analogue on-air broadcast mixer
with digital I/O.
Typical Applications: Community,
student, hospital and small scale radio
broadcast mixer, secondary or backup
on-air mixer.
Features:
• Wide range of input and output
channels.
• PFL/cue buttons & bus.
• Fader-start operation of equipment.
• Automatic monitor muting on miclive.
• Light controlling remote outputs.
• Optional EQ on input modules.
• Gram amp input options.
• 2 main audio buses, allowing you
to broadcast on the PRG bus while
recording on the AUD bus.
• Bus output selection on each module.
• Separate 1U power supply.
190
S2 is a modular broadcast mixer which
offers digital audio quality with analogue
reliability in a modular format. S2 has
both digital and analogue input channels,
together with simultaneous analogue and
digital outputs.
Following on from the Sovereign range of
audio mixers, S2 combines all the features
needed of a radio broadcast mixer in a
stylish, flush-mounting chassis
Innovative Design
• The S2 chassis is available in 5 module
width sections, allowing 5, 10, 15, 20, 25
and 30 module width mixers. This means
S2 can be used for small newsrooms or
large on-air situations.
• Modular “pop-up” input and output
channels means that the mixer can
be maintained simply and quickly.
Hot-swappable input channels can be
individually removed and repaired whilst
still on-air.
• The angle of the meterbridge can be
varied and set for best viewing position.
• Any module can be in any position so
that customising the mixer for your own
purposes is easy.
• Large back-lit buttons allow you to see
the status of the mixer at all times.
• The fitted rear panel hides all cable
connections but can be simply unclipped
and reclipped for maintenance.
• The modular design of the desk gives you
the flexibility to expand it at a later date
• You can even add another S2 mixer
and link them together with a bus
connector cable to allow for split
desk configurations.
Radio Broadcast Mixer Using S2-25 Chassis
and Script Space.
Radio Broadcast Mixing Consoles - S2 Input Channels
S2 Input Channels
Superb Audio Quality and
Unquestionable Reliability
• The S2 uses Crystal semiconductor parts
to allow input and output of digital audio
signals up to 24 bit, 96kHz sample rate.
• The analogue signal paths use low
noise circuitry to provide superb audio
performance well capable of satisfying
radio listeners worldwide.
• The high reliability and build quality of S2
minimises the chance of failure, avoiding
lost air-time. Each module is individually
checked twice before being assembled into
the finished chassis and the whole unit is
tested before shipping.
• ALPS long throw 100mm faders give a
smooth, repeatable response and the XLR
Neutrik connectors used are an industry
standard.
• The use of VCAs controlled by the faders
ensures tight stereo tracking and eliminates
mechanical and electronic noise.
• S2 has a separate 1U power supply
providing regulated, ripple-free power to
the mixer. There is an optional switcher
to control 2 power supplies providing
redundancy.
• High quality stainless steel is used for the
chassis and screws to prevent corrosion in
high humidity environments.
• Each channel is metal coated internally to
provide exceptional EMC screening.
There is a wide choice of input channels for
the S2 mixer. Each channel has a number of
common features:
Assigning an Input Channel to an
Output Bus
Switches at the top of the channel are used
to select the output group routing, to either
Program, Audition, or both output buses.
Selecting the PGM and/or AUD buttons
routes the channel audio to the PGM and/
or AUD mix buses. The buttons illuminate
green to indicate the routing status.
Changing the Input Level
Coarse gain is set using pre-set
potentiometers on the channel circuit
board which allow unbalanced inputs to be
used on mono and stereo line inputs. The
front panel TRIM control allows fine gain of
±12dB.
Changing the Signal Pan, or Balance
The BAL/PAN control is used on mono
channels to pan the mono input signal in
the stereo image and on stereo channels to
balance the stereo image.
CUE/PFL (Pre Fade Listen)
Selecting the CUE/PFL button routes the
pre-fader input signal to the monitoring
system where the signal can be heard via
headphones and/or loudspeakers. The
button lights green when CUE/PFL is active
and a jumper option is available to cancel
the CUE/PFL selection when the fader
is raised. This button works with both a
momentary and latched operation - if held
down, the selection is cancelled on release,
otherwise the button is alternate action.
Fader Start & ON Button Control
The ON button works in conjunction with
the 100mm long-throw carbon fader and
is used to control channel remotes (e.g.
starting a CD player), routing and timers,
etc. When unlit, the channel is off. Flashing
red indicates that the channel has been
selected to ON but not routed to either
PGM or AUD. Steady red indicates that the
channel is ON and “armed”, ready for the
fader to be raised. Raising the fader changes
the illumination to green indicating that
the channel is live. Alternatively, with the
button unlit the fader may be raised and the
channel can be operated simply by selecting
ON. The illumination in this case toggles
between unlit (channel OFF) and green
(channel ON).
either operate manually, be permanently
on, or permanently disabled. Also, the ON
button can be configured to operate in a
number of modes, altering remote start
functions, and the channel ON function
can be controlled remotely by automation
playout systems, such as RCS® or VCS.
Scribble Pad
A scribble pad is provided at the bottom of
the channel for user labelling of the channel
function.
RCS® is a registered trademark of Radio Computer
Systems Inc.
VCS is a wholly-owned subsidiary of SciSys plc.
Programmable Button Settings
To make the S2 modules as flexible as
possible, every button on each channel can
be set to a number of different modes to
aid the use of the mixer and allow for extra
functionality. Each button can be set to
191
Radio Broadcast Mixing Consoles - S2 Input Channels
S2-CMM Mic/Mic Channel
The mic/mic input channel is a dual mono
microphone input with a button switch to
select either Mic 1 or Mic 2.
Each Mic input has jumper settable
phantom power available at +48V and a
high pass filter to remove low frequency
rumbles. When enabled, the LF response
of the microphone is rolled off at 125kHz,
6dB per octave. If the Mic input is used for
the presenter’s mic it can also become the
talkback mic.
Equalisation is fitted as standard on this
channel and is enabled by the EQ button,
S2-CMM Mic/Mic Channel.
providing 10dB cut and boost at HF (6.5kHz)
and LF (100Hz).
The BAL/PAN control operates as follows:
Full anti-clockwise pans the signal to the
left and increases the signal by 3dB (right
channel reduces by 70dB); full clockwise
pans the signal to the right and increases
the signal by 3dB (left channel reduces by
70dB).
There are logic remote input controls for
Mic Cough muting and Reverse Talkback,
together with 2 separate output controls for
Mic Cue lights, momentary or latching.
The remote outputs on the 9 way D-type
plug are fed from NPN opto-isolators.
A balanced line level insert send and return
is available for the Mic input.
Both Mic inputs have a jumper selectable
output to facilitate either Control Room
speaker or Studio speaker muting and the
channel can also be used to control either
one of two separate timer displays.
This is useful for hooking up an outboard
effects unit to be used with the microphone.
Gold plated Neutrik XLR connectors are used
for the audio inputs and outputs.
S2-CML Mic/Line
Channel
The mic/line input channel is a mono input
with a button switch to select either Mic or
Mono Line.
For the Mic input, phantom power at +48V
can be configured. The Mic input has a
jumper selectable high pass filter to remove
low frequency rumbles.
If the Mic input is used for the presenter’s
mic it can also become the talkback mic. The
talkback function will be retained when the
channel input is switched to Line.
S2-CML Mic/Line Channel Rear.
S2-CMM Mic/Mic Rear.
S2-CML Mic/Line Channel.
192
Radio Broadcast Mixing Consoles - S2 Input Channels
Equalisation is fitted as standard on this
channel and is enabled by the EQ button,
providing 10dB cut and boost at HF (6.5kHz)
and LF (100Hz).
remote input controls for Mic Cough muting
and Reverse Talkback. There are output
controls for Mic Cue lights and separate
line remote start/stop controls by fader or
ON button. Both latching and momentary
contacts are catered for. The remote outputs
on the 9 way D-type plug are fed from NPN
opto-isolators.
A balanced line level insert send and
return is available for the Mic input for the
insertion of an effects, or voice, processor
into the mic channel. The send and return
is disabled in Line mode. There are logic
The Mic input provides a jumper selectable
48v
Mic
Line
Mic 2
Mic
Gain
Insert
Line
48v
Line
Trim
VCA
Trim
Routing
EQ
HF/LF
EQ
PAN
PFL
In addition the Line input can be configured
with all the features of the Mic input
when used as a Mic input with an external
microphone amplifier, e.g. for monitor
muting purposes.
Gold plated Neutrik XLR connectors are used
for the audio inputs and outputs.
To left
Pgm output
The S2-CS dual stereo line channel has two
balanced stereo inputs on Neutrik
XLR connectors.
The INP 1/INP 2 button selects which input
is routed through the channel. The button
is illuminated in red to indicate when Input
2 is selected. The operation of the button is
inhibited when the channel is “live”.
To right
Pgm output
VCA
Inse RT
Send
Fader
CPU
To timer
Bus
Inse RT
Return
Mic cue light
Cough/rev t/b
PGM AUD PFL C/FEED T/B
L R L R L R 1 2 1 2
Mutes
S2-CMM Mic/Mic Channel and S2-CML Mic/Line Channel.
output to facilitate either Control Room
speaker or Studio speaker muting and the
channel can also be used to control either
one of two separate timer displays.
S2-CS Dual Stereo
Line Channel &
S2-CSE Dual Stereo
Line Channel with EQ
PGM
Line remotes
T/b enable
Aud
Mic
Line
EQ
Cue
PFL
ON
To meter
Select
S2-CS Dual Stereo Line and S2-CSE Dual Stereo Line with EQ Rear.
S2-CS Dual Stereo Line Channel.
S2-CSE Dual Stereo Line Channel.
193
Radio Broadcast Mixing Consoles - S2 Input Channels
There are separate logic remote output
controls for both of the two inputs providing
start/stop functions by fader, or ON button.
Remotes, etc, are triggered when the fader is
up and the channel ON button shows green.
The start function can be configured to be
either momentary (500mS) or latched for
each input. The remote outputs on the 9 way
D-type plug are fed from NPN opto-isolators.
Continuous momentary start can be enabled
such that when the fader is up and the
channel is on, each press of the ON button
triggers a momentary start.
clockwise shifts the signal to the left and
increases the signal by 6dB (right channel
reduces by 6dB); full clockwise shifts the
signal to the right and increases the signal
by 6dB (left channel reduces by 6dB).
The 100mm VCA fader provides unity gain
when fully open. The channel input signal is
routed to the outputs whenever the fader is
open, the ON button is selected and either
or both of the routing buttons are selected.
The S2-CSE channel is the dual stereo
line channel as above but fitted with
equalisation, providing 10dB cut and boost
at HF (6.5kHz) and LF (100Hz).
Two remote pins can also be configured for
general use as either inputs, outputs, latching, The EQ button places the equalisation in
or momentary and active high or low. Uses for and out of the signal path. The button is
this include channel live indication or specific illuminated in yellow when the EQ is active.
channel function remote control.
The channel can be used to control either
one of two separate timer displays.
The BAL/PAN control adjusts the stereo
balance in the following manner : Full anti-
S2-CSG Stereo Line & Gram Channel.
S2-CSGE Stereo Line & Gram Channel with EQ.
194
S2-CSG Stereo Line & Gram Channel &
S2-CSGE Stereo Line & Gram Channel with EQ
The S2-CSG channel is the dual stereo line
channel fitted with an RIAA input amplifier
on the second input. The balanced stereo
line input is on Neutrik XLR connectors
and the unbalanced RIAA stereo input on
phonos.
Selecting the LINE/GRAM button switches
the channel input between Stereo Line and
RIAA . The button is illuminated in red to
indicate when Gram (RIAA) is selected. The
operation of the button is inhibited when
the channel is “live”.
The channel is used for connecting a
turntable or record deck. The GRAM inputs
have RIAA equalisation suitable for magnetic
pick-up cartridges.
There are separate logic remote output
controls for both of the two inputs
providing start/stop functions by fader,
or ON button. Remotes, etc, are triggered
when the fader is up and the channel ON
S2-CSG Stereo Line & Gram and S2-CSGE Stereo Line & Gram with EQ Rear.
Radio Broadcast Mixing Consoles - S2 Input Channels
button shows green. The start function
can be configured to be either momentary
(500mS) or latched for each input. The
remote outputs on the 9 way D-type plug
are fed from NPN opto-isolators.
Continuous momentary start can be enabled
such that when the fader is up
and the channel is on, each press of the
ON button triggers a momentary start.
Two remote pins can also be configured
for general use as either inputs, outputs,
latching, or momentary and active high
or low. Uses for this include channel live
indication or specific channel function
remote control.
The BAL/PAN control adjusts the stereo
balance in the following manner : Full anticlockwise shifts the signal to the left and
increases the signal by 6dB (right channel
reduces by 6dB); full clockwise shifts the
signal to the right and increases the signal
by 6dB (left channel reduces by 6dB).
The 100mm VCA fader provides unity gain
when fully open. The channel input signal is
routed to the outputs whenever the fader is
open, the ON button is selected and either
or both of the routing buttons are selected.
The S2-CSGE channel is the stereo line
& gram channel as above but fitted with
equalisation, providing 10dB cut and boost
at HF (6.5kHz) and LF (100Hz). The EQ
button places the equalisation in and out of
the signal path. The button is illuminated in
yellow when the EQ is active.
S2-CDS Digital Dual Stereo Channel
The digital dual stereo channel has two 24bit 96kHz digital inputs with an INP 1/INP
2 button switch to select between them.
The button is illuminated in red to indicate
when Input 2 is selected. The operation of
the button is inhibited when the channel
is “live”.
Either input can be configured to be
balanced AES/EBU on standard XLR, or S/
PDIF on phono sockets, selected by internal
jumpers.
An additional jumper option can be used to
decode emphasis when indicated by certain
status bits in the incoming data stream.
The digital signals are converted to analogue
and from this point the channel functions
are the same as the S2-CS Dual Stereo Line
channel with the addition that when the ON
button flashes green, it means that there is
no valid digital input.
The channel can be used to control either
one of two separate timer displays.
S2-CDS Digital Dual Stereo Channel and S2-CDSE Digital Dual Stereo Channel
with EQ and Rear.Rear.
S2-CDS Digital Dual Stereo Channel.
S2-CDSE Digital Dual Stereo Channel with EQ.
195
Radio Broadcast Mixing Consoles - S2 Input Channels
S2-CDSE Digital Dual
Stereo Channel
with EQ
S2-CDAS Dual Stereo Line Channel &
S2-CDASE Dual Stereo Line Channel with EQ
This channel is as the digital dual stereo
channel but fitted with equalisation.
The S2-CDAS dual stereo line channel has
one 24-bit 96kHz AES/EBU digital input and
one balanced analogue stereo input, both
on Neutrik XLR connectors.
The HF and LF controls are used to adjust
the equalisation of the signal. The HF
control boosts and cuts the signal by 10dB
at 6.5kHz. The LF control boosts and cuts
the signal by 10dB at 100Hz. The EQ button
places the equalisation in and out of the
signal path and is illuminated yellow when
the EQ is active.
Both S2-CDS and S2-CDSE channels can be
used to control either one of two separate
timer displays.
The INP 1/INP 2 button selects which input
is routed through the channel. The button
is illuminated in red to indicate when Input
2 the balanced analogue source is selected.
The operation of the button is inhibited
when the channel is “live”.
The digital signals are converted to analogue
and from this point the channel functions
are the same as the S2-CS Dual Stereo Line
channel with the addition that when Input 1
is selected and the ON button flashes green,
it means that there is no valid digital input
signal being received.
There are separate logic remote output
controls for both of the two inputs providing
start/stop functions by fader, or ON button.
Remotes, etc, are triggered when the fader
is up and the channel ON button shows
green. The start function can be configured
to be either momentary (500mS) or latched
Input 1 is configured to be the AES/EBU
input on standard XLR. An additional jumper
option can be used to decode emphasis
when indicated by certain status bits in the
incoming data stream.
S2-CDAS Dual Stereo Line Channel and S2-CDASE Dual Stereo Line Channel with
EQ Rear.
S2-CDAS Dual Stereo Line Channel.
S2-CDASE Dual Stereo Line Channel with EQ.
196
Radio Broadcast Mixing Consoles - S2 Input Channels
S2-C6SS 6 Way Stereo Source Select
Channel with EQ
for each input. The remote outputs on the
9 way D-type plug are fed from NPN optoisolators.
Continuous momentary start can be enabled
such that when the fader is up and the
channel is on, each press of the ON button
triggers a momentary start.
Two remote pins can also be configured
for general use as either inputs, outputs,
latching, or momentary and active high
or low. Uses for this include channel live
indication or specific channel function
remote control.
The BAL/PAN control adjusts the stereo
balance in the following manner : Full anticlockwise shifts the signal to the left and
increases the signal by 6dB (right channel
reduces by 6dB); full clockwise shifts the
signal to the right and increases the signal
by 6dB (left channel reduces by 6dB).
The 100mm VCA fader provides unity gain
when fully open. The channel input signal is
routed to the outputs whenever the fader is
open, the ON button is selected and either
or both of the routing buttons are selected.
The S2-CDASE channel is the dual stereo
line channel as above but fitted with
equalisation, providing 10dB cut and boost
at HF (6.5kHz) and LF (100Hz).
The EQ button places the equalisation in
and out of the signal path. The button is
illuminated in yellow when the EQ is active.
The 6 way stereo line source select
channel has 6 balanced stereo inputs on
a 25 way D-type connector selectable by
a 6 way mechanical interlocking switch
bank, numbered 1 - 6. A depressed button
indicates the selected input.
The switch bank also controls 6 sets of
remote outputs allowing each source
to have a set of start/stop remotes. The
remote starts can be set as momentary or
latching. Continuous momentary start can
also be enabled such that when the fader is
up and the channel is on, each press of the
ON button triggers a momentary start.
The channel is fitted with EQ as standard
and other functions are similar to the S2CSE Dual Stereo channel with EQ.
The HF and LF controls are used to adjust
the equalisation of the signal. The HF
control boosts and cuts the signal by 10dB
at 6.5kHz. The LF control boosts and cuts
the signal by 10dB at 100Hz. The EQ button
places the equalisation in and out of the
signal path and is illuminated yellow when
the EQ is active. The channel can be used
to control either one of two separate timer
displays.
The channel can be used to control either
one of two separate timer displays.
S2-C6SS 6 Way Stereo Source Select Channel With EQ and Rear.
S2-C6SS 6 Way Stereo Source Select Channel With EQ.
197
Radio Broadcast Mixing Consoles - S2 Input Channels
S2-CSMM Stereo Mix
Minus Channel
The stereo mix-minus channel has one
balanced stereo line level input and a
balanced stereo mix-minus output, on
XLR connectors. The channel is intended
for use where a remote stereo source,
such as another studio connected via
ISDN, requires a stereo cleanfeed return.
The cleanfeed is generated by the mixminus method from either the PGM or
AUD outputs, depending on the routing
selection.
Selecting the CUE/PFL & T/B button routes
the pre-fader input signal to the monitoring
system where the signal can be heard via
headphones and/or loudspeakers.
In addition, talkback is routed to
the cleanfeed output. This enables
the presenter/technical operator to
communicate with the remote source prior
to going live to air. PFL can be automatically
disabled when the fader
is opened.
The outputs on this channel can be
configured by jumper settings as a mono
sum of mix-minus on the left channel
and continuous talkback on the right
channel (for some ISDN codecs and hybrid
applications).
The fader, or ON button, control a logic
remote output function. The remote
function can be configured to be either
momentary or latched. The remote outputs
on the 9 way D-type plug are fed from NPN
opto-isolators.
Continuous momentary start can be enabled
such that when the fader is up and the
channel is on, each press of the ON button
triggers a momentary start.
Two remote pins can also be configured
for general use as either inputs, outputs,
latching, or momentary and active high
or low. Uses for this include channel live
indication or specific channel function
remote control.
Selecting the PGM and/or AUD buttons
routes the channel audio output to the PGM
and/or AUD mix buses. Selecting the PGM
button also routes the PGM main output
via the mix minus system to the cleanfeed
output. Selecting the AUD button routes the
AUD main output to the mix minus system.
The mix minus is derived from the PGM
output when selecting both PGM and AUD.
Equalisation is fitted as standard on this
channel and operates in the same was as on
the S2-CSE channel.
The channel can be used to control either
one of two separate timer displays.
S2-CSMM Stereo Mix Minus Channel Rear.
S2-CSMM Stereo Mix Minus Channel.
198
Radio Broadcast Mixing Consoles - S2 Input Channels
Routing
Left
Trim
Input 1
Line Left
Right
Trim
Input 1
Line Right
EQ
LINE 1/2
VCA
BAL
1
3
2
B
AES/SPDIF
2
1
3
2
AES/SPDIF Input
Select
1
Aud
Inp 1
inp 2
EQ
C / Feed
Left
Null
C / Feed
Right
Null
Cue
PFL
On
From PGM
Output L
From PGM
Output R
D/A
3
Input 1
AES/EBU
Input 2
AES/EBU
Input 1 A
SPDIF B
Input 2
Input 1 remotes
Input 2 remotes
The telco input channel controls the
connection to a telephone balance unit (or
hybrid). It has one balanced line level mono
input on Neutrik XLR, and one balanced
line level cleanfeed output to return to the
hybrid.
CPU
PGM
A
PFL
Fader
Right
Trim
Input 2
Line Right
To S2-OMC &
S2-OMS CPU
VCA
Left
Trim
Input 2
Line Left
Input 2
RIAA
Option
TRIM
HF/LF
EQ
PGM AUD PFL C/FEED T/B
L R L R L R 1 2 1 2
Mutes
S2-CT Telco Channel
1
3
2
Digital Input Version
Stereo Mix-Minus Version
6 x Stereo
Bal Inputs
6 way Input
And Remote
Output Selector
6 x Remote
Start/Stop
Stereo Selector Version
Left Input Amp
Right Input Amp
Selecting the CUE/PFL & T/B button routes
the pre-fader input signal to the monitoring
system where the signal can be heard
via headphones and/or loudspeakers. In
addition, talkback is automatically routed
back to the caller via the cleanfeed system.
This enables the presenter, or technical
operator, to communicate with the caller
prior to putting the caller live to air. PFL can
be automatically disabled when the fader
is opened.
There is a logic remote output to place the
hybrid “on hold” by using the LINE HOLD
button. The remote output on the 9 way
D-type plug is fed from an NPN opto-isolator
and it can be made momentary or latching.
Selecting the PGM and/or AUD buttons
routes the channel audio output to the
PGM and/or AUD mix buses. The buttons
are illuminated in green to indicate the
routing status. Changing the status of the
PGM button is inhibited when the channel
is “live”.
S2-CSG Stereo Line With Gram (S2-CSGE with EQ), S2-CS Dual Stereo (S2-CSE with EQ), S2-CDS Digital Dual
Stereo (S2-CDSE with EQ), S2-CSMM Stereo Mix Minus and S2-C6SS Stereo Source Select.
S2-CT Telco Input Channel Rear.
S2-CT Telco Input Channel.
199
Selecting the PGM button also routes the
channel to one of the two true cleanfeed
buses (jumper selectable). When selecting
only the AUD button, for example to
record off-line telephone conversations,
a cleanfeed is generated via the channel
mix-minus system. When both PGM and
AUD are selected the channel operates in
the true cleanfeed mode, sending the PGM
signal to the cleanfeed bus.
Equalisation is fitted as standard and is
enabled by the EQ button, providing 10dB
cut and boost at HF (6.5kHz) and LF (100Hz).
The EQ button places the equalisation in
and out of the signal path and is illuminated
yellow when the EQ is active.
Routing
Input
Trim
Input
This blank channel is used to fill in areas of
the mixer where you don’t require an input
or output channel.
S2-CB Blank Channel.
200
HF/LF
EQ
EQ
PFL
PAN
PGM AUD PFL C/FEED T/B
L R L R L R 1 2 1 2
PFLTo
Meter Select
To Talkback
Output
VCA
C/Feed
Output
Hybrid Latch
Fade R
CPU
PGM
Aud
S2-CT Telco Input Channel.
S2-CB Blank Channel
VCA
Trim
Mutes
Radio Broadcast Mixing Consoles - S2 Input Channels
EQ
Cue
T/B
ON
Cleanfeed
Selection
And routing
Null
L From AUD
R Output
Mix Bus
Radio Broadcast Mixing Consoles - S2 Output Channels
S2 Output Channels
There are six different output channels
available, for controlling monitoring in both
a Control Room and separate Studio, with
the two main output channels also having a
master fader option.
The master output channels provide
balanced analogue audio outputs as well
as simultaneous AES/EBU or S/PDIF digital
audio outputs for both PRG and AUD buses.
The analogue mono output can be selected
from PRG or AUD and meter selection is
available to show either the PRG bus, AUD
S2-OMC Control Room Monitor Channel
bus, or to follow the Control Room Monitor
selection.
Both Control Room and Studio Monitor
channels are available for controlling what’s
routed to the presenter’s and guest’s
headphones and monitor speakers.
2 external inputs can also be monitored.
A minimum of 3 channels need to be fitted
into a mixer: channels S2-OMC, S2-ODP (or
S2-ODPF) and S2-ODA (or S2-ODAF) must be
fitted and S2-OMS is optional.
The control room monitor channel is common to every mixer and is used for the
monitoring of various sources on the control room loudspeakers (MONITOR) on a 6.35mm
stereo jack socket and presenter’s headphones (PHONES) on 6.35mm stereo jack sockets.
Front panel control knobs are used to vary the monitor and headphone levels and
headphones can be plugged in on the front or the rear of this channel. The control room
monitor loudspeaker and headphone levels are variable between 0 (cut off) and 10 (max).
A four way electronically interlocking switch
bank selects the source routed to the
speakers and headphones from either of
two external inputs (EXT 1, EXT 2), the PGM,
or AUD output. The buttons illuminate to
show the selected source, PGM and AUD in
green, EXT 1 and EXT 2 in red. The external
inputs can be used for monitoring an off air
signal or another studio output.
An illuminated Mute LED shows when a live
microphone channel in the control room has
muted the speakers, to prevent feedback.
Mute outputs on the remote connector of
the PGM output channel can be used to
remotely illuminate “Mic Live” lights.
AUTO CUE/PFL buttons adjacent to each
level control allow the monitoring of PFL
when an input channel has been selected
to CUE/PFL. In addition, SPLIT CUE/PFL can
be selected to the headphones, which will
place the selected source in mono in one
ear and PFL in mono in the other.
This channel controls the signal conditioning
for the talkback input and output and routes
these signals to and from the talkback bus.
S2-OMC Control Room Monitor Rear.
S2-OMC Control Room Monitor Channel.
201
Radio Broadcast Mixing Consoles - S2 Output Channels
S2-OMS Studio
Monitor Channel
The studio monitor channel is optional
and is used for the monitoring of various
sources on the studio loudspeakers
(MONITOR) on a 6.35mm stereo jack socket
and studio headphones (PHONES) on a
6.35mm stereo jack socket.
A front panel control is used to vary
the monitor level. The studio monitor
loudspeaker level can be variable between
0 (cut off) and 10 (max). The output will
need to be fed to a suitable power amplifier
to drive the loudspeakers. Headphones
A four way electronically interlocking
switch bank selects the source routed
to the speakers and headphones from
either of two external inputs (EXT 1, EXT
2), the PGM, or AUD output. The buttons
illuminate to show the selected source,
PGM and AUD in green, EXT 1 and EXT
2 in red. External inputs can be used for
monitoring an off air signal or another
studio output.
An illuminated Mute LED shows when a
live microphone channel in the studio has
muted the speakers. Mute outputs on
the remote connector of the AUD output
channel can be used to remotely illuminate
“Mic Live” lights.
A separate Studio T/B button is provided to
allow the presenter/engineer to talk to the
studio monitors and/or headphones and
internal jumpers allow this talkback to
replace or dim the selected source.
PGM Aud
O/P O/P
L R L R
Rev T/B
A Global Talkback system can be configured
to allow every contribution point in the
control room and studio to communicate
with each other via their microphone
and headphones.
ǁŝƚŚŝŵƉĞĚĂŶĐĞƐŽĨϯϱɏĂŶĚĂďŽǀĞ
can be driven directly from the channel
headphone connector on the rear panel.
Timers
Internal jumpers can be configured to allow
talkback to replace or dim the selected
source, dimming independently on monitor
speakers or headphones.
PFL T/B
O/P O/P
L R 1 2
A Global Talkback system can be configured
to allow every contribution point in the
control room and studio to communicate
with each other via their microphone and
headphones.
PGM
Aud
EXT 1
To S2-CT
Telco
Cleanfeed
Selection
L&R
EXT 2
Auto
Cue
Auto
Cue
Studio
T/B
From Mutes
Bus
Split
Cue
Studio C/Room
202
Cue/PFL Mute/
Dim Talkback
Global T/B
Level
Pres/Studio
Headphones
Rev T/B
Talkback
Output
To Monitor Meters
S2-OMS Studio Monitor Channel and S2-OMC Control Room Monitor Channel.
Green illuminated AUTO CUE/PFL buttons
adjacent to each level control allow the
monitoring of PFL when an input channel
has been selected to CUE/PFL, either to
the monitors or headphones.
S2-OMS Studio Monitor Rear.
S2-OMS Studio Monitor Channel.
Level
CPU
External
Inputs
C/room/
Studio
Monitors
Radio Broadcast Mixing Consoles - S2 Output Channels
S2-ODP Digital PGM Output Channel and S2-ODPF Digital PGM Output with
Master Fader.
The digital output status bits (consumer or
professional), output sample size (16, 20
or 24 bits) and sample rate (automatically
or by internal jumpers from 32 kHz to 96
kHz) can be set. The digital output can be
synchronised from an external input, or
The S2-ODPF channel also has a 100mm
VCA output fader for production use which
provides unity gain when fully open.
Fader Option
From Left
PGM Bus
Analogue PGM
(AUD) Output Left
From Right
PGM Bus
Analogue PGM
(AUD) Output Right
From CPU
Aud
CR
S2-ODP Only MON
From
Mutes Bus
Meter
Select
Analogue Mono
Output
AES/SPDIF
From L&R
PFL Bus
2
CPU
Digital
Audio
Interface
1
Output
Select
3
A/D
PGM
1
3
2
A
B
Meterbridge
3
The illuminated LOCK LED shows that the
digital output is locked to the onboard
master clock, incoming word clock or AES/
EBU or S/PDIF sync signal. The channel
automatically locks to a valid sync clock,
flashing if sync is lost. Synchronisation can
be jumper disabled.
PFL T/B
O/P O/P
L R 1 2
2
from an on-board master clock. There are
4 different sync modes : Master, Auto, Auto
Lock and Slave.
PGM Aud
O/P O/P
L R L R
1
The main power input from the S2-PSU
is on this channel. Button switches at the
top of the channel select the mono output
source from either PGM or AUD. The Mono
compatible output could be used to feed
a mono transmitter or any station output
monitoring system that requires a mono
signal, such as a background music system.
Timers
This channel is common to every mixer and provides a digital and analogue output from
the PGM mix bus. The digital output is available as a balanced AES/EBU signal via a
standard XLR or as S/PDIF on phono sockets. The balanced analogue PGM output and
mono output is on a 15 way D-type connector. This also carries the two sets of latching
relay contacts for the control room mutes.
REV T/B
S2-ODP Digital PGM Output Channel &
S2-ODPF Digital PGM Output Channel with Master Fader
AES/EBU PGM
(Aud) Output
SPDIF PGM
(Aud) Output
SPDIF Sync
AES/EBU Sync
AES/SPDIF
Word Clock
From
Talkback
Bus
S2-ODP Digital PGM Output Channel (S2-ODPF with Master Fader) and S2-ODA Digital AUD Output
Channel (S2-ODAF with Master Fader).
S2-ODP Digital PGM Output Channel.
S2-ODPF Digital PGM Output Channel with Master Fader.
203
Radio Broadcast Mixing Consoles - S2 Output Channels
S2-ODA Digital AUD Output Channel &
S2-ODAF Digital AUD Output Channel with Master Fader
This channel is also common to every
mixer and provides a digital and analogue
output from the AUD mix bus. The digital
output is available as a balanced AES/EBU
signal via a standard XLR or as S/PDIF on
phono sockets. The balanced analogue AUD
outputs are available on a 15 way D-type
connector which also carries the two sets
of latching relay contacts for the studio
mutes.
Button switches at the top of the channel
select the monitoring source to the
meterbridge. Meters in the meterbridge
which are connected to the “Monitor
Meter” position will display the selected
signal from the electronically interlocking
PGM, AUD, or CR MON buttons. The CR
MON signal source is whatever is being
monitored on the presenter’s headphone
on the control room monitor channel, pre
talkback and level control. This signal could
be PGM, AUD, EXT 1, EXT 2, or PFL, as
selected by the presenter.
The digital output status bits (consumer or
professional), output sample size (16, 20
or 24 bits) and sample rate (automatically
or by internal jumpers from 32 kHz to 96
kHz) can be set. The digital output can be
synchronised from an external input, or
from an on-board master clock. There are
4 different sync modes : Master, Auto, Auto
Lock and Slave.
The illuminated LOCK LED shows that the
digital output is locked to the onboard
master clock, incoming word clock or AES/
EBU or S/PDIF compatible sync signal. The
S2-ODA Digital AUD Output Channel.
S2-ODAF Digital AUD Output Channel with Master Fader.
204
channel automatically searches for a sync
signal on the Word Clock, or the selected
digital input, and automatically locks to
a valid sync clock. If sync is lost then the
indicator will flash. Synchronisation can be
jumper disabled.
The S2-ODAF channel also has a 100mm
VCA output fader for production use which
provides unity gain when fully open.
S2-PG Penny & Giles
Conductive Plastic
Fader
The input and output channels of an S2
mixer can optionally be fitted with high
quality Penny & Giles faders.
S2-ODA Digital AUD Output Channel and S2-ODAF Digital AUD Output Channel
with Master Fader Rear.
Radio Broadcast Mixing Consoles - S2 Meterbridge Modules
S2 Meterbridge Modules
PGM Aud
O/P O/P
L R L R
Rev T/B
Timers
In the meterbridge area you can choose
from four styles of metering, a phase meter,
a dual timer, a PFL/Talkback loudspeaker,
2 talkback modules, 3 switch panels and a
range of blanking plates. S2’s meterbridge
modules are freely assignable so that you can
position them exactly where you want them.
The dual meter panels are used for
monitoring console signal levels. Up to
three different meter panels can be housed
in the meterbridge. The meters can be
configured internally to be fed from one of
three signal sources, the selected source
indicated by a LED:
Select
The output of the meter switch
on the AUD output module
which can show PGM, AUD,
or CR MON, which is the
selected monitor source on the
control room monitor module,
(EXT 1, EXT 2, PGM, or AUD)
interrupted by PFL.
PFL T/B
O/P O/P
L R 1 2
From Monitor Output
S2-ML
REV T/B
S2-MT
Dual Timer, Clock And Date Display
CUE/PFL
Talkback
Loudspeaker
CUE/PFL REV T/B
St ar t
St op
Reset
Mode
St ar t
S t op
Reset
Mode
Program PGM output which can
optionally be interrupted
by PFL.
PFL
The output of the PFL bus.
S2 Without
Meterbridge
The S2 is available without a meterbridge.
This does affect some of the features
available on the mixer. Contact [email protected]
sonifex.co.uk for further information and
pricing.
In the meterbridge area you can choose
from four styles of metering, a phase meter,
a dual timer, a PFL/Talkback loudspeaker, 2
talkback modules and a range of blanking
plates. S2’s meterbridge modules are freely
assignable so that you can position them
exactly where you want them.
To cleanfeed
Left (S2-CSMM)
PGM Meters
Monitor Meters
To Cleanfeed
Right (S2-CSMM)
PFL Meters
Phase Meters
S2-MPPM Dual PPM Meter Panel
S2-MPPT True Dual PPM Meter Panel
S2-MVU Dual VU Meter Panel
S2-ML53 Dual LED Bar Graph Meter
S2-MPH PHase Meter Panel
S2-MPPM/T Meterbridge PPM Panel, S2-MVU Meterbridge VU Meter Panel, S2-ML53 Meterbridge LED Meter Panel, S2-MPH Meterbridge Phase Meter Panel, S2MTBS Station Master and S2-MTB6 6 Way Talkback Panels, S2-MT Meterbridge Dual Timer Panel and S2-ML Meterbridge Loudspeaker Monitor and Talkback Panel.
205
Radio Broadcast Mixing Consoles - S2 Meterbridge Modules
S2-MPPM Meterbridge
PPM Meter Panel
The S2-MPPM panel has two fast-reading
moving-coil meters with PPM-style scale.
S2-MVU Meterbridge
VU Meter Panel
The PPMs have a 1-7 scale and are configured
such that a 1kHz signal, at 0dB at the PGM
output, will indicate a meter reading of 4.
Each mark on the PPM scale indicates a 4dB
The S2-MVU two moving coil VU meters are
change in signal level.
S2-ML53 Meterbridge
LED Meter Panel
S2-MPH Meterbridge
Phase Meter Panel
The LED meter consists of 53 LEDs showing
both VU and peak representation. Both PPM
and VU scales are shown, one on each side
of the LED bar-graph.
The S2-MPH is a phase meter which
operates on the selected source to the
meters. 5 LED indicators show the phase
angle in 45 degree steps from 0 (in-phase) to
180 (out of phase).
A separate phase meter can be linked to
each meter in the meterbridge.
206
configured such that a 1kHz signal, at +4dBu
at the PGM output, will indicate a meter
reading of 0VU.
Radio Broadcast Mixing Consoles - S2 Meterbridge Modules
S2-MT Meterbridge
Dual Timer Panel
Each of the dual timers can be used for
timing events triggered by Timer 1, Timer 2,
or the Mic fader open signal. In addition the
upper timer is used for displaying the time
of day and the lower timer for displaying
the date. The illumination pattern of the red
LEDs indicates the timer display mode.
The date and time can be synchronised to
a very accurate internal clock or optionally
from an external source, MSF, DCF, SMPTE,
or RS232 time code string.
S2-ML Meterbridge
Loudspeaker Monitor
Panel
The S2-ML speaker is used for monitoring
PFL and reverse talkback in the control room
directly from the mixer. Two control knobs
are provided for adjusting the levels of the
PFL and Talkback signals from cut-off at 0
to unity gain at 10. The speaker is muted
automatically when the control room mute
is active, to prevent feedback.
S2-MSB1 S2
Meterbridge
Switch Panel
With 1 Button
S2-M6SS Meterbridge
6 Way Source Select
Panel
This is a meterbridge panel,
1 channel wide, with 1
button which can be used
for bespoke control of
the S2, or for your own
purposes, e.g. equipment
control.
Contact Sonifex with your particular mixer
control requirements.
S2-MSB3 S2
Meterbridge Switch
Panel With 3 Buttons
This is a meterbridge panel,
1 channel wide, with 3
buttons which can be
used for bespoke control
of the S2, or for your own
purposes, e.g. equipment
control, or for transmission
control linked to an RB-OA3
on-air switcher.
Contact Sonifex with your
particular mixer control
requirements.
The meterbridge source selector panel
produces a stereo analogue audio output
from 6 selectable stereo analogue sources,
which can be connected to a stereo input
channel within the mixer, or the EXT 1
and EXT 2 monitor inputs, to provide up to
12 external monitor inputs. There are
6 illuminated push buttons, which select and
indicate the current channel selection. The 6
stereo sources are connected on the rear of
the panel via a connector, which uses exactly
the same pin configuration as the stereo
audio source input connector on the
S2-C6SS channel.
207
Radio Broadcast Mixing Consoles - S2 Meterbridge Modules
S2-MTBS Meterbridge S2-MTB6 Meterbridge Alternative Button
6 Way Talkback Panel Text
Station Master
For meterbridge panels, the buttons can be
Talkback Panel
supplied with different printed script. A few
This is used for interfacing with a Sonifex
Station Master studio switcher and talkback
router, or the RB-OA3 on-air switcher, and
has buttons for talkback, Offer and Accept
switching and profanity delay control.
examples are shown here, but for a full list,
please go to the Sonifex website:
www.sonifex.co.uk/s2/s2-buttons.shtml
This is used for communicating with up to
6 other locations, studios or mixers, and
with the Sonifex TB-6D and TB-6R talkback
intercoms. While a button is pressed,
the switch is illuminated and the talkback
is active. The buttons work with both a
momentary and latched operation - if held
down, the selection is cancelled on release.
Otherwise the button is alternate action.
208
S2-MB1-5 Meterbridge
Blanking Plates 1-5
Meterbridge blanking plates are available
in channel widths from 1 to 5 inclusive and
are used to fill areas in the meterbridge not
occupied by active modules.
Radio Broadcast Mixing Consoles - S2 Chassis
Chassis Sizes
Model No.
S2 Chassis Sizes
The S2 chassis is available in 5 channel width sections, allowing 5, 10, 15, 20, 25 and 30
channel width mixers. This means S2 can be used for small newsrooms or large on-air
situations.
Number of Channels
S2-05
5
S2-10
10
S2-15
15
S2-20
20
S2-25
25
S2-30
30
S2-05 Chassis
S2-15 Chassis
S2-25 Chassis
Chassis with 5 channel width sections.
Chassis with 15 channel width sections.
Chassis with 25 channel width sections.
S2-10 Chassis
S2-20 Chassis
S2-30 Chassis
Chassis with 10 channel width sections.
Chassis with 20 channel width sections.
Chassis with 30 channel width sections.
209
Radio Broadcast Mixing Consoles - S2
S2-7SS Script Space
S2-10SS Script Space
The script space occupies an area of 7
channel widths, and is designed to hold
documents or scripts for the mixer operator
when desk space is at a premium.
The S2-7SS Script Space is 266mm wide x
390mm deep.
This script space is 10 channels wide and
is large enough for a small keyboard.
The S2-10SS Script Space is 380mm wide x
390mm deep.
Technical Specification For S2
Input/Output Impedances
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Mixer Width
Type
(cm)
Width
(inches)
Depth
(cm)
Depth
(inches)
Height
(cm)
Height
(inches)
S2-30 137cm
53.9”
70cm
27.6”
45cm
17.7”
36kg
Gross
Weight
(lbs)
Nett
Weight
(kg)
Nett
Weight
(lbs)
79lbs
32kg
70lbs
S2-25 117cm
46.0”
70cm
27.6”
45cm
17.7”
35kg
77lbs
31kg
68lbs
S2-20 100cm
39.4”
70cm
27.6”
45cm
17.7”
33kg
73lbs
29kg
64lbs
S2-15
80cm
31.5”
70cm
27.6”
45cm
17.7”
27kg
59lbs
24kg
53lbs
S2-10
60cm
23.6”
70cm
27.6”
45cm
17.7”
25kg
55lbs
22kg
48lbs
S2-05
60cm
23.6”
70cm
27.6”
45cm
17.7”
22kg
48lbs
20kg
44lbs
210
Stereo Inputs (unity -86dB
gain, one channel routed):
Stereo Inputs (unity -83dB
gain, two channels routed):
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Distortion
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Total Harmonic
Distortion:
0.015% at 1kHz, 0dB
0.025% at 10kHz, 0dB
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Crosstalk
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Inter-channel:
< -90dBu
Input/Output Gain Range
Stereo:
-90dBu at 1kHz
Mic Input:
Preset pot +13dB to +66dB ref
-50dBu, TRIM pot ±12dB
Equalisation
Mono Line Input:
Preset pot -6dB to +10dB ref
0dBu, TRIM pot ±12dB
HF Shelving at 6.5kHz: ±7dB
Stereo Line Input:
Preset pot -6dB to +10dB ref
0dBu, TRIM pot ±12dB
Pan Range:
Telco Input:
Preset pot -6dB to +10dB ref
0dBu, TRIM pot ±12dB
Pot position: Fully clockwise,
centre, fully anti-clockwise;
Left: Off, 0dB, +3dB;
Right: +3dB, 0dB, Off
respectively.
±6dB
LF Shelving at 100Hz: ±7dB
Range
Telco Output:
Preset pot -6dB to +4dB ref 0dBu
Mix Minus Input:
Preset pot -6dB to +10dB ref
0dBu, TRIM pot ±12dB
Balance Range:
Mix Minus Output:
Preset pot -3dB to +3dB ref 0dBu
Mic Input:
Digital Input:
0dBFS = +12dBu on input;
TRIM pot ±12dB allowing 0dBu
to +24dBu
Digital I/O
Digital Output:
0dBFS = +18dBu
Mic Input:
Gross
Weight
(kg)
-89dB
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Frequency Response
S2 Mixer Weights and Boxed Dimensions
Stereo Inputs (unity
gain, no routing):
40Hz to 20kHz, -1dB,+0dB
(-3dB at 130Hz with HPF in)
Line Inputs:
20Hz to 20kHz, - 0.5dB,+0dB
RIAA Input:
30Hz to 16kHz ±1.5dB RIAA
equalised
Noise (20Hz to 20kHz)
Common Mode Rejection Ratio
Sync Input Sample
Rate:
30kHz - 100kHz
Output Sample Rates 32kHz, 44.1kHz, 48kHz, & 96kHz
(Using Onboard Clock):
(Using Sync Input):
30kHz - 100kHz
Output Sample Width: 16, 20, 24 bit
Output
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Output Load:
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recommended
+26dBu balanced
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Maximum Output
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Stereo Inputs (fader
down, no routing):
-89dB ref 0dB
Power
Stereo Inputs (fader -89dB
down, one channel routed):
> 100dB at 70dB gain
Power (S2-PSU):
Filtered IEC, switchable 115V,
230V, fused, 210W max.
Radio Broadcast Mixing Consoles - S2
S2 Mixer Placement In Furniture
276mm
40mm
Woodwork
Rear Edge
Solutions Single Mixer Furniture Package Showing
an S2-25 Mixer.
180mm
506mm
10mm
7°
196mm S2-5
387mm S2-10
117mm
23mm
577mm S2-15
768mm S2-20
958mm S2-25
1149mm S2-30
S2 Placement in Cut-Out Viewed from Above.
Woodwork
Front Edge
Mounting Holes 5mm Diameter
505mm
521mm 565mm
S2 Mixer Viewed from Side Profile.
211
Radio Broadcast Mixing Consoles - S2 Power Supply
S2-PSU Power Supply
S2-PSU Front Panel.
The S2-PSU is a 1U high rack-mount unit,
which supplies power to the entire range
of S2 mixers. A lead with a special 9 pin
D-Type socket on each end is used to
connect the power supply to the S2-ODP
PGM output channel on the mixer.
LEDs on the front of the unit indicate
whether the power to the voltage rails is
being supplied correctly.
Dimensions (S2-PSU)
(Raw):
48cm (W) x 24.3cm (D) x 4.4cm (H)
19” (W) x 10” (D) x 3.5” (H)
(Boxed):
54cm (W) x 41cm (D) x 16cm (H)
21.3” (W) x 16.1” (D) x 6.3” (H)
Weight (S2-PSU)
Nett: 6kg
Gross: 7.8kg
Nett: 13.2lbs Gross: 17lbs
212
S2-PSU Rear Panel.
S2-BI Bus Interlink
Cable
This is used to connect two S2 chassis so
that they can function as one mixer.
Note that a maximum total of 30 channels is
allowed in a split mixer configuration.
Radio Broadcast Mixing Consoles - S2 Power Supply
S2-PSUS Dual Power Supply Switcher
S2-PSUS Front Panel.
The S2-PSUS is a passive power switcher
which takes the power feeds from
2 x S2-PSU units and switches between
them in the event of failure. Two trailing
leads 0.5m in length connect to 2 x S2-PSU
units. With LED failure indicators and GPI
alarms, the S2-PSUS is the perfect dual
redundant power supply module.
S2-PSUS Rear Panel.
Dimensions (S2-PSUS)
(Raw):
48cm (W) x 23cm (D) x 4.4cm (H)
19” (W) x 9” (D) x 1.7” (H)
(Boxed):
55cm (W) x 39.3cm (D) x 8.5cm (H)
21.6” (W) x 15.5” (D) x 3.4” (H)
Weight (S2-PSUS)
Nett: 1.58kg Gross: 2kg
Nett: 3.5lbs Gross: 4.4lbs
213
Sonifex S2 Solutions Furniture.
Photo courtesy of Huddersfield Hospital Radio.
214
Radio Studio Packages
Sonifex has been manufacturing and distributing complete studio packages for over 14 years
and they’ve been a remarkable success with installations in more than 100 studios since
initiation.
The S2 Solutions studio package consists of the S0, S1 or S2 mixer mounted in modular
ash technical furniture and surrounded by leading brands of high quality broadcasting
equipment, carefully integrated to provide an easy to operate professional studio.
Radio Studio Packages
216
S2 Solutions Single and Split Mixer Furniture
218
Portable Radio Studio Package
220
Monitor Speaker Stands
221
Sonifex S2 Solutions Furniture.
Photo courtesy of Dixie Media.
215
Radio Studio Packages
Radio Studio Packages
Category: Radio Studio Furniture.
Product Function: Modular radio studio
furniture.
Typical Applications: Simple furniture
solutions for college, hospital, community
and commercial radio stations.
Features: Well made quality woodwork,
tried and tested design, options for rack
sizes, pre-wired cabling, easy assembly.
Sonifex has been manufacturing and distributing complete studio packages since 1998 and
they’ve been a remarkable success with installations in over 100 studios since initiation.
The S2 Solutions studio package consists of the S2 digital I/O analogue mixer mounted
in modular ash technical furniture and surrounded by leading brands of high quality
broadcasting equipment carefully integrated to provide an easy to operate professional
studio.
With years of experience of manufacturing
and distributing radio studio equipment,
Sonifex is perfectly positioned to advise on
the proper equipment required to get your
radio station up and running, on time and
to budget.
Good-looking Furniture
Oozing Professionalism
One of the main reasons why people
keep coming back to use our S2 Solutions
packages is because of the high quality of
the technical furniture, in deep solid ash
with high quality, hard wearing blue/grey
linoleum surfaces. The surfaces are not only
robust but also have good acoustic qualities
- items dropped on the surface produce a
216
dull thud to which the microphone is not
sensitive.
Appearances are important, even on radio!
As your customers and visitors walk into
your studios, they are going to feel that
they’ve walked into a professional,
smooth-running, radio station.
The furniture is modular, so it can be
used for a small L-shaped production or
talks studio, through to a larger U-shaped
on-air studio. It’s easy to fit together with
large internal spaces for cable routing,
so from every angle, the furniture looks
neat, orderly and professional. If you
need additions to the woodwork, or
modifications, we can produce woodwork
to your design which can integrate with our
standard furniture to save costs.
Ease of Assembly & Ease of Use
The furniture has been designed to be easily
shipped and then quickly assembled by
you on-site. The woodwork is packaged in
protective cartons and simple instructions
allow you to assemble the furniture rapidly
and with a minimum of tools.
The SOL-B12 12U support base pod has
adjustable feet and provides a solid platform
for the SOL-SB1000, or SOL-SB1600,
side-return. The side return can also be
supported by SOL-LGG80 grey legs which
have adjustable feet ensuring that the worktops are steady and level.
The studio furniture is ergonomically
SOL-B12 12U Support Base Pod.
designed to house all of the technical
equipment and to place frequently used
items within easy reach of the presenter.
There is a choice of top-pods to house the
equipment SOL-D8 - This is an 8U high ash top-pod
which can be placed virtually anywhere (*)
on the work-surface to suit the presenter.
It usually contains 2 x CD players and 2 x
compact flash recorders, but can house
any equipment.
SOL-D12 - This 12U high ash top-pod allows
more space for further equipment that
needs to be close to the presenter.
SOL-T12 - This consists of 2 x 6U racks
mounted on legs above the work-surface, so
that you can use the space underneath for
CD storage, a laptop or for paperwork.
Radio Studio Packages
S2 Solutions Technical Specification
SOL-D8 8U Top Pod.
SOL-D12 12U Top Pod.
Pre-Wired Cabling
The packages come complete with prewired
cabling, each cable connecting directly
from the equipment to the mixer, so that
your studio can be up and running in only
a few hours. Cabling includes the power
distribution, remote control and audio
connections to and from the mixer. All the
cables can be hidden from view allowing
a simple, tidy installation. SOL-CVRC cable
covers can be fitted wherever you need
cable access through the desktop surface,
for example, for microphone leads, or
PC monitor cables. For more complex
installations, the cabling can be wired
to Krone blocks for ease of future
adjustment and maintenance.
* Note that whilst the SOL-D8 and
SOL-D12 are movable on the desktop,
their position needs to be relatively
fixed in order to cover the cable hole
underneath the units which routes
cables from the top-pod to the mixer.
SOL-LGG80
Brushed
Stainless
Leg.
SOL-T12 Straight 2 x 6U Top Pod on Legs.
SOL-CVRC Cable Cover.
For a detailed quote on the S2 Solutions furniture and radio studio packages, contact
Sonifex outlining your requirements. It’s useful to know the type and quantity of studios
that you need, the kind of equipment that you’re looking for and your budget. We can
then tailor a solution to meet your needs.
Specification For Solutions
Equipment Type
Description
Standard Qty Single
Mixer Furniture
Standard Qty Split
Mixer Furniture
1
SOL-D8
S2 Solutions 8U Desk Top 19” Rack
1
SOL-D12
S2 Solutions 12U Desk Top 19” Rack
0
0
SOL-T12
S2 Solutions 12U Top 19” Rack Straight Double On Legs
0
0
SOL-B12
S2 Solutions 12U Support Base 19” Rack
1
1
SOL-B12P
S2 Solutions Portable 12U Support Base With Small Desk Top
0
0
SOL-SB1000
S2 Solutions 1000mm (W) x 600mm (D) Side Return Desk Top
39” (W) x 24” (D)
0
2
SOL-SB1600
S2 Solutions 1600mm (W) x 600mm (D) Side Return Desk Top
63” (W) x 24” (D)
1
0
SOL-MB1300
S2 Solutions 1300mm (W) x 900mm (D) Mixer Desk Top
51” (W) x 35” (D)
0
0
SOL-MB1500
S2 Solutions 1500mm (W) x 900mm (D) Mixer Desk Top
59” (W) x 35” (D)
1
0
SOL-MBC
S2 Solutions Split Desk Centre Mixer Desk Top
0
1
SOL-MBCT
S2 Solutions Split Desk Centre Talks Table Mixer Desk
Top With Leg And Cable Hole Cover
0
0
SOL-MBL
S2 Solutions Split Desk Mixer Left Hand Corner Desk Top
0
1
SOL-MBR
S2 Solutions Split Desk Mixer Right Hand Corner Desk Top
0
1
SOL-LGG80
S2 Solutions Table Legs (Pair) Brushed Stainless 80mm (3”) Dia
2
4
SOL-CVR60
S2 Solutions 600mm (24”) Desk End Cover Panel
0
0
SOL-CVR80
S2 Solutions 810mm (32”) Desk End Cover Panel
2
0
SOL-CVRC
S2 Solutions Cable Hole Covers 80mm (3”) Inc Fitting
2
3
SOL-MS11
S2 Solutions Monitor Speaker Stands, 11cm High, Black, Pair
SOL-MS21
S2 Solutions Monitor Speaker Stands, 21cm (8”) High, Black, Pair
0
0
SOL-MS32
S2 Solutions Monitor Speaker Stands, 32cm (13”) High, Black, Pair
0
0
SOL-MS40
S2 Solutions Monitor Speaker Stands, 40cm (16”) High, Black, Pair
0
0
SOL-MS53
S2 Solutions Monitor Speaker Stands, 53cm (21”) High, Black, Pair
0
0
SOL-MST
S2 Solutions Monitor Speaker Stands, Tannoy 20cm High, Black, Pair
SOL-TK
S2 Solutions Tool Kit
1
1
SOL-FR
S2 Solutions Cage Nut And Screw Kit For 10 Panels (40 Sets)
1
2
217
Radio Studio Packages
S2 Solutions Single and Split Mixer Furniture
There are two main types of layout
depending on whether you want a single
large mixer, or a split console with 2
smaller mixers connected together, but
leaving a centre space for PC monitors.
S2 Solutions Single Mixer Furniture
This furniture is for use if you want to use a
single mixer, i.e. not a split-frame. With both
guest and presenter positions, the package
is ideal for hospital radios, RSLs (restricted
service licences), sports grounds, colleges and
university media studies departments,as well
as community and commercial radio stations.
The S2 Solutions Single Mixer Package
features the Sonifex S2 broadcast console
combined with quality broadcasting
equipment. The basic furniture is L-shaped
and can be installed either left or righthanded, with the SOL-CVR80 end panel
Single Mixer Furniture, L-Shaped.
information on the input channels available.
A simple on-air studio package can consist
of 1 high quality presenter microphone
on an adjustable microphone arm, 1 guest
microphone on a table stand, 2 pairs of
headphones and a pair of small book-shelf
speakers mounted on desktop speaker
stands, such as the SOL-MS21 pair, behind
the mixer.
protecting the end which is on display.
An additional side return can be added to
make it U-shaped, allowing more space for
turntables, a PC, or a further guest position.
The desktop into which the mixer rests
can be either 1.3m (SOL-MB1300) or 1.5m
(SOL-MB1500) in width, depending on the
space that you have available.
The configuration of the S2 mixer depends
on the equipment that you need for your
studio. See the S2 mixer brochure for more
Single Mixer Furniture with SOL-MBT Talks
Table Top.
There are various audio sources readily
available to the presenter to create
programme content. For the playback of
advertisements, jingles, sweepers, trails,
news items and music, 2 x broadcast
flashcard recorders can be provided.
2 x broadcast CD players for backing music
playout and a CD recorder is ideal for
Solutions Single Mixer Furniture Package.
218
Single Mixer Furniture, U-Shaped.
giving guests copies of interviews, or
for sending commercials to clients. For
PC playout automation systems, we can
Radio Studio Packages
recommend manufacturers to contact and
these can be integrated into the package
quite easily.
The 12U support base pod contains all the
remaining studio equipment. An approved
telephone hybrid, mixer power supply and
power amplifier can be included. An optoisolated, illuminated “mic live” sign can be
located outside the studio door and a studio
clock is also essential, although this can be
built into the S2 mixer if required.
It’s worth noting that the choice, variety and
quality of available equipment is large and
it changes frequently. So, we’ll recommend
the best equipment available at the time of
your enquiry.
S2 Solutions Split Mixer Furniture
This furniture is for use with an S2 split
mixer (i.e. 10-15 channel frame on one side
and 10-15 channel frame on the other). The
S2 Solutions split mixer furniture comprises
Split Mixer Furniture.
U-shaped woodwork which allows space for
up to 3 TFT monitor screens in the centre
of the presenter’s vision for automated
playout, or editing, systems.
Talkback panels can be added and, when
combined with the Sonifex Station Master
studio switcher and router, a number of
studios can be connected together to
control on-air switching.
SOL-CVR80 End Panel.
We can offer you a complete installation
service quoting any equipment that you
would like. We’ve installed central technical
areas as well as studios and fitted krone
cabling on some installations, providing a
turnkey solution for smaller radio stations.
Split Mixer Furniture with SOL-MBCT Talks Table
Desk Top.
This package has space for 2 guest seating
positions, behind the mixers, and is ideal
for small scale radio, local radio or smaller
studios in larger stations.
The choice of equipment is similar to that
of the S2 Solutions single mixer furniture
except that with the larger space available,
you can expand the range and quality of
the other equipment chosen, e.g. you
may need an additional guest microphone
and pair of headphones, a headphone
distribution amplifier, a turntable, an
additional telephone hybrid to have 2 callers
on-air simultaneously or an audio codec to
conduct outside broadcasts.
Solutions Dual Mixer Furniture Package.
219
Radio Studio Packages
Portable Radio Studio Package
Portable Radio Studio Package
The Portable Radio Studio Package is a radio
studio solution for schools, colleges and
many other applications. Ideal for users who
want to move equipment between rooms or
who have limited space.
Features of the System
• Compact and easy to use.
• Pre-wired, professional equipment.
• Robust ash-finished construction with
locking wheels.
• Extendable side wing creating more
useable space..
• Sound deadening surface with safe
rounded corners.
• Choice of industry standard PC hard disk
radio playout systems available.
• Studio equipment and automation
upgrade options available.
• Sonifex brightly coloured SignalLED MicLive sign.
• Headphones and presenter & guest
microphones.
• Advanced automatic output
level control.
Portable Radio
Studio Package
Rear View.
Portable Radio
Studio Package
Front View.
220
Portable Radio
Studio Package
Side View.
Radio Studio Packages
Monitor Speaker Stands
SOL-MS21, SOL-MS32, SOL-MS40,
SOL-MS53 Monitor Speaker Stands
Category: Speaker Mounts.
Product Function: For desktop mounting
monitor speakers.
Typical Applications: In any studio
(radio, TV, recording, voiceover,
production) to mount speakers above
the level of the desktop.
Features:
• Hard wearing.
• Cable hole for hiding cable.
• Desktop mounting.
Made from steel, the stands are strong
and resilient with a black semi-gloss stove
enamel coating. The stands are tubular
and have a cable hole, surrounded
by a black grommet, which sits just
underneath the top round plate so that
speaker cables can be routed into the
tube and hidden inside the tube through
to the underside of a work surface. The
top plate is round and has 4 countersink
holes for screwing into the base of a
speaker. The bottom plate has 4 arms
each with a straight hole for bolting the
speaker stand onto a desktop or similar
surface.
The mounts are supplied with M5 screws,
washers and nuts for the base and No.
6 x 5/8” wood-screws for the top.
S2 Solutions Monitor Speaker Stands
Stock Code
Description
SOL-MS21
Monitor Speaker Stands,
21cm High, Black, Pair
SOL-MS32
Monitor Speaker Stands,
32cm High, Black, Pair
SOL-MS40
Monitor Speaker Stands,
40cm High, Black, Pair
SOL-MS53
Monitor Speaker Stands,
53cm High, Black, Pair
SOL-MS53
SOL-MS40
SOL-MS32
SOL-MS21
SOL-MS21 monitor speaker stands, 21cm (8”) high,
black, pair.
SOL-MS32 monitor speaker stands, 32cm (13”)
high, black, pair.
SOL-MS40 monitor speaker stands, 40cm (16”)
high.
SOL-MS53 monitor speaker stands, 53cm (21”)
high.
221
Radio Studio Packages
S2 Solutions Dimensions
Portable Radio
Studio Package
Aerial View
S2 Solutions Single Mixer
Furniture
Aerial View
S2 Solutions Split Mixer Furniture
Aerial View
SOL-MBCT
1600
800
800
600
700
SOL-D8
1600
SOL-SB1600
SOL-MBL
SOL-MBC
Cut-Out
For
S2 Mixer
400
Front View
1500 (SOL-MB1500)or
1300 (SOL-MB1300)
SOL-SB
1000
Cut-Out
For
S2 Mixer
SOL-MBR
Cut-Out
For
S2 Mixer
SOL-SB
1000
1000
2000
SOL-D8
1000
600
900
2200
Side View
SOL-D8
1515
Side View
Side View
425
SOL-D8
425
SOL-CVR80
802.5
SOL-B12
All dimensions in mm.
222
631
736
SOL-B12
631
+ 40
736 + 40 (adjustable
leg height)
Talkback & Commentary
Talkback Control Unit
The CM-TB8 is a talkback control unit providing 8 channels of 4-wire communication,
housed in a 1U rack mount enclosure. There are two options to add PSTN or GSM interfaces.
CM-TB8 8 Channel Talkback Control Unit
CM-TB8T Talkback Control Unit, CM-TB8 with CM-TBT Dual PSTN Telephone Hybrid
CM-TB8G Talkback Control Unit, CM-TB8 with CM-TBG Dual GSM Interface
224
227
228
Professional Gooseneck Microphone
CM-GM1 Professional Gooseneck Condenser Microphone
229
Commentator Units
There are two commentator units catering for single and dual commentary positions.
CM-CU1 Commentator Unit, 1 Commentator & Line Input
CM-CU21 Commentator Unit, 2 Commentators & 1 Guest
230
234
Beltpacks
CM-BH4W & X Belt-Pack 4-Wire Headphone Amp
CM-BHA Belt Pack Headphone Amplifier With Limiter & Loudspeaker
238
239
Talkback
The Sonifex Talkback Intercoms provide a cost effective way of enabling talkback
between up to 7 areas in a studio complex.
TB-6D & TB-6R 6 Way Talkback Intercoms
240
223
Talkback & Commentary - CM-TB8
CM-TB8 8 Channel Talkback Control Unit
The CM-TB8 is a powerful and highly
flexible self contained communications
unit allowing multiple 4-wires and IFBs
within a single 1U rack space. It offers
easy connection to any industry standard
talkback system including the Sonifex
CM-TLL line listen unit, the CM-BHA and
CM-BH4W belt-packs plus the CM-CU1 and
CM-CU21 commentator units.
The CM-TB8 8 Channel Talkback Control
Unit is ideal in an OB truck or small
studio. The CM-TB8 is a talkback control
unit providing 8 channels of 4-wire
communication, housed in a 1U rack mount
enclosure.
The unit has individual connections for both
a headset on 5 pin XLR and microphone/
Category: Talkback & Commentary.
Product Function: Talkback controller
offering 8 channels of 4 wire/IFB with
options for GSM & POTs comms.
Typical Applications: Ideal in an OB
van/truck or small production gallery
where talkback is needed between up
224
to 8 different areas/groups, which could be
commentators, cameramen, presenters, etc.
Features:
• 8 Channel talkback control unit in a 1U
rack mount package.
• Separate microphone and headset inputs
with phantom power and individual gain
settings.
• Microphone talk level can be instantly
adjusted or preset.
• Talk level LEDs and built-in limiter.
• Configurable ‘Group Talk’ function.
• Interruptible Foldback and direct 4-Wire
•
•
•
•
•
operation selectable on each channel
with optional talkback-microphone mix
function.
2 external IFB (Interruptible Foldback)
inputs available to each channel.
Optional telephone hybrid or GSM ports
allow direct connection to a telephone
line for remote communications.
Individual monitor level control for each
channel.
Microphone output feed which is gain
adjusted & level limited.
Ethernet connection for remote
configuration with built-in web server.
• 6 stored configuration templates with 4
user editable sets.
• Front panel loudspeaker with separate
feed for an external loudspeaker.
• Loudspeaker mute and dim controls,
also available on remote input.
• Remote ‘tally’ outputs indicating when
each channel is active.
• Remote inputs providing a call
indication on each channel.
• Built in tone generator for channel
identification.
Talkback & Commentary - CM-TB8
headphones on XLR/stereo TRS jack, with
the current selection indicated on the front
panel. The gain setting for each input type can
be individually set between +6dB and +68dB
allowing for a wide range of microphone
types and +48V phantom power is also
available. Small talk level adjustments can
be made quickly, using the front panel rotary
control with a maximum adjustment range
of ±12dB. This adjustment can be disabled if
required to stop inadvertent changes. A level
limiter automatically adjusts the microphone
gain if the signal level exceeds a preset level,
and the current talk level is indicated on the
front panel when it exceeds 0dB and +8dB. A
separate output on the rear panel provides
a balanced, gain adjusted and level limited
microphone signal on 3 pin XLR.
Each channel can be set to either 4-Wire or
Interruptible Foldback operation. In 4-Wire
mode the 4-Wire input signal is routed
to the monitor channel (headphones/
headset & loudspeaker) at the level set by
the monitor level potentiometer. When the
TALK button is pressed, the microphone
signal is routed to the rear panel talkback
output connector, and the level of the feed
to the loudspeaker is reduced to the level
preset by the dim potentiometer.
When in IFB mode, the channel’s own
4-Wire input, or one of two external IFB
inputs, is routed to the talkback output
connector and can be selected as the
interruptible signal. It is also possible to
configure each channel so that the input
signal is mixed with the microphone input
rather than being interrupted by it.
When a channel’s TALK button is active,
the level of the signal routed to the
loudspeaker is automatically reduced by a
preset amount, which ranges between full
on and off. The TALK button also acts as a
call indicator. When the signal level on the
4-Wire input of an inactive channel is above
-12dB, the talk button illuminates green.
The TALK and GROUP TALK buttons all have
a momentary action that will automatically
latch on if they are briefly pressed.
The group talk function allows a group
of preselected channels to be activated
simultaneously. The buttons on all activated
channels are illuminated as is the GROUP
TALK button. Each channel can be included
or excluded from group talk function, and by
default, all channels are included.
Each channel has a separate monitor level
control which sets the level of the signal that
is mixed with the other active channels. This
signal is then fed to both the loudspeaker
and headphone monitor channels and its
overall level is controlled by the monitor
level potentiometer. The loudspeaker has
front panel mute and dim attenuation level
controls. An external loudspeaker connection
on the rear panel provides a level adjusted
output and when this output is used, the
signal to the internal speaker is muted.
The unit has a built in line-up tone generator
for easy channel identification and cabling
CM-TB8 Front View.
CM-TB8 Rear View Showing Optional CM-TB8T Card.
225
Talkback & Commentary - CM-TB8
checks. The test-tone generator has 2
modes. In manual mode, tone is routed to
a channel when the corresponding TALK
button is pressed. In auto mode, a tone
pulse is generated on each channel to
uniquely identify it.
A rear panel mounted remote output
connector provides a ‘tally’ indication of
active talk channels and group talk status
via open collector driven outputs. Other
‘tally’ outputs indicate the status of the
loudspeaker dim and mute controls and
also if any of the audio input activated call
indications are active.
configuration templates that allow settings
to be stored and recalled at a later date.
Two of the configuration sets are read only
and cannot be changed. The other four
can be used to store customised settings.
This allows the unit to be preconfigured
for a range of user scenarios which can be
instantly recalled.
Specification For CM-TB8
The Ethernet port allows configuration
settings to be modified using the built-in
web server and an on-board configuration
mode allows several of the configuration
options to be changed without the need of
a PC.
Frequency Response:
Microphone Input
Gain Range:
+6dB to +68dB
The CM-TB8 talkback control unit is
powered from a universal mains input
between 85-264V AC at 47-63Hz.
Close-Up Of The CM-TB8 Front Left Hand Side.
226
Connections
-6dBu
Microphone:
XLR-3 pin female (electronic
balanced)
C.M.R.R.:
>60dB
Headphones:
6.35mm stereo jack socket
Frequency Response:
±0.5dB 20Hz to 15kHz Ref
50dB gain @ 1kHz
Headset:
XLR-5 pin female (input
electronic balanced)
4-Wire Inputs:
8 x XLR-3 pin female
(transformer balanced)
IFB Inputs:
2 x XLR-3 pin female
(transformer balanced)
Talkback Outputs:
8 x XLR-3 pin male
(transformer balanced)
Microphone Output:
XLR-3 pin male (electronic
balanced)
Headphone Output
Gain Range:
-60dB to +18dB
Maximum Output Level: +18dBu
±0.5dB 20Hz to 15kHz
4-Wire And IFB Inputs
Maximum Input Level:
+18dBu
Frequency Response:
±1dB 20Hz to 15kHz
Talkback Outputs
Loudspeaker Output:
6.35mm stereo jack socket
Maximum Output Level: +18dBu
Remote Inputs:
15-way ‘D’-type socket
Frequency Response:
Remote Outputs:
15-way ‘D’-type socket
±1dB 20Hz to 15kHz
Power Output:
6W
Noise:
More than 80dB below full
output
Ethernet:
RJ-45, 10/100Mbps
Telephone:
4 x RJ11 6/4 (2 x line,
2 x handset), present on
CM-TB8T only
Mains Input:
Universal filtered IEC,
continuously rated
85-264VAC, 47-63Hz, 25W
Fuse Rating:
Anti-surge fuse 2A 20 x 5mm
Equipment Type
An optionally fitted telephony add-in card
(CM-TB8T or CM-TB8G respectively) allows
up to 2 channels to be connected to a
telephone or GSM network. This allows
the operator and the talkback users to
communicate with a connected caller and
both 4-wire and IFB operation is available.
The CM-TB8 provides a wide range of
user configurable options and there are 6
Mute to full volume via front
panel potentiometer
Maximum Input Level:
Loudspeaker
A separate remote input connector provides
an additional call input indication on each
channel. When an input is activated, the
corresponding talk button flashes. Inputs to
remotely operate the loudspeaker dim and
mute controls are also available.
Volume:
CM-TB8:
8 channel talkback control
unit
CM-TB8T:
Add-in card with 2 x
telephone hybrids
CM-TB8G:
Add-in card with 2 x
GSM lines
Physical Specification
CM-TB8 Home Page.
CM-TB8 Configuration Channels Page.
Close-Up Of The CM-TB8 Front RightHand Side.
Dimensions (Raw)
Including Connectors:
57cm (W) x 52cm (D) x15cm (H)
22.4” (W) x 20.5” (D) x 5.9” (H)
(1U)
Dimensions
(Boxed):
58cm (W) x 52cm (D) x 14cm (H)
22.8” (W) x 20.5” (D) x 5.5” (H)
Weight:
Nett: 5.54kg Gross: 6.5kg
Nett: 12.2lbs Gross: 14.3lbs
Talkback & Commentary - CM-TB8T
CM-TB8T Talkback Control Unit, CM-TB8 with
CM-TBT Dual PSTN Telephone Hybrid
Category: Talkback & Commentary.
Product Function: Talkback controller
with dual PSTN comms.
Typical Applications: Ideal in an OB van/
truck or small production gallery where
talkback is needed to a remote location, or
other country.
Features:
• Two independent telephone hybrids.
• Ethernet & front panel control.
• Auto-answer & auto disconnect.
• 8 Speed dials available.
• IFB on telephone channels.
line conditions. This setting also configures
the relevant auto disconnect mode which
can be enabled if required. An option to
auto answer an incoming telephone call can
also be enabled.
Specification For CM-TB8T
Bandwidth to
Telephone Line:
125Hz – 3.6kHz, -3dB ref 1kHz
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Impedance:
impedances set via country code
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Impedance Range:
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Calls can be made and received by using an
external telephone handset connected to
each hybrid channel, or directly on the
CM-TB8 using the channel controls. Each
hybrid can independently speed dial one
of the 8 user defined telephone numbers
stored on the unit.
Connections:
2 x RJ11 Socket – line
2 x RJ11 Socket – handset
CM-TBT :
Add-in card for CM-TB8 with
2 x telephone hybrids.
CM-TB8T:
CM-TB8 including CM-TBT.
Physical Specification
Dimensions (Raw):
15cm (W) x 10cm (D) x 3cm (H)
5.9” (W) x 3.9” (D) x 1.2” (H)
Dimensions
(Boxed):
22.9cm (W) x 12.7cm (D) x 7.6cm (H)
9.0” (W) x 5.0” (D) x 3.0” (H)
IFB modes are fully supported on the
telephone channels allowing selected signals
to be routed to a connected telephone call.
Weight:
Nett: 0.2kg Gross: 0.3kg
Nett: 0.4lbs Gross: 0.7lbs
CM-TB8T PSTN Configuration Page.
CM-TB8T Channels Config Page.
CM-TBT Dual PSTN Card For the CM-TB8T.
The CM-TB8T is the CM-TB8, 8 Channel
Talkback Control Unit, with the CM-TBT
add-in card fitted.
The CM-TBT is a dual PSTN, telephone
hybrid expansion card. It extends the
talkback capability of the CM-TB8 by
allowing telephone calls to be made and
received on two independent analogue
direct exchange or PSTN lines.
The hybrids are fully configurable and are
easy to setup via the CM-TB8 embedded
web server. Simply select your location
from a country code list to automatically
configure the telephone interface for local
227
Talkback & Commentary - CM-TB8G
CM-TB8G Talkback Control Unit, CM-TB8 with
CM-TBG Dual GSM Interface
Category: Talkback & Commentary.
Product Function: Talkback controller
with dual GSM comms.
Typical Applications: Ideal in an OB van/
truck or small production gallery where
talkback is needed from a remote location,
or to staff in a remote location.
Features:
• Two separate quad band GSM circuits.
• Ethernet & front panel control.
• Auto-answer & auto disconnect.
• 8 Speed dials available.
• IFB on GSM channels.
CM-TBG Dual GSM Card For the CM-TB8G.
The CM-TB8G is the CM-TB8, 8 Channel
Talkback Control Unit, with the CM-TBG
add-in card fitted.
The CM-TBG is a dual GSM telephone
expansion card. It extends the talkback
capability of the CM-TB8 by allowing mobile
telephone calls to be made and received
via two independent quad band mobile
telephone modules.
Telephone calls are made and received using
the embedded web server or directly using
the channel controls. The CM-TB8 can store
8 telephone numbers and each channel can
speed dial the required number.
IFB modes are fully supported on the
telephone channels allowing selected
input signals to be routed to a connected
telephone call.
Module Type:
Quad-Band EGSM 850 / 900 / 1800
/1900MHz
Output Power:
Class 4 (2W) @ 850 / 900MHz
Class 1 (1W) @ 1800 / 1900MHz
Sensitivity:
-107 dBm (typ.) @ 850 / 900MHz
-106 dBm (typ.) @ 1800 / 1900MHz
Approvals:
Fully type approved conforming to
R&TTE, CE, GCF, FCC, PTCRB, IC, Anatel
Connections:
2 x SMA Socket – antenna
CM-TBG :
Add-in card for CM-TB8 with
2 x GSM line.
CM-TB8G :
CM-TB8 including CM-TBG.
Physical Specification
The GSM modules are configured via
the CM-TB8 embedded web server. This
provides access to features including
network selection, telephone number
storage and phone logs. Each module has
its own internal SIM card slot and separate
antenna connection which is mounted on
the rear panel of the CM-TB8.
CM-TB8G GSM Configuration Page.
228
Specification For CM-TB8G
CM-TB8G GSM Enable Page.
Dimensions
(Raw):
15cm (W) x 10cm (D) x 3cm (H)
5.9” (W) x 3.9” (D) x 1.2” (H)
Dimensions
(Boxed):
22.9cm (W) x 12.7cm (D) x 7.6cm (H)
9.0” (W) x 5.0” (D) x 3.0” (H)
Weight:
Nett: 0.2kg Gross: 0.3kg
Nett: 0.4lbs Gross: 0.7lbs
Talkback & Commentary - CM-GM1
CM-GM1 Professional Gooseneck Condenser
Microphone
Category: Talkback & Commentary.
Product Function: Gooseneck mic for
use with commentary and talkback
systems.
Typical Applications: Use with the CMTB8, CM-CU21, CM-CU1 or any system
which supports an XLR mic input. Large
This is a professional gooseneck condenser
microphone which has a back electret
condenser element with an ultra-cardioid
polar pattern.
A red LED circling the head of the
microphone illuminates when phantom
power is applied to the mic. At the base of
the mic is a male XLR plug connector to plug
into intercoms and talkback systems.
It is 35cm long and has two flexible sections
near the head and the base of the mic
so that it can be moved into different
orientations.
scale conferences, public addresses and
outdoor speeches.
Features:
• High performance, extremely clear
sound.
• High sensitivity, low self noise.
• Exceptional frequency response.
• Durable flexible gooseneck.
• XLR male base connection.
• Red LED around collar illuminates when
phantom power is on.
• 350mm flexible neck.
Specification For CM-GM1
Element:
Black electret condenser
Polar pattern:
Ultra cardioid
Frequency
Response:
50Hz – 15kHz
Sensitivity:
-38dB +- 3dB (0dB=1V/Pa at 1kHz)
Phantom Power
Requirement:
9V-52V DC phantom.
Equipment Type
CM-GM1 :
Professional gooseneck condenser
microphone
Physical Specification
Dimensions
(Raw):
47cm (L) x 1.5cm (Dia)
18.5” (L) x 0.6” (Dia)
Dimensions
(Boxed):
50cm (W) x 4cm (D) x 4cm (H)
19.7” (W) x 1.6” (D) x 1.6–” (H)
Weight:
Nett: 0.1kg Gross: 0.2kg
Nett: 0.2lbs Gross: 0.4lbs
229
Talkback & Commentary - CM-CU1
CM-CU1 Commentator Unit 1 Commentator & Line Input
•
Category: Talkback & Commentary.
Product Function: Commentator unit for
a single commentary position.
Typical Applications: Used in sports
grounds and arenas by commentators to
produce commentary & hear talkback.
Features:
• Commentator position with individual
output; plus main mix output.
• Line input with configurable output
and headphone presence.
• Talkback outputs selectable by the
commentator, with built in limiter and
adjustable gain.
• 5 Return audio circuits that are
routed to headphones along with the
The CM-CU1 offers the same high quality,
robust specification as the CM-CU21, with a
feature set tailored to single commentator
applications.
It provides a fully featured commentator
position and a line-level input. The unit has
an individual commentator output, with
an additional output providing a mix of
commentator and line input audio. A limit
230
•
•
•
•
•
•
•
commentator input. Any source can be
individually level controlled and sent to
left, right or both ears.
A wide input gain range and switchable
phantom power allows support of a
variety of microphone types including
low output ribbon, headsets and high
output phantom powered microphones.
User preset adjustable limiter on main
output with limit activity indication.
Built in line-up tone for initial cabling
checks.
21 Segment LED PPM meter showing
main output level.
Transformer balanced inputs & outputs.
Fitted formica ‘Sharpie’ designation
strips.
Robust mechanical design with
protection for potentiometer knobs.
Universal AC mains, or 12-24 volt DC
power.
indicator on the main panel shows when
the adjustable limiter on the main output
is active, and a bright 21 segment LED PPM
meter, which can be disabled, shows the
main output level.
The line input can be configured to remain
present at the mix output even if the
commentator is off air, making it useful
for routing crowd effects or pre-recorded
CM-CU1 Commentator Unit Top View.
Talkback & Commentary - CM-CU1
material to the programme feed. Its
presence in the commentator’s headphones
is also configurable to suit the application.
Four talkback output channels, with a built
in limiter, are available to the commentator
and they can be linked to provide
simultaneous operation. Activation of one
or more talkback channels removes the
commentator audio from the main output
until all talkback channels are deactivated.
A wide input gain range and switchable
phantom power allows support of a variety
of microphone types including low output
ribbon mics, headsets and high output
phantom powered microphones. There
are 5 transformer balanced return audio
inputs which, along with the commentator
input, are routed to the headphone monitor
controls. With these headphone controls,
any monitor input can be adjusted and sent
to either the left, right or both ears.
CM-CU1 Commentator Unit Front View.
CM-CU1 Commentator Unit Rear View.
231
Talkback & Commentary - CM-CU1
A GPIO port gives an open-collector driven
output indication of any active On-Air or
talkback channel and four remote inputs
provide an external call/alert to each of the
talkback controls.
The unit has a built in line-up tone generator
for easy channel identification and cabling
checks. When enabled the test-tone is
routed to the required output simply
by pressing the corresponding on air or
talkback button. An automatic tone mode
repeatedly sends different quantities of tone
bursts to each output to make it effortless
for a single technician to set up.
An On-Air lock switch prevents the On-Air
button from being accidentally deactivated
and a T/B Gang switch links the operation of
talkbacks A, B, C & D.
A flexible range of options allow the
Commentator Unit to be customised to
better suit the operational and user’s
personal requirements. These include
options for momentary or latching operation
of the talkback buttons, permanently
enabling or disabling the commentator
input, default headphone signal routing and
main panel indicator illumination levels.
The CM-CU1 Commentator Unit can be
powered from a universal mains input
between 85-264V AC at 47-63Hz, or from a
12-24V DC input.
232
CM-CU1 Commentator Unit Iso View.
Talkback & Commentary - CM-CU1
Specification For CM-CU1
Transformer Balanced Talkback Outputs
Front Panel Operational Controls
Rear Panel Connections
Maximum Output
Level Limited:
+8dBu
Phantom:
Switch to enable 48V microphone
phantom power
Prog, A, B, C & D Inputs: 5 x XLR-3 pin female (transformer
balanced)
Equivalent Input Noise: 130dB Ref. 80dB gain with
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Frequency Response:
±0.5dB 20Hz to 22kHz
Ref 0dBu @ 1kHz
Mic Gain:
Comm Output:
1 x XLR-3 pin male (electronic
balanced)
Equivalent Input Noise: 128dB Ref. 50dB gain with
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Noise:
-94dB Average weighting
Ref +8dBu
Microphone gain potentiometer
and range select switch selects
+20dB to +56dB or +50 to +86dB
Rear Panel Operational Controls
Talkback A, B, C & D
Outputs:
4 x XLR-3 pin male (transformer
balanced)
C.M.R.R.
>60dB
THD & N:
<0.02% Ref. 1kHz @ +8dBu
On Air Lock:
Gain Adjustment:
0dB to +12dB
1 x XLR-3 pin male (transformer
balanced)
±0.5dB 20Hz to 22kHz
Ref 50dB gain @ 1kHz
Switch to prevent On Air control
from being deactivated
Momentary push-button to
override on air lock
Mix Output:
Frequency Response:
T/B Gang:
Switch to link operation of
talkbacks A, B, C & D
T/B Gain:
0dB to +12dB
Limiter:
Switch to enable limiter on mix
output
Limit level adjustment
potentiometer
Test Tone:
Switch to enable 1 kHz line-up
tone
Microphone Input
Gain Range:
+20dB to +86dB
Maximum Input Level:
-6dBu
Electronically Balanced Comm Output
Electronically Balanced Line Input
Maximum Output Level: +25dBu
Gain Range:
-83dB to +10dB
Frequency Response:
Maximum Input Level:
+25dBu
±0.2dB 20Hz to 22kHz
Ref 0dBu @ 1kHz
-98dB Average weighting
Ref+8dBu
<0.002% Ref. 1kHz @ +8dBu
C.M.R.R.
>70dB
Noise:
Frequency Response:
±0.2dB 20Hz to 22kHz
Ref 0dBu @ 1kHz
THD & N:
GPIO Port:
15-way ‘D’-type socket
Mains Input:
Universal filtered IEC, continuously
rated 85-264VAC, 47-63Hz, 10W
Fuse Rating:
Anti-surge fuse 1A 20 x 5mm
DC Input:
XLR-4 pin male 12V DC, 650mA
Typical, 850mA Maximum
24V DC, 325mA Typical, 425mA
Maximum
Earth Point:
M4 stud.
Transformer Balanced Monitor Inputs
Headphone Output
Gain Range:
-66dB to +11dB measured @
Headphone output
Gain Range:
Maximum Output Level: +19dBu
Configuration Options – Accessible via Setup Mode
CM-CU1
Maximum Input Level:
+21dBu
Frequency Response:
On Air Control Mode:
Physical Specification
C.M.R.R.
>55dB
±0.5dB 20Hz to 22kHz Ref 0dBu
@ 1kHz
Frequency Response:
±0.5dB 20Hz to 22kHz
Ref 0dBu @ 1kHz
Noise:
-85dB Average weighting
Ref +8dBu
Talkback Button Action: Momentary, latching or both
Transformer Balanced Mix Output
THD & N:
<0.005% Ref. 1kHz @ +8dBu
Maximum Output Level +24dBu
(Limiter Off):
Main Panel Operational Controls & Indicators
Frequency Response:
±0.5dB 20Hz to 22kHz
Ref 0dBu @ 1kHz
Noise:
-94dB Average weighting
Ref +8dBu
THD & N:
<0.02% Ref. 1kHz @ +8dBu
THD & N:
<0.065% Ref. 1kHz @ +18dBu
1% Distortion Point:
+18dBu @ 24Hz
-66dB to +11dB
Headphone Routing:
Disabled, normal or permanently
on
Line-level input permanently
routed/routed if on air to
Commentator headphones
Commentator input permanently
routed/not routed to headphone
monitors
Equipment Type
Commentator unit, 1 presenter
Dimensions (Raw):
27cm (W) x 23cm (D) x 7cm
(H - front) x 9.5cm (H - rear)
10.6” (W) x 9” (D) x 2.8” (H - front
x 3.7” (H - rear)
Dimensions (Boxed):
38cm (W) x 35cm (D) x 20cm(H)
15” (W) x 13.8” (D) x 7.8” (H)
Weight:
Nett: 3.3kg Gross: 4.3kg
Nett: 7.3lbs Gross: 9.4lbs
On-Air Button:
Illuminated button with latching
action
Talkback A, B, C & D
Buttons:
Illuminated buttons with
selectable momentary and/or
latching action
Line Level Input Routing: Line-level input permanently
routed/not routed/routed if on air
to main output
Accessories
CM-CUTC:
Commentator unit transport case
Headphone Monitor
Controls:
Input source level adjustment
potentiometers
Headphone channel selection
switches
Display Preferences:
Limit LED colour and brightness
Button LED brightness
PPM meter LED brightness/disable
Dimensions (Raw):
53cm (W) x 21cm (D) x 41.5cm (H)
20.9” (W) x 8.3” (D) x 16.3” (H)
Dimensions (Boxed):
line input mode level
potentiometer
Permanent Option
Backup:
Save/load options to/from
Permanent backup or return to
factory defaults
60cm (W) x 26cm (D) x 42cm (H)
23.6” (W) x 10.2” (D) x 16.5” (H)
Line Input Level:
Weight:
Net: 4.0kg Gross: 5.0kg
Net: 8.8lbs Gross: 11.0lbs
PPM Meter:
21 segment display showing mix
output level
Front Panel Connections
Limit LED:
Indicates main output limiter is
active, colour & brightness level is
selectable
Headphone Output:
1 x locking headphone jack socket
Microphone Input:
1 x XLR-3 pin latching female
Line Input:
1 x XLR-3 pin latching female
Power LED:
Indicates operating mode; normal
or setup
233
Talkback & Commentary - CM-CU21
CM-CU21 Commentator Unit 2 Commentators & 1 Guest
•
Category: Talkback & Commentary.
Product Function: Commentator unit for
two commentary positions and a guest
position.
Typical Applications: Used in sports
grounds and arenas by commentators to
produce commentary & hear talkback.
Features:
• 2 Commentator positions plus a guest
position, each with individual outputs;
plus 2 mix outputs.
• Talkback outputs selectable by the
two main commentators, with built in
limiter & adjustable gain.
• 4 Return audio circuits that are routed
to headphones along with all the
commentator inputs. Any source can
be individually level controlled and
•
•
•
•
•
•
•
•
sent to left, right or both ears.
A wide input gain range and switchable
phantom power on each commentary
position allows support of a variety of
microphone types including low output
ribbon, headsets and high output
phantom powered microphones.
Individually switched 48V phantom
power on all microphone inputs.
User preset adjustable limiter on
main mix outputs with limit activity
indication.
Built in line-up tone for cabling checks.
21 Segment LED PPM meter showing
main output level.
Transformer balanced inputs and
outputs.
Fitted formica ‘Sharpie’ designation
strips.
Robust mechanical design with
protection for potentiometer knobs.
Universal AC mains, or 12-24 volt
DC power.
The CM-CU21 is a high quality, portable
Commentator Unit. Its sturdy construction
and flexibility of features make it suitable
for use in a wide variety of environments.
Sonifex has a solution for those
situations where a separate commentary
unit is required, or where ISDN/pots
communications, or a complex digital
solution aren’t needed. The CM-CU21
stand alone commentator unit is a rugged,
reliable solution which includes all the
features needed to ensure that up to two
play-by-play announcers can provide their
expertise, together with a guest position for
CM-CU21 Commentator Unit Top View.
234
Talkback & Commentary - CM-CU21
further analysis by a visiting ex-professional,
pundit (color commentator in the USA), or
co-commentator. The guest position can
alternatively be used as a line-level input
for special effects such as the sounds of
the action and spectators, also heard in the
background.
Each of the 3 announcer positions has
an individual output, with two additional
outputs providing a mix of all active
commentary channels. A limit indicator on
the main panel shows when the adjustable
limiter on the mix outputs is active, and a
bright 21 segment LED PPM meter, which
can be disabled, shows the main output
level. Two talkback output channels, with
a built in limiter and adjustable gain,
are available to each of the two main
commentators and they can be linked to
provide simultaneous operation.
A wide input gain range and switchable
phantom power on each commentary
CM-CU21 Commentator Unit Front View.
CM-CU21 Commentator Unit Rear View.
235
Talkback & Commentary - CM-CU21
position allows support of a variety of
microphone types including low output
ribbon mics, headsets and high output
phantom powered microphones. The
microphone level controls are accessible on
the front panel but can be protected from
adjustment by an optional cover plate if
required. There are 4 transformer balanced
return audio inputs which, along with the
commentator inputs, are routed to the
headphone monitor controls. These could
be used to pass referee remarks to the
sportscasters, as well as comments from
guests in the studio.
The headphone input controls are very
comprehensive allowing either of the two
main announcers total control so that any
monitor input can be adjusted in volume
and sent to either the left, right or both
ears. The switches are slightly recessed so
that they can’t easily be knocked and the
volume knobs are fitted so that the bottom
skirt of the knob is below the level of the
panel, so that they can’t fall off. Fitted
Formica ‘Sharpie’ designation strips are
used under each of the level controls so
that different input and source types can be
dynamically annotated.
A GPIO port gives an independent opencollector driven output indication for any
of the three active On-Air or four talkback
channels. Four remote inputs provide an
external call/alert to each of the talkback
controls which flashes the selected T/B
button amber when used, to indicate active
talkback.
The unit has a built in line-up tone generator
for easy channel identification and cabling
checks. When enabled, the test-tone is
routed to the required output simply
by pressing the corresponding on air or
talkback button.
An On-Air lock switch prevents the On-Air
buttons from being accidentally deactivated
and this can be over-ridden with a single
push button on the rear panel for individual
circumstances where an announcer needs to
go off air. A talkback (T/B) gang switch links
the operation of talkbacks A & B to make
the talkback buttons easier to use by the
announcers.
CM-CU21CP Commentator Unit Front Cover Plate.
A flexible range of options allow the
CM-CU21 to be customised to better
suit the operational and user’s personal
requirements. These include options for
momentary or latching operation of the
talkback buttons, permanently enabling or
disabling any of the commentator inputs,
default headphone signal routing and main
panel indicator illumination levels.
The CM-CU21 Commentator Unit can be
powered from a universal mains input
between 85-264V AC at 47-63Hz, or from
a 12-24V DC input, both on the rear panel.
A transport case, the CM-CU21TP, is also
available to protect the CM-CU21 which
also has space for two headsets and/or
microphones.
Accessories
CM-CU21CP: This is an optional
Commentator Unit Front Cover Plate which
is attached to the front of the CM-CU21
to avoid accidental tampering with the
microphone gain and phantom power
control settings.
CM-CUTC: This is an optional carry case
to transport the CM-CU1 or CM-CU21
commentator unit along with 2 x headsets
and 2 x ribbon mics.
CM-CUTC Commentator Unit Transport Case.
236
Talkback & Commentary - CM-CU21
Specification For CM-CU21
Electronically Balanced Comm Outputs
Rear Panel Operational Controls
Rear Panel Connections
Maximum Output
Level:
+25dBu
On Air Lock:
Prog, A, B & C Inputs:
4 x XLR-3 pin female
(transformer balanced)
Equivalent Input Noise: 130dB Ref. 80dB gain with
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Frequency Response:
±0.2dB 20Hz to 22kHz
Ref 0dBu @ 1kHz
Switch to prevent On Air controls
from being deactivated
Momentary push-button to
override on air lock
Comm1-3 Outputs:
3 x XLR-3 pin male (electronic
balanced)
Equivalent Input Noise: 128dB Ref. 50dB gain with
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Noise:
-98dB Average weighting
Ref+8dBu
T/B Gang:
Switch to link operation of
talkbacks A & B
Talkback A & B Outputs: 2 x XLR-3 pin male (transformer
balanced)
C.M.R.R.
>60dB
THD & N:
<0.002% Ref. 1kHz @ +8dBu
T/B Gain:
0dB to +12dB
Mix 1 & 2 Outputs:
Frequency Response:
±0.5dB 20Hz to 22kHz
Ref 50dB gain @ 1kHz
Headphone Outputs
Limiter:
Switch to enable limiter on
mix outputs
Limit level adjustment
potentiometer
GPIO Port:
15-way ‘D’-type socket
Mains Input:
Universal filtered IEC,
continuously rated
85-264VAC, 47-63Hz, 10W
Fuse Rating:
Anti-surge fuse 1A
20 x 5mm
DC Input:
XLR-4 pin male 12V DC,
650mA Typical, 850mA Maximum
24V DC, 325mA Typical,425mA
Maximum
Earth Point:
M4 stud.
Microphone Input
Gain Range:
+20dB to +86dB
Maximum Input Level: -6dBu
Gain Range:
-70dB to +7dB
Electronically Balanced Line Input
Maximum Output Level: +19dBu
Gain Range:
Frequency Response:
±0.5dB 20Hz to 22kHz
Ref 0dBu @ 1kHz
Test Tone:
-85dB Average weighting
Ref +8dBu
Configuration Options – Accessible via Setup Mode
-83dB to +10dB
Maximum Input Level: +25dBu
C.M.R.R.
>70dB
Noise:
Frequency Response:
±0.2dB 20Hz to 22kHz
Ref 0dBu @ 1kHz
THD & N:
On Air Control Mode:
<0.005% Ref. 1kHz @ +8dBu
Switch to enable 1 kHz
line-up tone
Disabled, normal or
permanently on
Transformer Balanced Monitor Inputs
Main Panel Operational Controls & Indicators
Talkback Button Action: Momentary, latching or both
Gain Range:
On-Air 1-3 Buttons:
Headphone Routing:
-66dB to +11dB measured
@ Headphone output
Illuminated buttons with
latching action
C.M.R.R.
>55dB
Talkback A & B Buttons: Illuminated buttons with
selectable momentary and/or
latching action
Frequency Response:
±0.5dB 20Hz to 22kHz
Ref 0dBu @ 1kHz
Headphone Monitor
Controls:
Input source level
adjustment potentiometers
Headphone channel
selection switches
Line Input Level:
Guest position line input
mode level potentiometer
PPM Meter:
21 segment display showing
mix output level, brightness
level Is selectable
Maximum Input Level: +21dBu
Transformer Balanced Main Outputs
Maximum Output Level
(Limiter Off):
+24dBu
Frequency Response:
±0.5dB 20Hz to 22kHz
Ref 0dBu @ 1kHz
Noise:
-94dB Average weighting
Ref +8dBu
THD & N:
<0.02% Ref. 1kHz @ +8dBu
THD & N:
<0.065% Ref. 1kHz
@ +18dBu
1% Distortion Point:
Limit LED:
Indicates mix output limiter is
active, colour & brightness
level is selectable
Power LED:
Indicates operating mode;
normal or setup
+18dBu @ 24Hz
Transformer Balanced Talkback Outputs
Front Panel Operational Controls
Maximum Output
Level Limited:
+8dBu
Phantom:
Switch to enable 48V microphone
phantom power for positions 1 & 2
Frequency Response:
±0.5dB 20Hz to 22kHz
Ref 0dBu @ 1kHz
Phantom/Line:
Noise:
-94dB Average weighting
Ref +8dBu
Switch to enable 48V microphone
phantom power or select line
input mode for position 3
Mic Gain:
Microphone gain potentiometer
and range select switch selects
+20db to +56dB or +50dB to +86dB
THD & N:
<0.02% Ref. 1kHz @ +8dBu
Gain Adjustment:
0dB to +12dB
Display Preferences:
Permanent Option
Backup:
Position 3 line mode input
routed/not routed to
positions 1 & 2 headphones
Position 1-3 inputs permanently
routed/not routed to headphone
monitors
Limit LED colour and
brightness
Button LED brightness
PPM meter LED brightness
disable
Save/load options to/from
permanent backup or return
to factory defaults
Front Panel Connections
Headphone Outputs:
Microphone Inputs:
`
Equipment Type
CM-CU21
Commentator unit,
2 presenter and 1 guest
positions
Physical Specification
Dimensions (Raw):
27cm (W) x 23cm (D) x 7cm
(H - front) x 9.5cm (H - rear)
10.6” (W) x 9” (D) x 2.8” (H - front)
x 3.74” (H - rear)
Dimensions (Boxed):
38cm (W) x 35cm (D) x 20cm (H)
15” (W) x 13.8” (D) x 7.8” (H)
Weight:
Nett: 3.34kg Gross: 4.32kg
Nett: 7.35lbs Gross: 9.5lbs
2 x locking headphone jack
socket for positions 1 & 2
1 x headphone jack socket
for position 3
Accessories
2 x XLR-3 pin latching female
for positions 1 & 2
CM-CUTC:
Microphone/Line Input: 1 x XLR-3 pin latching
female for position 3
2 x XLR-3 pin male (transformer
balanced)
CM-CU21CP:
Commentator unit front
cover plate
Commentator unit transport case
Dimensions (Raw):
53cm (W) x 21cm (D) x 41.5cm (H)
20.9” (W) x 8.3” (D) x 16.3” (H)
Dimensions (Boxed):
60cm (W) x 26cm(D) x 42cm (H)
23.6” (W) x 10.2” (D) x 16.5” (H)
Weight:
Net: 4.0kg Gross: 5.0kg
Net: 8.8lbs Gross: 11.0lbs
237
Talkback & Commentary - Belt Packs
CM-BH4W Belt-Pack 4-Wire Headphone Amp
Category: Talkback & Commentary.
Product Function: Battery powered
microphone and headphone amps allow
use for talkback & 4-wire applications.
Typical Applications: Used in TV studio
floors, sports grounds and arenas by
presenters and cameramen to produce
content and to hear talkback.
Features:
• Transformer balanced inputs & outputs.
• 4-wire and IFB modes.
• Low battery warning indicator.
• 150 hours from a single 9V PP3 battery.
Specification For CM-BH4W & X
Mic In to Line Out
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dynamic microphone
Limiter Input Threshold: -60dBu
Line Output Level:
+2dBu
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Line In to Headphone Out
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Input Gain/Volume:
CM-BH4W Front, Rear and Angled View.
The CM-BH4W is a portable, battery
powered 4 wire communication unit for use
in a multitude of TV and radio broadcasting
applications. A battery powered mic input
and headphone output allow monitoring of
talkback and creation of content.
The CM-BH4W uses audio transformers
at its input and output, providing galvanic
isolation from outside interference, ensuring
your communications are loud and clear,
every time.
In 4-wire mode, the unit acts as a 2-way
intercom, with input audio routed to the
headset earpiece, and microphone audio
is routed to the output upon pressing the
238
talk button. The output level is maintained
at +2dBu peak level by a built-in AGC.
In IFB mode, the input audio is routed
to the output, and is interrupted by the
microphone signal when the talk switch
is pressed. The talk switch can easily be
converted from momentary to latching
operation to suit your application.
A low battery warning LED flashes when
the 9V PP3 battery has depleted to 6V,
giving plenty of notice to swap the battery.
In addition, the CM-BH4W automatically
switches off when the headset jack is
removed, and the low power circuitry gives
up to 150 hours use from a single alkaline
9V PP3 battery.
Adjustable off to +10dB
Maximum Input Level:
+24dBu
Maximum HP Output
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+7dBu (ref 9V battery voltage,
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Impedance:
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Connections
Line Input:
1 x XLR 3 pin female socket
(balanced)
Line Output:
1 x XLR 3 pin male plug
(balanced)
Headset (CM-BH4W):
1 x ¼” (6.35mm) mono
unbalanced jack socket
Headset (CM-BH4WX): 1 x XLR 4 pin male plug
(balanced)
Equipment Type
CM-BH4W:
Belt-pack 4-wire headphone
amplifier
CM-BH4WX:
Belt-pack 4-wire headphone
amplifier with 4 pin XLR
headset connector
CM-BH4W Top and Bottom View.
The CM-BH4WX is is a version of the CMBH4W with a 4 pin XLR headset connection
instead of a stereo jack connector. In all other
respects it is the same as the CM-BH4W.
Physical Specification
Dimensions
(Raw):
12.5cm (W) x 6.7cm (D) x 4.8cm (H)
4.9” (W) x 2.6” (D) x 1.9” (H)
Dimensions
(Boxed):
22.9cm (W) x 12.7cm (D) x 7.6cm (H)
9.0” (W) x 5.0” (D) x 3.0” (H
Weight:
Nett: 0.3kg
Nett: 0.6lbs
Gross: 0.4kg
Gross: 1.0lbs
CM-BH4WX Top View.
Talkback & Commentary - Belt Packs
CM-BHA Belt Pack Headphone Amplifier With Limiter & Loudspeaker
Category: Talkback & Commentary.
Product Function: Battery powered
headphone amp allows use for talkback
& 4-wire applications.
Typical Applications: Used in TV studio
floors, sports grounds and arenas by
presenters and cameramen to hear
talkback.
Features:
• Wide signal range capability.
• Transformer balanced inputs & outputs.
• Adjustable limiter protects the listener.
• Low battery warning indicator.
• 200 hours from a single 9V PP3 battery.
The CM-BHA is a portable, battery
powered headphone amplifier for use in
live applications such as in-the-field news
reports.
The CM-BHA has a wide signal range and can
apply gain of up to +20dB to the incoming
signal. The volume control simultaneously
modifies the output volume and the input
gain, allowing input signals in excess of
+24dBu to be reproduced without distortion.
CM-BHA Front, Rear and Angled View.
CM-BHA Top and Bottom View.
The CM-BHA has an adjustable limiter to
protect the listener from excessive levels
indicated by a blue LED. The limiter is only
enabled when an earpiece or headphones are
used, and when it is activated by the toggle
switch. It is disabled when no jack is inserted, Losing a cue feed due to batteries running
allowing a higher volume to be passed to the out is not an option. With the CM-BHA, a
speaker for “squawk mode” use.
low battery warning LED flashes when the
9V PP3 battery has depleted to 6V, giving
plenty of notice to swap the battery. The low
power circuitry of the CM-BHA gives up to
200 hours use from a single alkaline 9V PP3
battery.
Specification For CM-BHA
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Maximum HP
+16dBu (ref 9V battery voltage,
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+12dBu (ref 6V battery voltage,
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DŝŶŝŵƵŵ,W
Impedance:
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Maximum
Speaker Level:
+12dBu (ref 9V battery voltage,
<1% THD)
+6dBu (ref 6V battery voltage,
<1% THD)
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Input Limiter
Threshold:
Adjustable -14dBu to 0dBu
Maximum Input
Level:
+28dBu
CM-BHA Belt Pack Headphone Amplifier with Limiter
& Loudspeaker
Input Gain/
Volume:
Adjustable -70dB to +20dB
Physical Specification
Frequency
Response:
20Hz – 20kHz ±0.5 (ref 0dBu 1kHz)
Dimensions
(Raw):
12.5cm (W) x 6.7cm (D) x 4.8cm (H)
4.9” (W) x 2.6” (D) x 1.9” (H)
Noise:
<-75dBu A-weighted (ref 0dB gain)
Dimensions
(Boxed):
22.9cm (W) x 12.7cm (D) x 7.6cm (H)
9.0” (W) x 5.0” (D) x 3.0” (H)
Distortion:
<0.015% (ref +8dBu input, 0dB gain,
Weight:
Nett: 0.3kg Gross: 0.4kg
Nett: 0.7lbs Gross: 0.9lbs
Equipment Type
239
Talkback & Commentary - TB-6D & TB-6R
TB-6D & TB-6R 6 Way Talkback Intercoms
The Sonifex Talkback Intercoms provide
a cost effective way of enabling talkback
between up to 7 areas in a studio complex.
Talkback refers to the standard method of
communication between broadcast studios.
Category: Talkback Intercoms.
Product Function: Provides
communication & talkback between up
to 7 studios/areas.
Typical Applications: Talkback within a
facility or outside broadcast vehicle.
Features:
• Up to 7 areas can be connected.
• DC signalling system.
• Momentary & latched button
operation.
• Built-in electret mic.
• Front panel mic optional.
• 2 stereo external inputs.
• Built-in loudspeaker.
• Headphones with level control.
Each studio can have either a desktop, a
rack-mount, or mixer-mounted intercom,
usually containing a row of switch buttons.
Each switch button connects to another
studio when pressed, and routes the
presenter audio to it.
There are 2 products in the Sonifex
Talkback Intercom range :
The TB-6D and TB-6R are used for general
talkback between studios and use DC
signalling, also known as ground lifting or
ground signalling, to communicate with
each other. These units can be installed
in various locations throughout a studio
complex and linked to one another, with
each unit being able to talkback to up to 6
others connected together.
Most other talkback systems in general
operation also use DC signalling, so you
should be able to add these units into an
existing installation.
TB-6D 6 Way Talkback Intercom, desktop
free standing
TB-6R 6 Way Talkback Intercom, 19” rack
mounted
The TB-6D and TB-6R are general
talkback interfaces which can
be connected together to
form a talkback system.
TB-6D 6 Way Talkback Intercom Front View.
240
Desktop Or Rackmount
The TB-6D and TB-6R are identical in
operation, with the TB-6D used for desktop
operation and the TB-6R available as a 2U
rack-mount, for use for example, in a central
apparatus room or central technical area.
Talkback For Up To 7 locations
Up to 7 locations or studios can
communicate with each other.
On each panel, there are 6 front panel
buttons for talkback selection and 2 for
external inputs. When a button is pressed,
the button lights up and the talkback is
active to that location.
Talkback & Commentary - TB-6D & TB-6R
Press & Release, Press & Hold
Operation
The buttons work with both a momentary
and latched operation. If you simply press
the button, talkback will be on and can be
cancelled by pressing the button again. If
you press and hold the button, the selection
is cancelled when released. So, if you
wanted to listen to an external input, such
as the radio station feed, or off-air monitor,
you could simply press the EXT 1 button. If
you receive talkback from another studio
when the external input buttons are also
pressed, the external inputs can be either
TB-6D Rear View.
muted or mixed, depending on the setting
of an internal jumper.
The talkback outputs and input are balanced
signals on a 15-way male D-type plug and
the talkback input can be adjusted from
-28dB to 9dB using a rear panel preset pot.
Onboard Electret Mic or External Mic
You can speak to the separate locations
using the onboard front-panel electret mic
or there is a separate external mic input on
a 3 pin XLR female connector, into which you
can plug a suitable gooseneck microphone.
There is a rear panel switch to select
between the electret and separate input
mic. The mic input gain range is adjustable
via a rear panel preset pot between 74dB
and 53dB.
monitoring a local source, such as a portable
flashcard recorder/player, or the unbalanced
soundcard output from a PC editing/playback
package.
Monitoring External Inputs
The unit can monitor two external sources.
The EXT 1 input is a balanced stereo input
on a rear panel 9 pin D-type socket. The EXT
2 input is an unbalanced stereo input on the
9 pin D-type socket as well as dual phono
sockets. EXT 1 is useful for monitoring a
distributed signal, such as a radio tuner
off-air feed, or an outside source routed
from an ISDN codec. EXT 2 is useful for
If you already use a mixer which has a
talkback input and output, or continuous
talkback output, or just a line level output
which you want to use, by changing internal
jumpers the intercom can be configured to
use these connections instead of the onboard microphone and speaker/headphones.
For example, if you have a mic channel on a
mixer assigned to talkback and the presenter
is using headphones for monitoring the
mixer talkback, the Talkback Intercom can
be used purely for talkback switching and for
monitoring external inputs.
Built-in Loudspeaker & Headphones
For monitoring, you can use either the
built-in 1W loudspeaker, or headphones
on a 6.35mm unbalanced stereo jack. The
headphone output provides 150mW into 32ϲϬϬɏŚĞĂĚƉŚŽŶĞƐ͘tŚĞŶƚŚĞŚĞĂĚƉŚŽŶĞƐ
are used, the speaker is automatically
muted. The monitor speaker can be muted
via a remote contact on the 9 pin D-type
External Input connector, e.g., when used in
an area with live microphones. Both monitor
and headphone levels are fully adjustable
between -60dB and 9dB using the front panel
volume control.
241
Talkback & Commentary - TB-6D & TB-6R
Specification For Talkback Intercoms
Audio & Power Specification
Talkback Input/Outputs: 15 way male D-type plug (TB-6D
& TB-6R) for 6 balanced talkback
outputs and 1 balanced talkback
input
Talkback Input
Gain Range:
9dB to -28dB adjusted by rear
panel preset pot
External Inputs & Mute:
9 way female D-type socket, for stereo
balanced external input 1, stereo
unbalanced external input 2 (or external
talkback input/output), talkback control
output and mute input
External Input 2:
Dual phono socket, stereo
unbalanced in parallel with the 9 way
female D-type socket
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/ŵƉĞĚĂŶĐĞ͗
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хϰϬŬɏďĂůĂŶĐĞĚ
džƚĞƌŶĂůdĂůŬďĂĐŬ
Input Impedance:
хϯϬŬɏƵŶďĂůĂŶĐĞĚ
džƚĞƌŶĂůdĂůŬďĂĐŬ
Output Impedance:
фϴϬɏƵŶďĂůĂŶĐĞĚ
External Microphone Input:3 pin female XLR socket
TB-6R Front View.
TB-6R Front View.
DŝĐ/ŶƉƵƚ/ŵƉĞĚĂŶĐĞ͗
ϭ͘ϱŬɏďĂůĂŶĐĞĚ
Microphone Input
Gain Range:
74dB to 53dB adjusted by rear
panel preset pot
Headphone Output:
6.35mm (1/4”) jack, unbalanced
stereo capable of driving 150mW
ŝŶƚŽϯϮͲϲϬϬɏŚĞĂĚƉŚŽŶĞƐ
Headphone Output
Gain Range:
9dB to <-60dB (off) adjusted by
front panel pot
Loudspeaker Power:
1W
Mains Input Power:
85V - 264V AC, 47-63Hz, max 10W
Fuse Holder:
1A anti-surge fuse
Equipment Type
TB-6D:
TB-6R:
6 way talkback intercom, desk units
6 way talkback intercom, 19” rackmount
Physical Specification
TB-6R Rear View.
242
Dimensions
(TB-6D Raw):
22.5cm (W) x 22.4cm(D) x 9.1cm(H)
8.8” (W) x 8.8” (D) x 3.6” (H)
Dimensions
(TB-6D Boxed):
33.0cm (W) x 26.0cm (D) x 15.0cm (H)
13” (W) x 10” (D) x 6” (H)
Dimensions
(TB-6R Raw):
48.3cm (W) x 18.5cm (D) x 8.9cm (H)(2U)
19” (W) x 7.3” (D) x 3.5” (H) (2U)
Dimensions
(TB-6R Boxed):
52.0cm (W) x 30.5cm (D) x 16.5cm (H)
20.5” (W) x 12” (D) x 6.5” (H)
Weight (TB-6D):
Nett: 1.25kg Gross: 2.5kg
Nett: 2.75lbs Gross: 5.5lbs
Weight (TB-6R):
Nett: 2.6kg Gross: 4.5kg
Nett: 5.7lbs Gross: 9.9lbs
Talkback & Commentary - TB-6D & TB-6R
Radio Studio Application Using
TB-6D & TB-6R
A typical application would be to
distribute talkback and audio feeds
around a radio or TV studio complex, as
in this diagram.
RECEPTION-AREA
Talkback
EXT 1
MAIN ON-AIR STUDIO 1
Talkback
EXT 1
TB-6D or
TB-6D
TB-6R
TALKBACK
MONITOR
LEVEL
EXT 1
6
TB-6R
NEWS AREA
Portable MD/
Flashcard
Recorder/Player
Talkback
EXT 1
The EXT 2 input could be used in the
news or sales area for monitoring local
sources such as a PC audio output.
PRODUCTION STUDIO/ON-AIR
STUDIO 2
Talkback
EXT 1
TB-6D or
TB-6R
TALKBACK
Linking the studios in this way allows all
locations to talk to each other together
with the ability to monitor 2 external
feeds, with independent control of
monitor muting and headphone/speaker
levels.
MIC
INPUT
INTERCOM
5
3
2
1
The TB-6D and TB-6R units are daisychained together from location to
location and the radio station output
from an off-air receiver is distributed on
input EXT 1 of each unit.
EXT 2
4
HEADPHONE
MONITOR
LEVEL
EXT 2
EXT 1
EXT 2
MIC
INPUT
INTERCOM
6
5
4
HEADPHONE
3
2
1
TB-6R
TB-6D
SALES AREA
CENTRAL TECHNICAL AREA
Talkback
TB-6R
TALKBACK
PC Soundcard
EXT 1
EXT 2
MONITOR
LEVEL
EXT 1
EXT 2
INTERCOM
6
5
4
HEADPHONE
MIC
INPUT
TB-6R
3
1
2
Off-Air
Receiver
Talkback
EXT 1
EXT 2
Outside
Sourcing,
eg. codec
TB-6D
243
244
Telephony & Communications
Telephone Hybrids
Telephone hybrids, or telephone balance units (TBUs) provide the interface between
professional audio equipment and the public telephone network.
247
Line Powered Hybrids
`
CM-TBU Line Powered Telephone Balance Unit
CM-TLL Line Powered Telephone Line Listen Unit
248
248
249
Analogue Telephone Hybrids
HY-03, HY-03S & HY-03T Analogue Telephone Hybrids
250
250
Digital Telephone Hybrids
DHY-03, DHY-03S & DHY-03T Digital Telephone Hybrids
DHY-03EC Automatic Digital Telephone Balance Unit Eurocard
DHY-04, DHY-04S & DHY-04T Automatic Digital TBU, AES/EBU & Analogue I/O
With Ethernet
252
252
255
Digital GSM/2G Hybrids
DHY-04G, DHY-04GS & DHY-04GT Automatic GSM Hybrid, AES/EBU & Analogue I/O
With Ethernet
258
256
258
HD Voice Hybrids
DHY-04HD, DHY-04HDS & DHY-04HDT Automatic HD Voice & GSM Hybrid, AES/EBU
& Analogue I/O With Ethernet
260
Remote Control
Sonifex Switchboard Software - Hybrid Remote Control
263
263
260
IP Streamers
The Pro Audio Streamers are a range of three IP to audio and audio to IP streamers
which have professional analogue and digital inputs and outputs.
Pro Audio Streamers
PS-SEND Audio to IP Streaming Encoder
PS-PLAY IP to Audio Streaming Decoder
PS-AMP IP to Speakers Streaming Decoder
265
266
268
270
245
246
Telephony & Communications - Telephone Hybrids
What Is A Telephone Hybrid?
Telephone hybrids, or telephone balance
units (TBUs) provide the interface between
professional audio equipment and the
public telephone network. They provide
protection for your equipment and the
public telephone lines, allowing for
varying line signals and line conditions.
Automatically cancelling out the unwanted
signal they also facilitate two-way
communication down a single telephone
line.
• The DHY-04 DSP based telephone hybrid.
Each telephone hybrid has a telephone
• The DHY-04G DSP based GSM hybrid.
line connection, a handset connection and
separate terminals for audio input and output • The DHY-04HD DSP based HD Voice &
GSM hybrid.
from a broadcast mixer, or other professional
audio source.
The extremely compact CM-TBU and
CM-TLL units are portable and powered
A large proportion of Sonifex hybrids are
from the telephone line, providing a basic
used in radio and television broadcasting
voice interface to a 4-wire circuit with
applications for allowing external callers to
separate level control of send and receive
be connected to the studio mixing console.
signals, useful for talkback applications.
Most of the other units are supplied to
communication operations for allowing
The analogue HY-03 hybrid is suitable for
extremely effective conversion between
most general telephony applications and is
4-wire audio circuits and standard telephone
often used in radio and TV stations, trading
lines.
floors and conferencing centres. The HY-03
can be used for any application where a
A ringing detector can be used when you
clean telephone signal is required and the
need to answer a call automatically, for
line is not subject to signal delay.
instance, if a journalist files a report to a
PC recorder over a telephone line, the call
The DHY-03 offers near perfect
can be picked up after a set number of rings
performance, using DSP power to
by the ringing detector. Both the HY-03 &
dramatically improve the unit’s operation.
DHY-03 have a built in ringing detector that is
The DHY-03 offers the features of the HY-03,
enabled by one of the configuration settings
but has some other benefits:
switches on the rear panel.
• Echo cancellation is possible and
Line Powered, Analogue, Digital or
distortion of other mixed signals is greatly
GSM or HD Voice?
improved.
Sonifex offer a few different hybrid units:
• Digital hybrids are more tolerant to
fluctuating line conditions and are
• The CM-TBU & CM-TLL line powered
especially suitable for dealing with
hybrids.
calls that have a slight signal delay, for
• The HY-03 analogue telephone hybrid.
example, satellite and conference calls.
• The DHY-03 DSP based telephone hybrid.
The
DHY-03 can recall signals from its memory
buffer and allow for these delays without
impairing performance.
The DHY-04 is a redesign of the DHY-03
hybrid and adds Ethernet connectivity with
a built-in web browser for configuration,
control and dialling, combined AES/EBU
digital and analogue audio inputs and front
panel speed-dial buttons.
The DHY-04G uses a GSM SIM card to
connect callers on the 2G/GSM cellular
network to a radio/TV mixing console
connected to the DHY-04G 4 wire input and
output.
The DHY-04HD uses a 3G SIM card to
connect callers on the 3G cellular networks
using HD Voice wideband audio to provide
a high quality audio call. This is ideal for
broadcasters and journalists providing a
frequency response up to 7kHz (twice that
of a normal GSM or POTS connection).
It converts 3G or GSM calls to the 4 wire
audio signal to and from a connected mixing
console.
Note: The CM-TBU, CM-TLL, HY-03 , DHY-03
and DHY-04 products are operating on
analogue telephone lines, not ISDN or IP
digital lines. The “analogue” and “digital”
refer to the processing used in the units.
247
Telephony & Communications - Line Powered Hybrids
•
CM-TBU Line Powered Telephone
Balance Unit
The CM-TBU line-powered telephone
balance unit is compatible with all
analogue direct exchange lines and
provides a 4-wire communications system
to interface with the telephone network.
The high degree of separation between send
and receive signals makes it suitable for
use in telephone IFB (interrupted foldback)
applications and the high drive capacity
at the 4-wire output enables a presenter’s
earpiece to be connected directly to the unit
without an external amplifier.
This extremely compact unit is powered
from the telephone line and provides an
interface to a 4-wire circuit with separate
level control of send and receive signals.
Optimum rejection of the input signal on the
248
4-wire output is achieved in a bridge circuit
by adjusting three elements (NULL, R-BAL and
C-BAL) via potentiometers which simulate
the complex line impedance. This can be
used to compensate for local line variations
or to adapt to the telephone systems of
other countries, where line characteristics
may differ. Optimization of the sidetone
rejection does not involve the use of any test
equipment and can be easily carried out while
the system is in use.
Although the signal level being sent to the
line can be manually adjusted using the ‘SEND
LEVEL’ control over a wide range, the level
control is followed by a limiter that prevents
the telephone line signal level becoming
overloaded or distorted. The limiter drives a
•
Category: Line Powered Telephone
Hybrids.
Product Function: Provides separation
between send and receive signals on an
analogue telephone network and provides
professional level balanced input & output
signals.
Typical Applications: Talkback applications,
e.g. to get cue feed to a remote presenter
from a distant studio, hospital/community
radio talkshows, house of worship remote
listen & contribution to service.
Features:
• Isolated, full-duplex 4-wire interface to
‘LIMIT’ LED to indicate the onset of limiting.
Although the output stage can drive a
presenter’s earpiece in a telephone IFB
application, the ‘RECEIVE LEVEL’ control may
not be accessible to the presenter, who is
normally situated some distance from the
unit. The presenter may then require a local
control of the earpiece signal level. The CMTBU can be used to supply the correct signal
level to a suitable battery powered earpiece
belt-pack unit.
To enable communication between the
4-wire circuit and the telephone network,
once the 4-wire and telephone line cable
connections are made to the unit, the ‘LINE
CONNECT’ switch can be pressed to power
the unit from the line. This is indicated
•
•
•
•
•
•
direct non-digital telephone exchange
lines.
Line powered, requiring no battery or
external power.
Simple optimization of sidetone rejection
with any country’s telephone system.
LEDs indicating ‘Ring’, ‘Line Hold’ and
‘Limit’ conditions.
Input level control with line-sensing
limiter and limit indicator.
High drive output with level control for
direct feed to presenter’s earpiece, etc.
Loop-through RJ11 line sockets provide
universal connection to line and
telephone set.
Connection to the telephone set is
maintained while the unit is in use.
Small, rugged aluminium case.
by the ‘ON’ LED, and can either be done
after an outgoing call has been dialled on
a telephone set connected to the unit, or
to answer an incoming call after the ‘RING’
LED is seen to flash. Note that a telephone
set is not required for incoming calls unless
an audible ring is required. If the sidetone
level at the 4-wire output is found to be
excessive, the outgoing signal level can be
reduced using the ‘SEND LEVEL’ control or
the balance controls can be adjusted to
minimize it.
The unit is supplied with a connector and
cable kit that enables connections to be
made to both UK Telecom or the universal
RJ11 sockets used in most telephone
networks around the world.
Telephony & Communications - Line Powered Hybrids
Specification For CM-TBU
4-Wire Input
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Input Connector:
3 pin XLR female connector
CM-TLL Line Powered Telephone Line Listen Unit
Input Level Range: -12dBU to +4dBU before
limiting when connected to
an average line
4-Wire Output
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Output Connector: 3 pin XLR male connector
Output Level Range: -6dBU to +6dBU, for average
line level
Sidetone Rejection: 30dB to 40dB average,
depending on line
characteristics
2-Wire Off Hook
Voltage:
6V minimum
2 Wire Connectors: RJ11 socket - line
RJ11 socket - handset
Front Panel Operational Controls
Line Connect:
Push button with indicator
Send Level:
Small rotary control
Receive Level:
Large rotary control
Null Balance:
Recessed preset
potentiometer
R Balance:
Recessed preset
potentiometer
C Balance:
Recessed preset
potentiometer
Ring LED:
Indicates incoming ringing
Limit LED:
Indicates limiter active
On LED:
Indicates connection to the
telephone line
Equipment Type
CM-TBU
Line powered telephone
balance unit
Physical Specification
Dimensions
(Raw):
7.7cm (W) x 8.3cm (D) x 4.2cm (H)
3.0” (W) x 3.3” (D) x 1.7” (H)
Dimensions
(Boxed):
22.9cm (W) x 12.7cm (D) x 7.6cm (H)
9.0” (W) x 5.0” (D) x 3.0” (H)
Weight:
Nett: 0.25kg Gross: 0.75kg
Nett: 0.60lbs Gross: 1.7lbs
Category: Line Powered Telephone
Hybrids.
Product Function: Provides a professional
balanced audio output from the PSTN
telephone line.
Typical Applications: Remote caller feed
into seminars, remote cue & feedback
source.
Features:
• Audio interface to direct analogue
exchange lines - receive calls with
electrical isolation from the line.
• Line powered, requiring no battery or
external power.
• LEDs indicating ‘Ring’ and ‘On Hold’
conditions.
• Loop-through line sockets provide
in-line connection with existing
telephone.
• Existing telephone remains connected
when the unit is in use.
• Line connections to British or
International sockets via supplied
cable kit.
• Small, rugged aluminium case with
XLR3 male 4-wire connector.
The CM-TLL provides a low-loss interface
to receive audio from a telephone line. The
line-powered unit is compatible with all
analogue direct exchange lines and includes
an LED indication of incoming calls.
Specification For CM-TLL
4-Wire Output
KƵƚƉƵƚ/ŵƉĞĚĂŶĐĞ͗ ϭϱϬё͕ƚƌĂŶƐĨŽƌŵĞƌĐŽƵƉůĞĚ
Output Connector: 3 pin XLR male connector
Output Level:
Typically 0dBU for average
line level
2-Wire
The unit is intended to be used with an
earpiece amplifier to receive an audio feed
by telephone from a studio.
The line and telephone set are connected
to the unit via the RJ11 connectors in order
to make an outgoing call. A telephone may
not be required if calls are only incoming,
because telephone line ringing is indicated
by a flashing LED built into the unit. In either
case, the ‘LINE CONNECT’ switch is pressed
to make the line connection. The ‘ON’ LED
indicates that the unit is powered and that
audio is routed to the line via the XLR plug.
Off Hook Voltage:
6V minimum
2 Wire Connectors: RJ11 socket - line
RJ11 socket - handset
Front Panel Operational Controls
Line Connect:
Push button with indicator
Ring LED:
Indicates incoming ringing
On LED:
Indicates connection to the
telephone line
Equipment Type
CM-TLL:
Line powered telephone line
listen unit
Physical Specification
Dimensions
(Raw):
7.7cm (W) x 8.3cm (D) x 4.2cm (H)
3.0” (W) x 3.3” (D) x 1.7” (H)
Dimensions
(Boxed):
22.9cm (W) x 12.7cm (D) x 7.6cm (H)
9.0” (W) x 5.0” (D) x 3.0” (H)
Weight:
Nett: 0.20kg Gross: 0.70kg
Nett: 0.50lbs Gross: 1.6lbs
249
Telephony & Communications - Telephone Hybrids HY-03
HY-03, HY-03S & HY-03T Analogue Telephone Hybrids
Features:
• Fully automatic - adapts to varying line
conditions and has automatic signal
limiting.
• Local and remote line hold switching Category: Analogue Telephone Hybrids.
calls can be remotely switched through
Product Function: Provides separation
a mixing console.
between send and receive signals on an
• Momentary or permanent latching
analogue telephone network and provides
remotes can be enabled.
professional level balanced input & output
• ĂůĂŶĐĞĚŵŝĐͬůŝŶĞŝŶƉƵƚͲϭϬŬɏďĂůĂŶĐĞĚ
signals.
input selectable for 0dBu clean feed line,
Typical Applications: Radio & TV station talk
or microphone level with adjustable gain.
shows, telephony interface to the mixer.
• Balanced output - 0dBu low impedance
The analogue HY-03 telephone hybrid sets
the standard as an excellent value, high
quality telephone hybrid.
The analogue HY-03 hybrid is suitable for
most general telephony applications and is
often used in radio and TV stations, trading
floors and conferencing centres.
The HY-03 can be used for any application
where a clean telephone signal is required
and the line is not subject to signal delay.
HY-03 Single Free-Standing Automatic
Analogue TBU - Front & Rear.
250
balanced output, with output gain
adjustment.
• Mixed output - the output can be a mix
of the caller and mic/line input signals for
recording both sides of the telephone
conversation.
• Integrated ring detector - automatic call
answering after a pre-determined
number of rings.
• Fitted with K-break disconnect detection
as standard with an option for dial-tone
disconnect using the optional
HY-03DTD board. The HY-03DTD board
•
•
•
•
can be configured to disconnect on
recognizing the dial tone used in a
specific country.
Line limiter, bandpass filter and output
noise gate with preset threshold
providing low distortion crystal
clear audio.
28dB typical line balance rejection.
Built in power supply with switchable
115V, or 230V, mains input.
BABT approval compliant with European
PTT specifications.
Telephony & Communications - Telephone Hybrids HY-03
Which Format Is Most Suitable ?
The HY-03 analogue hybrids are available
in three models :
Specification For HY-03
Connections
Audio Specification
Mic/Line Input:
Line Output:
XLR 3 pin male
•
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Microphone Mode:
ϮϬϬɏďĂůĂŶĐĞĚ
Telephone Line:
RJ11 6/4 socket
Clean Feed
Limiting Input:
+4dBu
Telephone
RJ11 6/4 socket
Handset/Instrument:
Microphone
Level Range:
From 74dB to 40dB adjusted by
preset pot
Bandwidth to
Telephone Line:
250Hz - 4kHz, -3dB ref 1kHz
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Impedance:
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dĞůĞƉŚŽŶĞ>ŝŶĞ
Impedance Range:
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Order Code
Description
Output Level Range:
+8dB to -14dB adjusted by preset
pot
HY-03DTD
Dial tone detect add-on board
HY-03CON
Front panel conversion kit, HY-03S
19” (48cm) rack-mount
front to HY-03 free standing
HY-03SCON
Front panel conversion kit, HY-03
free standing to HY-03S
19” (48cm) rack-mount front
HY-03TCON
Front panel conversion kit, HY-03 or
HY-03S, to HY-03T
19” (49cm) rack-mount front
•
•
HY-03 Single free-standing automatic
analogue TBU.
HY-03S Single 19” rack mounted
automatic analogue TBU with
ringing detector.
HY-03T Twin 19” rack mounted
automatic analogue TBU.
HY-03S Single Rackmount Automatic Analogue TBU -- Front & Rear.
HY-03T Twin Rackmount Automatic Analogue TBU -- Front & Rear.
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ϭϬŬɏďĂůĂŶĐĞĚϬĚ
Line Mode (Clean Feed):
XLR 3 pin female with push button
mic/line selection
Remotes:
9-way D-type socket
Power:
IEC mains (CEE22)
Each unit is supplied with:
1 x RJ11 to RJ11 telephone line lead
1 x RJ11 to BT plug telephone line lead
1 x BT handset socket to RJ11 plug adapter,
1 x IEC mains lead fitted with moulded mains plug and
1 x handbook.
Accessories
Rejection Ratio:
45dB on 1kHz tone, typically 28dB
on complex waveforms,
reference peak level of 0dB
Ring Detector
Sensitivity:
1 ring to 6 rings
Power:
230V 50Hz, or 115V 60Hz.
6W for HY-03.
Order Code
Description
Height
Width
Depth*
Weight
Total Nett
Weight
Total Gross
HY-03
Automatic analogue
TBU, free standing
4.5cm
1.8”
21.8cm
8.6”
17.5cm
7”
1.25kg
2.75lbs
2.0kg
4.4lbs
HY-03S
Automatic analogue
TBU, 19” rack mounted
4.5cm (1U)
48.3cm
1.8” (1U) (19” rack width)
17.5cm
7”
1.30kg
2.9lbs
2.1kg
4.6lbs
HY-03T
Twin automatic analogue
TBU, 19” rack mounted
4.5cm (1U)
48.3cm
1.8” (1U) (19” rack width)
17.5cm
7”
2.60kg
5.7lbs
4.0kg
8.8lbs
*Depth is measured from the front to the end of the connectors fitted to the back of the unit. Note: If you are ordering the
HY-03 for use in the USA, add the word “US” after the product code. The HY-03 uses different circuitry for US telephone
exchanges.
251
Telephony & Communications - Telephone Hybrids DHY-03
DHY-03, DHY-03S & DHY-03T Digital Telephone Hybrids
Category: Digital Telephone Hybrids.
Product Function: Provides separation
between send and receive signals on an
analogue telephone network, provides
professional level balanced input &
output signals and has echo cancellation.
Typical Applications: Radio & TV station
talk shows, telephony interface to the
mixer.
Features:
• Fully automatic - adapts to varying
line conditions and has automatic
signal limiting.
• Fully adaptive echo cancellation to
127msec - default is 24msec.
• 76dB typical line balance rejection
offering superb performance and
crystal clear audio.
• Front panel input and output gain
controls.
• Front panel LED metering of receive
and send signals.
• Built-in conferencing for 2 hybrids, so
that a single telco channel on a mixing
desk can receive 2 calls.
252
• Integrated ring detector - automatic
call answering after a pre-determined
number of rings.
• Automatic call disconnection. Fitted with
K-break, line polarity reversal and dial
tone disconnect detection, defined by the
country selection.
• Automatic ducking facility allows the
talent to ‘shout-down’, or talk over, a
caller by reducing the gain of the caller’s
signal if it goes above a certain level.
• Local and remote line hold switching calls can be remotely switched through a
mixing console.
• Line hold/release button to control line
hold circuit, illuminates to indicate the
status of the line and flashes to show
ring status.
• DTMF tone recognition allowing a optoisolated GPI output to be made on
receipt of selected DTMF tones, e.g. for
starting a studio automation recorder
automatically to record a remote
telephone interview.
• International operation with built-in
configurable settings for each country.
Country selection allows the unit to
provide line impedance and a simulation
circuit to match the country.
• RS232 serial port for remote control of
the TBU, DTMF tone dialling and firmware
•
•
•
•
•
•
•
•
upgrades to add new country settings.
Remote port distributes the remote line
connect switch and tally output, a
momentary/latch selector and the
DTMF detect output.
The remote line connect switch can be
either momentary or latching in its
action.
Balanced mic/line input - 10k balanced
input selectable for 0dBu clean feed
line, or microphone level with
adjustable gain.
Balanced output - 0dBu low impedance
balanced output, with output gain
settings.
Record output - the conferencing
output can be set via a jumper to give
a mix of the caller and mic/line input
signals for recording both sides of the
telephone conversation.
Line limiter, bandpass filter and output
noise gate with preset threshold
providing low distortion audio.
Built in universal power supply between
90V AC and 250V AC, 47-63Hz, IEC
mains input.
ETSI approval compliant with European
PTT specifications.
The digital DHY-03 telephone hybrid is
probably the best performing digital hybrid
in the world, with simply stunning line
balance rejection figures. For the best
sounding audio calls you’re likely to hear,
you should specify the DHY-03.
The DHY-03 offers near perfect
performance, using DSP power to
dramatically improve the unit’s operation.
The DHY-03 offers the features of the
HY-03, but has some other benefits:
Echo cancellation is possible and distortion
of other mixed signals is greatly improved.
Digital hybrids are more tolerant to
fluctuating line conditions and are especially
suitable for dealing with calls that have a
slight signal delay, for example, satellite
and conference calls. The DHY-03 can
recall signals from its memory buffer and
allow for these delays without impairing
performance.
Telephony & Communications - Telephone Hybrids DHY-03
Which Format Is Most Suitable ?
The DHY-03 digital hybrids are available
in three models :
• DHY-03 Single free-standing automatic
digital TBU.
• DHY-03S Single 19” rack mounted
automatic digital TBU.
• DHY-03T Twin 19” rack mounted
automatic digital TBU.
DHY-03 Single Free Standing Automatic Digital TBU - Front & Rear.
DHY-03S Single Rackmount Automatic
Digital TBU - Front & Rear.
DHY-03T Twin Rackmount Automatic
Digital TBU - Front & Rear.
253
Telephony & Communications - Telephone Hybrids DHY-03
DHY-03S Single Rackmount Automatic Digital TBU - Front.
Specification For DHY-03
Audio Specification
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ϭϬŬɏďĂůĂŶĐĞĚϬĚ Line Mode (Clean Feed):
Ring Detector
Sensitivity:
Off, 2, 4 or 6 rings
WŽǁĞƌƚŽ,zͲϬϯ͕
S & T:
Power to DHY-03EC:
Description
hŶŝǀĞƌƐĂůϭϮɏƉŽǁĞƌƐƵƉƉůLJ͗ϵϬƚŽ
250V AC; 47-63Hz; fused 1A
DHY-03CON
Front panel conversion kit, DHY-03S to
DHY-03
±15 V DC @ 160mA per rail or
regulated +5V DC @ 600mA
DHY-03SCON
Front panel conversion kit, DHY-03 free
standing to DHY-03S 19” (48cm) rack-mount
front
DHY-03TCON
Front panel conversion kit, DHY-03 or
DHY-03S, to DHY-03T 19” (48cm)
rack-mount front
DHY-03CONF
Conference cable to connect 2 x DHY-03
units
DHY-03RLY
Latching handset relay option for DHY-03, S
and T (included in current models).
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DHY-03EC)
/ŶƉƵƚ/ŵƉĞĚĂŶĐĞͲ
Microphone Mode:
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Input Level
Gain Range:
+6dB, 0dB, and -6dB adjusted by
3-position front panel switch, +10dB
jumper
Mic/Line Input:
Microphone Level
Gain Preset:
From 65dB to 35dB
(not for DHY-03EC)
XLR 3 pin female, with push-button
mic/line selection
Line Output:
XLR 3 pin male
Maximum Input
Levels:
Line +26dBu, mic -24dBu
Telephone Line:
RJ11 6/4 socket
Clean Feed
Limiting Input:
-4dBu for CTR21 setting, other
values available *
Bandwidth to
Telephone Line:
250Hz - 4kHz, -3dB ref 1kHz
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Impedance:
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impedance circuits *
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Conference/Record:
Output Level
Gain Range:
+6db, 0dB, and -6dB adjusted by
3-position front panel switch
Rejection Ratio:
76dB on tones or complex
waveforms, reference peak level of
0dB
SCi Dialpad Home Page.
Order Code
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Conferencing:
KƵƚƉƵƚ/ŵƉĞĚĂŶĐĞͲ ϱϬɏďĂůĂŶĐĞĚĨůŽĂƚŝŶŐϬĚƵ
Line Out:
Accessories
* These values are dependent on the actual country setting
selected on the DHY-03
Connections
Telephone
RJ11 6/4 socket
Handset/Instrument:
SCi Disconnect Method Settings Page.
Physical Specification
Conferencing or
Record Audio:
RJ45 socket
Order Code
Description
Height
Width
Depth*
Weight
Total Nett
Weight
Total Gross
Remotes:
9-way D-type socket
DHY-03
RS232 Serial:
9-way D-type socket
Automatic digital
TBU, free standing
4.5cm
1.8”
21.8cm
8.6”
17.5cm
6.9”
1.4kg
3lbs
2.2kg
4.8lbs
DHY-03S
Automatic digital
4.5cm (1U)
48.3cm
17.5cm
1.45kg
2.3kg
TBU, 19” (48cm)
rack mounted
1.8” (1U) (19” rack width)
6.9”
3.2lbs
5lbs
Twin automatic digital
4.5cm (1U)
48.3cm
17.5cm
2.80kg
4.4kg
TBU, 19” (48cm)
rack mounted
1.8”
(19” rack width)
6.9”
6.2lbs
9.7lbs
Automatic digital TBU
with ringing detector
Eurocard model
(PCB 10x16cm)
12.9cm (3U) 4.0cm (8E)
5” (3U)
1.6” (8E)
19.0cm
7.5”
150g
0.3lbs
500g
1.1lbs
Power:
IEC mains (CEE22)
Connections for
Eurocard:
64 pin DIN 41612 male (plug)
Each DHY-03, S & T unit is supplied with:
1 x RJ11 to RJ11 telephone line lead
1 x RJ11 to BT plug telephone line lead
1 x BT handset socket to RJ11 plug adapter
1 x IEC mains lead fitted with moulded mains plug
1 x handbook and warranty card.
DHY-03T
DHY-03EC
*Depth is measured from the front to the end of the connectors fitted to the back of the unit.
254
Telephony & Communications - Telephone Hybrids DHY-03
DHY-03EC Automatic Digital Telephone Balance Unit Eurocard
Category: Digital Telephone Hybrids.
Product Function: Provides separation
between send and receive signals on an
analogue telephone network, provides
professional level balanced input &
output signals and has echo cancellation.
Typical Applications: Radio & TV station
talk shows, telephony interface to the
mixer.
Features:
• Eurocard format to get many cards
into a small rackspace.
• Fully automatic - adapts to varying
line conditions and has automatic
signal limiting.
• Fully adaptive echo cancellation to
127msec - default is 24msec.
• 76dB typical line balance rejection
offering superb performance and
crystal clear audio.
• Front panel input and output gain
controls.
• Integrated ring detector - automatic
call answering after a pre-determined
number of rings.
• Automatic call disconnection.
The DHY-03EC eurocard single digital
telephone hybrid uses the same
technology as the DHY-03 but is based
in a card-style format for installation in
a eurocard rack frame, or in certain
broadcast mixers. Eurocards are supplied
without a power supply, or casing, and are
therefore significantly cheaper than the
other units. It is pin compatible with the
older DHY-02EC eurocard, but has the
outstanding performance and most
of the features of the DHY-03, with a
few differences:
• The analogue input is line level only,
though the 10dB professional/ consumer
jumper is retained.
• There is no conferencing facility and
consequently no record output option.
• The level meters are 2 tricolour LEDs.
• The level switches are now onboard 3
way jumpers.
• The remote outputs are connected
via slide switches which means that the
output signal can be either +5V, +15V or
pull down to ground.
• The handset is connected to the
telephone line via a divert relay.
DHY-03EC - Automatic Digital
Telephone Balance Unit Eurocard.
255
Telephony & Communications - Telephone Hybrids DHY-04
DHY-04 Single Automatic Digital TBU, AES/EBU & Analogue I/O With Ethernet
•
Category: Digital Telephone Hybrids.
Product Function: Provides separation
between send and receive signals on an
analogue telephone network, provides
professional level balanced input &
output signals and has echo cancellation.
Typical Applications: Radio & TV station
talk shows, telephony interface to the
mixer.
Features:
• Fully automatic - adapts to varying line
conditions and has automatic signal
limiting.
• Fully adaptive echo cancellation to
127msec - default is 24msec.
• 70dB typical line balance rejection
offering superb performance and
crystal clear audio.
• Front panel input and output gain
controls.
• Front panel LED metering of receive
and send signals.
• Built-in conferencing for 2 hybrids, so
that a single telco channel on a mixing
desk can receive 2 calls.
• Integrated ring detector - automatic
256
•
•
•
•
•
•
•
•
call answering after a pre-determined
number of rings.
Automatic call disconnection. Fitted
with K-break, line polarity reversal and
dial tone disconnect detection, defined
by the country selection.
Automatic ducking facility allows the
talent to ‘shout-down’, or talk over, a
caller by reducing the gain of the caller’s
signal if it goes above a certain level.
Local and remote line hold switching calls can be remotely switched through
a mixing console.
Line hold/release button to control line
hold circuit, illuminates to indicate the
status of the line and flashes to show
ring status.
DTMF tone recognition allowing a
opto-isolated GPI output to be made on
receipt of selected DTMF tones, e.g. for
starting a studio automation recorder
automatically to record a remote
telephone interview.
International operation with built-in
configurable settings for each country.
Country selection allows the unit
to provide line impedance and a
simulation circuit to match the country.
RS232 serial port for remote control of
the TBU & DTMF tone dialling.
Remote port distributes the remote
•
•
•
•
line connect switch and tally output, a
momentary/latch selector and the DTMF
detect output.
The remote line connect switch can be
either momentary or latching in its action.
Balanced mic/line input - 10k balanced
input selectable for 0dBu clean feed line,
or microphone level with adjustable gain.
Balanced output - 0dBu low impedance
balanced output, with output gain
settings.
Record output - the conferencing output
The DHY-04 Front & Rear Panels.
can be set via a jumper to give a mix of
the caller and mic/line input signals for
recording both sides of the telephone
conversation.
• Line limiter, bandpass filter and output
noise gate with preset threshold
providing low distortion audio.
• Built in universal power supply between
90V AC and 250V AC, 47-63Hz, IEC mains
input.
• ETSI approval compliant with European
PTT specifications.
Telephony & Communications - Telephone Hybrids DHY-04
The Best Telephone Hybrid in the World Just Got Better!
The DHY-04 telephone hybrid is an
enhanced redesign of the DHY-03, the best
performing telephone hybrid in the world.
It now has auto-sensing combined analogue
and AES/EBU inputs and outputs, front
panel speed dial buttons, together with an
Ethernet interface to allow web browser
access to the configuration and internal
settings. All whilst still retaining stunning
line balance rejection figures. For the best
sounding audio calls you’re likely to hear,
you should specify the DHY-04. Key new
features of the unit include:
• Auto-sensing combined analogue or AES/
EBU XLR input.
• AES/EBU sample rates up to 24 bit/96kHz
accepted.
• Configurable analogue or AES/EBU XLR
output.
• Ethernet port for remote configuration via
web browser GUI.
• Remote dialling and line hold control via
Ethernet.
• Ethernet network interface can generate
SNMP Traps for SNMP management
systems.
• DTMF dial tone recognition for reporter
remote access - a journalist can dial into
the unit which can recognise a preprogrammed DTMF numeric password to
automatically connect the journalist on-air.
• Four front panel speed-dial buttons for
dialling internally preset phone numbers.
• Front panel Redial button for redialling the
last number.
The DHY-04 Front Panel.
The DHY-04 Rear Panel.
The DHY-04S Front Panel.
The DHY-04S Rear Panel.
The DHY-04T Front Panel.
The DHY-04T Rear Panel.
257
Telephony & Communications - Telephone Hybrids DHY-04G
DHY-04G Single Automatic GSM Hybrid, AES/EBU & Analogue I/O With Ethernet
Category: Digital Telephone Hybrids.
Product Function: Provides separation
between send and receive signals on a
2G/GSM network, provides professional
level balanced input & output signals
and has echo cancellation.
Typical Applications: Radio & TV station
outside broadcast vehicles for talk
shows, telephony interface to the mixer.
Backup hybrid to cover failure of main
landline.
Features:
• Quad-Band EGSM 850 / 900 / 1800
/1900MHz.
• Rear panel 2G/GSM SIM card
insertion.
• Ethernet web server control and
configuration.
• Front panel speed dial buttons with
redial.
• Signal strength LED display.
• LEDs for SIM enabled and GSM
network availability.
• Automatic operation.
• Combined AES/EBU and analogue
input and output.
258
A new addition to the DHY-04 range is the
ability for the DHY-04G version to be used
on a GSM cellular (mobile) phone network
instead of a telephone (POTS) line. The
DHY-04G can accept a SIM card in the rear
panel slot and by connecting a suitable
GSM antenna, the DHY-04G can receive
and make high quality broadcast calls over
the cellular network, converting the GSM
call to the 4 wire audio signal to and from
a connected mixing console. The GSM
module used in the DHY-04G is quad-band
GSM, so it can take and make calls on any
2G network.
Ideal for studios in remote locations, for
OB vans and trucks on the move, and in
emergency situations where a telephone
landline can’t be guaranteed, the DHY-04G
offers outstanding performance.
The DHY-04G Front & Rear Panels.
The DHY-04G has all features of the DHY-04
(read cellphone/mobile features instead of
telephony features in the listed bullet point
specification) together with some additional
front panel indicators. There are two LEDs,
one for SIM enabled and one for GSM
Network availability. Additionally there is a
push button which allows the GSM signal
level to be displayed on the meter LEDs.
DHY-04G Home Page.
DHY-04G Configuration Page.
Telephony & Communications - Telephone Hybrids DHY-04G
Which Format Is Most Suitable ?
The DHY-04G GSM hybrids are available
in three models :
• DHY-04G Single free-standing GSM TBU.
• DHY-04GS Single 19” rack mounted GSM
TBU.
• DHY-04GT Twin 19” rack mounted GSM
TBU.
The DHY-04G Front Panel.
The DHY-04G Rear Panel.
The DHY-04GS Front Panel.
The DHY-04GS Rear Panel.
The DHY-04GT Front Panel.
The DHY-04GT Rear Panel.
259
Telephony & Communications - Telephone Hybrids DHY-04HD
DHY-04HD Single Automatic HD Voice & GSM Hybrid, AES/EBU & Analogue I/O With Ethernet
Category: Digital Telephone Hybrids.
Product Function: Provides separation
between send and receive signals on a
2G/GSM network, provides professional
level balanced input & output signals
and has echo cancellation.
Typical Applications: Radio & TV station
outside broadcast vehicles for talk
shows, telephony interface to the mixer.
Backup hybrid to cover failure of main
landline.
Features:
• Five band UMTS/HSPA+850 / 900 /
1800 / 1900 / 2100MHz.
• Rear panel 2G GSM or 3G SIM card
insertion.
• Ethernet web server control and
configuration.
• Front panel speed-dial buttons with
redial.
• Signal strength LED display.
• LEDs for SIM enabled and GSM
network availability.
• Automatic operation.
• Combined AES/EBU and analogue
input and output.
The DHY-04HD Front & Rear Panels.
The DHY-04HD HD Voice Hybrid is used on a 3G or GSM cellular (mobile) phone network
instead of a telephone (POTS) line. The DHY-04HD can accept a SIM card in the rear panel
slot and by connecting a suitable GSM antenna, it can receive and make high quality
broadcast calls over the cellular network, converting the 3G or GSM call to the 4 wire
audio signal to and from a connected mixing console. The module used in the DHY-04HD
is quad-band GSM and 5 band UMTS/HSPA+, so it can take and make calls on any 2G GSM,
or 3G network.
It is ideal for studios in remote locations,
for OB vans and trucks on the move, and in
emergency situations where a telephone
landline can’t be guaranteed, the DHY-04HD
offers outstanding performance.
260
The DHY-04HD has all features of the
DHY-04 together with some additional
front panel indicators. There are two
LEDs, one for SIM enabled and one for
GSM Network availability. Additionally
there is a push button which allows the
mobile signal level to be displayed on the
meter LEDs.
HD Voice:
HD Voice uses a coding system (also
known as WB-AMR) for audio data that
provides a significant enhancement on
the quality of cellular phone calls. It is
ideal for broadcasters and journalists
providing a frequency response up to
7kHz (twice that of a normal GSM or
POTS connection).
The use of HD Voice is dependent on 3
criteria:
1. The method has to be supported by
the network, (and this may be limited
by your contract), and when different
networks are involved in the call, the
interoperability between the networks.
2. The actual equipment used by both
ends of the call must be HD Voice
compatible. Both these requirements will
be established on call connection, which
leads to the third criteria.
Telephony & Communications - Telephone Hybrids DHY-04HD
Which Format Is Most Suitable ?
The DHY-04HD HD Voice hybrids are
available in three models :
• DHY-04HD Single free-standing HD Voice
Hybrid.
• DHY-04HDS Single 19” rack mounted HD
Voice Hybrid.
• DHY-04HDT Twin 19” rack mounted HD
Voice Hybrid.
The DHY-04HD Front Panel.
The DHY-04HD Rear Panel.
3. The signal quality can vary during the
call. Normally the hybrid will be used
in a fixed location (even OB trucks are
normally stationary), so the position of
the antenna can be refined for best signal
quality. However the far end may be from
a cellular phone, so may vary in quality
and which can lead to dynamic bandwidth
changes during a call. It is mostly true that
network providers will only handle HD
Voice on 3G networks though, in theory, it
should aldo be compatible with enhanced
2G networks.
The DHY-04HDS Front Panel.
The DHY-04HDS Rear Panel.
The DHY-04HDT Front Panel.
The DHY-04HDT Rear Panel.
261
Telephony & Communications - Telephone Hybrids
Specification For DHY-04, DHY-04G & DHY-04HD
Audio Specification Analogue Audio I/O
Connections
DHY-04 DHY-04G DHY-04HD
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Input Level Gain Range:
+6dB, 0dB, and -6dB adjusted by 3-position
front panel switch, +10dB jumper
Yes
Yes
Microphone Level Gain Preset:
From 65dB to 35dB
Yes
Yes
Maximum Input Levels:
Line +26dBu, mic -24dBu
Yes
Clean Feed Limiting Input:
-4dBu for CTR21 setting, other values available * Yes
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zĞƐ
Output Level Gain Range:
+6db, 0dB, and -6dB adjusted by 3-position
front panel switch
Yes
Yes
Yes
DHY-04 DHY-04G DHY-04HD
Mic/Line/AES-EBU Input:
XLR 3 pin female, with push-button mic/line
selection
Yes
Yes
Yes
Line/AES-EBU Output:
XLR 3 pin male
Yes
Yes
Yes
Telephone Line:
RJ11 6/4 socket
Yes
No
No
Telephone Handset/Instrument:
RJ11 6/4 socket
Yes
No
No
Yes
GSM Antenna:
SMA socket
No
Yes
Yes
Yes
Conferencing or Record Audio:
RJ45 socket
Yes
Yes
Yes
Yes
Yes
Remotes:
9-way D-type socket
Yes
Yes
Yes
Yes
Yes
Ethernet:
RJ45 socket
Yes
Yes
Yes
zĞƐ
zĞƐ
RS232 Serial:
9-way D-type socket
Yes
Yes
Yes
zĞƐ
zĞƐ
Power:
IEC mains (CEE22)
Yes
Yes
Yes
Accessories Order Code
Audio Specification Digital Audio I/O
Description
DHY-04CON
Front Panel Conversion Kit, DHY-04S to DHY-04
DHY-04SCON
Front panel conversion kit, DHY-04 free standing to DHY-04S 19” (48cm) rack-mount front
Front panel conversion kit, DHY-04 or DHY-04S, to DHY-04T 19” (48cm) rack-mount front
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zĞƐ
zĞƐ
DHY-04TCON
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DHY-04GCON
Front Panel Conversion Kit, DHY-04GS to DHY-04G
Sample Frequency Range:
30 - 100kHz (i.e. including 32kHz, 44.1kHz,
48kHz, 64kHz, 88.2kHz & 96kHz)
Yes
Yes
Yes
DHY-04GSCON
Front panel conversion kit, DHY-04G free standing to DHY-04GS 19” (48cm) rack-mount front
Signal Level:
2V/7V peak to peak min/max
Yes
Yes
Yes
DHY-04GTCON
Front panel conversion kit, DHY-04G or DHY-04GS, to DHY-04GT 19” (48cm) rack-mount front
Analogue Input Level for Full Scale Digits:
+18dBU
Yes
Yes
Yes
Maximum Input Level:
0dBFS but internally limited to -6dBFS
Yes
Yes
Yes
Maximum Output Level:
-6dBSFS
Yes
Yes
Yes
Bandwidth to Telephone Line:
250Hz - 4kHz, -3dB ref 1kHz
Yes
No
No
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ϲϬϬɏ͕ϵϬϬɏƉůƵƐϭϰŽƚŚĞƌĐŽŵƉůĞdž
impedance circuits *
Yes
No
No
80-88dB on complex waveforms, reference
peak level of 0dBFS
Yes
Yes
Yes
Off, 1, 2, 3, or 4 rings
Yes
Yes
Module Type:
Quad-Band EGSM 850 / 900 / 1800 /1900MHz
5-band UMTS/HSPA+ 850 / 900 /1800 / 1900 /
2100MHz
No
No
No
Output Power:
Class 4 (2W) @ 850 / 900MHz,
Class 1 (1W) @ 1800 / 1900MHz
Class 3 (0.25W 24dBm) @ UMTS
Class E2 (0.5W 27dBm) @ EDGE 850/900
Class E2 (0.4W 26dBm) @ EDGE 1800/1900
Sensitivity:
Approvals:
Telephone Line
Rejection Ratio:
Ring Detector Sensitivity:
DHY-04HDCON
Front Panel Conversion Kit, DHY-04HDS to DHY-04HD
DHY-04HDSCON
Front panel conversion kit, DHY-04HD free standing to DHY-04HDS 19” (48cm) rack-mount front
DHY-04HDTCON
Front panel conversion kit, DHY-04HD or DHY-04HDS, to DHY-04HDT 19” (48cm) rack-mount front
DHY-04CONF
Conference Cable to Connect 2 x DHY-04(G or HD) Units
Physical Specification
Order Code
Description
DHY-04 (Raw):
Automatic digital telephone
hybrid, free standing
DHY-04G (Raw):
Automatic digital GSM hybrid
TBU, free standing
Yes
DHY-04HD (Raw):
Automatic digital HD hybrid
TBU, free standing
Yes
No
No
Yes
Yes
Yes
DHY-04, DHY-04G & DHY-04HD (Boxed):
No
No
No
No
No
Yes
Yes
No
No
No
Yes
Yes
Yes
Yes
Yes
-107 dBm (typ.) @ 850 / 900MHz,
-106 dBm (typ.) @ 1800 / 1900MHz
-109 dBm (typ.) @ GSM 850 /900MHz
-110 dBm (typ.) @ DCS1800 /PCS1900MHz
-111 dBm (typ.) @ UMTS
No
No
No
No
No
Yes
Yes
No
No
No
No
No
Yes
Yes
Yes
Fully Type approved conforming with R&TTE,
European - CE, GCF, North America - FCC,
PTCRB, IC, Brazil - ANATEL
No
Yes
Yes
GSM Connection
Power Supply
Power to DHY-04, S & T
Universal 12W power supply: 90 to 250V AC;
47-63Hz; fused 1A
* These values are dependent on the actual country setting selected on the DHY-04
Yes
Yes
Yes
DHY-04S (Raw):
Automatic digital telephone
hybrid, rack mounted
DHY-04GS (Raw):
Automatic digital GSM hybrid
TBU, rack mounted
Height
Width
Total
Total
Depth* Nett Weight Gross Weight
4.5cm
1.8”
21.8cm
8.6”
17.5cm
6.9”
6cm
2.4”
34cm
13.4”
27cm
10.6”
4.5cm (1U)
48.3cm
17.5cm
1.8” (1U) (19” rack width) 6.9”
2.2kg
4.8lbs
1.45kg
3.2lbs
2.3kg
5lbs
2.80kg
6.2lbs
4.4kg
9.7lbs
DHY-04HDS (Raw): Automatic digital HD hybrid
TBU, rack mounted
DHY-04S, DHY-04GS & DHY-04HDS (Boxed):
DHY-04T (Raw):
Twin automatic digital
telephone hybrid, rack
mounted
DHY-04GT (Raw):
Twin automatic digital GSM
hybrid TBU, rack mounted
HY-04HDT (Raw):
Twin automatic digital HD
hybrid TBU, rack mounted
DHY-04T, DHY-04GT & DHY-4HDT (Boxed)
6.8cm
2.7”
58.8cm
23”
27cm
10.6”
4.5cm (1U)
48.3cm
17.5cm
1.8” (1U) (19” rack width) 6.9”
6.8cm
2.7”
58.8cm
23”
27cm
10.6”
*Depth is measured from the front to the end of the connectors fitted to the back of the unit.
262
1.4kg
3lbs
Telephony & Communications - Telephone Hybrids
Free Sonifex Switchboard Software - Remote Control of DHY-04/G/HD Hybrids
Software for remote control of the DHY04 range of digital telephone hybrids is
available for download free of charge.
The software runs on Windows 7/8/10
and allows the control of up to 24 digital
hybrids, including the GSM (DHY-04G)
and HD Voice (DHY-04HD) versions of the
products, all from one screen.
The software includes a Phone Book, a
Chat Screen, Speed Dials, a Dial Pad and a
Broadcast Clock.
It uses a local or networked SQL database
for shared access to the hybrids and the
interface is designed for use with a Full HD
resolution touchscreen, although it can be
used with traditional mouse and keyboard
inputs.
To download, please go to the Software
Downloads section of the Sonifex website.
263
264
Telephony & Commentary - Pro Audio Streamers
IP Streamers
Pro Audio Streamers
The Pro Audio Streamers are a range of
three IP to audio and audio to IP streamers
which have professional analogue and
digital inputs and outputs. They allow
audio to be streamed around a building,
wan or lan using IP audio and CAT5 cabling
infrastructure. Typical applications include:
• As a backup STL (studio to transmitter
link).
• For audio confidence monitoring in
remote locations, such as at a radio
transmitter site.
• For distribution of audio and music
around a building, such as for passing
audio to speakers in a conference room.
• As an internet-based IP music distribution
system.
• As a tannoy, paging or IP based public
address system.
• For in-house audio applications and
distribution.
• For streaming internet radio in bars and
clubs.
• For radio and music channels in hotel
rooms distributed via an IP network.
• For playing audio from a PC, a jukebox
application, or from a USB stick.
There are three products in the range
providing encoding/decoding and
streaming:
The PS-SEND converts an audio input to an
IP stream.
The PS-PLAY reads an IP stream and outputs
to balanced and unbalanced audio line
levels.
PS-SENDS Audio to IP Streaming Encoder.
The PS-AMP reads an IP stream and outputs
audio to stereo speakers.
PS-PLAYS IP to Audio Streaming Decoder.
PS-AMPS IP to Speakers Streaming Decoder.
265
Telephony & Commentary - Pro Audio Streamers
PS-SEND Audio to IP Streaming Encoder
based network to one or more PS-PLAY or
PS-AMP units or other proprietary servers
such as those for Icecast or Shoutcast.
Category: Pro Audio Streamers.
Product Function: Audio to IP converter.
Typical Applications: See the front page
of this section for application ideas.
266
Features:
• Audio to IP streamer.
• Built-in web server for configuration.
• Balanced & unbalanced analogue audio
inputs.
• AES/EBU, S/PDIF and TOSlink digital
audio inputs.
• Front panel headphone monitor with
volume control.
• 6 x GPI for triggering remote events.
The PS-SEND is a freestanding audio to IP
converter which is also available in a 1U
rack-mount as the PS-SENDS.
It receives audio from a number of user
selectable external stereo sources including
balanced and unbalanced analogue audio,
AES/EBU, S/PDIF & TOSlink digital audio.
Once an audio source is selected, the unit
encodes the audio in real time and sends it
to the network as an encoded stream. The
audio stream can be distributed over an IP-
All the configuration settings for the
unit are accessed via a local web-server
built into it. The type of encoding and
the transport mechanism are defined
by selecting the connection from a predefined list. The PS-SEND encodes an
audio source into an MP3 (from analogue
or digital inputs), G.711 or PCM (from
analogue inputs only) audio stream using
HTTP, RTP, raw UDP or raw TCP protocols,
including multicast support and the
following encoder types are available:
Mpeg1 & 2 Layer3, MP3-CBR (constant
bit-rate), PCM linear and A-law, U-law,
with 8kHz-48kHz sample rates . The unit
can configure its own IP address using
DHCP/BOOTP, IPzator or AutoIP. A readout
of the set IP address can be heard on
every reset using SONICIP technology, if
selected. Remote level monitoring is also
possible using SNMP traps.
Two red and green front panel LEDs
indicate what state the unit is currently
in, be it normal operational mode or
bootstrap mode, and also indicates the
current network connection status. A
blue LED denotes power to the unit. The
input being routed to the IP stream can be
Telephony & Commentary - Pro Audio Streamers
monitored on the front panel ¼” (6.35mm)
stereo jack socket in combination with a
headphone volume knob.
The rear panel has 2 x RJ45 connectors,
one for the 10/100Mbit Ethernet interface
and one for GPI connections. The PS-SEND
has 6 x GPIs which can be used to trigger
the sending of the audio stream and which
can also be used to trigger remote events
using an output relay on the PS-PLAY and
PS-AMP. There is a 9 way D-type RS232
serial connection for control of the unit by
automation systems and firmware updates.
The unit can be remote controlled via serial
connection, TCP or UDP.
Power to the unit is via a universal supply
85V - 264V fused IEC mains socket.
Specification For PS-SEND
Analogue Inputs: 2 x XLR 3 pin (balanced)( L&R)
2 x RCA phono (unbalanced)
( L&R)
Audio Codec Specifications PS-SEND
Analogue Max
18dBu XLR balanced
Input Level:
8dBu RCA phono unbalanced
WAV (IMA ADPCM+ 16bit PCM uncompressed: 8kHz
to 48kHz)
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Impedance:
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Analogue
Input SNR:
74dB
Input THD:
0.02% Relative
Interchannel
80dB (Ref FSD)
Isolation (Cross Talk):
Digital Inputs:
1 x AES/EBU XLR 3 pin female
1 x S/PDIF RCA phono
1 x TOSLink optical input
Analogue
Outputs:
1 x 6.35mm (¼”) jack headphone
socket
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GPIs (General
6 x GPIs, selectable via webpage
Purpose Inputs): control on RJ45 socket
Serial Port:
1 x 9 way D-type socket, used to
send control commands and
update firmware
Ethernet Port:
1 x RJ45 socket. Remote control
commands can be sent via TCP or
UDP as well as firmware updates
Mains Input:
Filtered IEC, 85 - 264VAC, 47 - 63 Hz,
10W, max
Fuse Rating:
Anti-surge fuse 1A 20 x 5mm
G.711 (U Law/A Law 8kHz to 48kHz sampling rate)
MP3 MPEGv1 Layer 3 (32, 44.1 and 48 kHz, CBR)
MP3 MPEGv2 Layer 3 (16, 22.05 and 24 kHz, CBR)
General Features
PS-SEND Home Page.
Supported network transport protocols
RTP - UDP
HTTP - TCP
Raw UDP
Raw TCP
Can also act as Icecast/Shoutcast source
SNMP - traps for remote management
DHCP, BOOTP, IPZator or AUTOIP - Dynamic IP address
resolution
SonicIP IP Address readout
Physical Specification
PS-SEND
Dimensions
(Raw):
22cm (W) x 13.7cm (D) x 4.3cm (H)
8.67” (W) x 5.39” (D) x 1.7” (H)
Dimensions
(Boxed):
34cm (W) x 27cm (D) x 6cm (H)
13.4” (W) x 10.6” (D) x 2.4” (H)
Weight:
Nett: 1.0kg
Nett: 2.2lbs
Gross: 1.7kg
Gross: 3.7lbs
PS-SEND Streaming Settings Page.
PS-SENDS
Dimensions
(Raw):
48.3cm (W) x 13.7cm (D) x 4.3cm (H)
19” (W) x 5.39” (D) x 1.7” (H)
Dimensions
(Boxed):
58.8cm (W) x 27cm (D) x 6.8cm (H)
23” (W) x 10.6” (D) x 2.7” (H)
Weight:
Nett: 1.1kg
Nett: 2.4lbs
Gross: 1.8kg
Gross: 4.0lbs
PS-SEND Network Settings Page.
267
Telephony & Commentary - Pro Audio Streamers
PS-PLAY IP to Audio Streaming Decoder
as those generated by the PS-SEND (MP3,
G.711 and PCM) and including Ogg Vorbis
audio files from external USB as well as from
sources such as Shoutcast, Icecast (Internet
radio), VLC and from RTP servers. The unit
can receive streams from HTTP (TCP/IP) and
RTP (UDP) protocols, as well as raw TCP and
UDP packets. The unit can also configure its
own IP address using DHCP/BOOTP, IPzator
or AutoIP. A readout of the set IP address
can be heard on every reset using SONICIP
technology, if selected.
It takes an IP audio feed and converts it to
a number of simultaneous stereo outputs:
balanced and unbalanced analogue audio,
AES/EBU, S/PDIF & TOSlink digital audio
outputs.
Two red and green front panel LEDs indicate
what state the unit is currently in, be it
normal operational mode or bootstrap
mode, and also indicates the current
network connection status. A blue LED
denotes power to the unit. The audio output
can be monitored on the front panel ¼”
(6.35mm) stereo jack socket in combination
with a headphone volume knob. The
analogue audio outputs can be switched
to be either a fixed level output or to be
controlled by the front panel volume knob.
The volume can also be adjusted, as well
as many other features, using an infra-red
remote control (available separately).
As for the PS-SEND, all the configuration
settings for the unit are accessed via a local
web-server built into it. The unit can decode
one of a number of audio streams, such
The PS-PLAY can be configured with up
to 3 sources. The sources are prioritized
in number order. If one has failed, the
next one will attempt to play. If all fail, an
PS-PLAY IP to Audio Streaming Decoder.
Category: Pro Audio Streamers.
Product Function: IP to audio converter.
Typical Applications: See the front page
of this section for application ideas.
Features:
• IP to audio streamer.
268
• Built-in web server for configuration.
• Balanced & unbalanced analogue audio
outputs.
• AES/EBU, S/PDIF and TOSlink digital
audio outputs.
• Front panel headphone monitor with
local & remote volume control.
• USB port acts as an audio source.
• Can decode from a multitude of
services, e.g. Shoutcast, Icecast.
• 2 x GPO relays controlled remotely
from a PS-SEND.
The PS-PLAY is a freestanding IP to audio
converter which is also available in a 1U
rack-mount as the PS-PLAYS.
Telephony & Commentary - Pro Audio Streamers
external USB drive will be used as a back-up
source. The external USB plugs into the USB
socket on the front of the unit. This enables
the PS-PLAY to act as a USB audio player,
playing any of the audio formats previously
mentioned.
Specification For PS-PLAY
The rear panel has 2 x RJ45 connectors,
one for the 10/100Mbit Ethernet interface
and one for GPIO connections. The PSPLAY has 2 output relay contacts which
can triggered remotely over IP from a
connected PS-SEND unit. There is a 9 way
D-type RS232 serial connection for control
of the unit by automation systems and for
firmware updates. The unit can be remote
controlled via serial connection, TCP or UDP
and remote management of the unit is also
possible using SNMP traps.
Power to the unit is via a universal supply
85V - 264V fused IEC mains socket.
Analogue
Outputs:
L&R)
2 x XLR 3 pin (balanced)( L&R)
2 x RCA phono (unbalanced)(
1 x ¼ inch (6.35mm) stereo jack
headphone socket
Analogue Max
Output Level:
18dBu XLR balanced
8dBu RCA phono unbalanced
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Impedance:
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Output SNR:
94dB
Output THD:
0.03% Relative
Interchannel
80dB (Ref FSD)
Isolation (Cross Talk):
Analogue
Output Gain
Range:
-60dB to 18dB via front panel
control knob, or optional IR
controller
Digital Outputs:
1 x AES/EBU XLR 3 pin female
1 x S/PDIF RCA phono
1 x TOSLink optical input
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USB Port:
1 x USB A socket
GPOs (General
Purpose
Outputs):
2 x switchable relay contacts
(simultaneously switched)
controlled from PS-SEND
Serial Port:
1 x 9 way D-type socket, used to
send control commands and
update firmware
Ethernet Port:
1 x RJ45 socket. Remote control
commands can be sent via TCP or
UDP as well as firmware updates.
IR Remote
Receiver:
Remote commands can be sent
using optional remote control via
built in IR sensor
Mains Input:
Filtered IEC, 85 - 264VAC, 47 - 63 Hz,
10W, max
Fuse Rating:
Anti-surge fuse 1A 20 x 5mm
Audio Codec Specifications PS-PLAY
G.711 (U Law/A Law 8kHz to 48kHz sampling rate)
WAV (IMA ADPCM+ 16bit PCM uncompressed: 8kHz
to 48kHz)
MP3 MPEGv1 Layer 3 (32, 44.1 and 48kHz, CBR +VBR +ABR)
MP3 MPEGv2 Layer 3 (16, 22.05 and 24kHz, CBR +VBR
+ABR)
Ogg Vorbis (floor 1)
PS-PLAY Home Page.
General Features
Supported network transport protocols
RTP - UDP
HTTP - TCP
Raw UDP
Raw TCP
SNMP - traps for remote management
DHCP, BOOTP, IPZator or AUTOIP - Dynamic IP address
resolution
SonicIP IP Address readout
Physical Specification
PS-PLAY
Dimensions
(Raw):
22cm (W) x 13.7cm (D) x 4.3cm (H)
8.67” (W) x 5.39” (D) x 1.7” (H)
Dimensions
(Boxed):
34cm (W) x 27cm (D) x 6cm (H)
13.4” (W) x 10.6” (D) x 2.4” (H)
Weight:
Nett: 1.0kg Gross: 1.7kg
Nett: 2.2lbs Gross: 3.7lbs
PS-PLAY Advanced Stream Settings Page.
PS-PLAYS
Dimensions
(Raw):
48.3cm (W) x 13.7cm (D) x 4.3cm (H)
19” (W) x 5.39” (D) x 1.7” (H)
Dimensions
(Boxed):
58.8cm (W) x 27cm (D) x 6.8cm (H)
23” (W) x 10.6” (D) x 2.7” (H)
Weight:
Nett: 1.1kg Gross: 1.8kg
Nett: 2.4lbs Gross: 4.0lbs
PS-PLAY Network Settings Page.
269
Telephony & Commentary - Pro Audio Streamers
PS-AMP IP to Speakers Streaming Decoder
The PS-AMP is a freestanding unit which
converts an IP audio stream directly to
speaker outputs.
Category: Pro Audio Streamers.
Product Function: IP to audio converter.
Typical Applications: See the front page
of this section for application ideas.
Features:
• IP to audio speaker terminals
streamer.
• Built-in web server for configuration.
• 1 x stereo speaker terminal outputs.
• Front panel headphone monitor with
local & remote volume control.
• USB port acts as an audio source.
• Can decode from a multitude of
services, e.g. Shoutcast, Icecast.
• 2 x GPO relays controlled remotely
from a PS-SEND.
270
It is also available in a 1U rack-mount as
the PS-AMPS. The PS-AMP has the same
feature-set as the PS-PLAY except that there
are no individual audio outputs other than
the speaker terminals. The PS-AMP uses
an integrated 2 x 15W D-class amplifier to
deliver audio directly to a pair of connected
speakers.
For further information on features please
refer to the PS-PLAY.
PS-AMP IP to Speakers Streaming Decoder
Telephony & Commentary - Pro Audio Streamers
Specification For PS-AMP
Analogue
Outputs:
2 x speaker connectors (2 each
black and red terminals)
1 x ¼” (6.35mm) stereo jack
headphone socket
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Headphone
Level Range:
-60dB – 18dB via front panel control
knob or optional IR controller
USB Port:
1 x USB A socket
GPOs (General
2 x switchable relay contacts
Purpose Outputs): (simultaneously switched)
controlled from PS-SEND
General Features
Supported network transport protocols
RTP - UDP
HTTP - TCP
Raw UDP
Raw TCP
SNMP - traps for remote management
DHCP, BOOTP, IPZator or AUTOIP - Dynamic IP address
resolution
SonicIP IP Address readout
PS-AMP Home Page.
Physical Specification
PS-AMP
Dimensions
(Raw):
22cm (W) x 13.7cm (D) x 4.3cm (H)
8.67” (W) x 5.39” (D) x 1.7” (H)
Dimensions
(Boxed):
34cm (W) x 27cm (D) x 6cm (H)
13.4” (W) x 10.6” (D) x 2.4” (H)
Weight:
Nett: 1.0kg
Nett: 2.2lbs
Serial Port:
1 x 9 way D-type socket, used to
send control commands and
update firmware
Ethernet Port:
1 x RJ45 socket. Remote control
commands can be sent via TCP or
UDP as well as firmware updates.
IR Remote
Receiver:
Remote commands can be sent using
optional remote control via built in IR
sensor
PS-AMPS
Dimensions
(Raw):
48.3cm (W) x 13.7cm (D) x 4.3cm (H)
19” (W) x 5.39” (D) x 1.7” (H)
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THD+N
Dimensions
(Boxed):
58.8cm (W) x 27cm (D) x 6.8cm (H)
23” (W) x 10.6” (D) x 2.7” (H)
Mains Input:
Filtered IEC, 85 - 264VAC, 47 - 63 Hz,
60W max
Weight:
Nett: 1.1kg
Nett: 2.4lbs
Fuse Rating:
Anti-surge fuse 1A 20 x 5mm
Gross: 1.7kg
Gross: 3.6lbs
Gross: 1.9kg
Gross: 4.2lbs
PS-AMP Advanced Stream Settings Page.
Audio Codec Specifications PS-AMP
G.711 (U Law/A Law 8kHz to 48kHz sampling rate)
WAV (IMA ADPCM+ 16bit PCM uncompressed: 8kHz
to 48kHz)
MP3 MPEGv1 Layer 3 (32, 44.1 and 48kHz, CBR +VBR +ABR)
MP3 MPEGv2 Layer 3 (16, 22.05 and 24kHz, CBR +VBR
+ABR)
Ogg Vorbis (floor 1)
PS-AMP Network Settings Page.
271
Sonifex SignalLED Sign.
Photo courtesy of ABC Erina & OnAir Solutions.
272
Studio Illuminated Signs
The SignalLED range of illuminated RGB LED signs are a range of signs designed for use
outside recording studios, on-air & production studios, meeting rooms, conference rooms
and for fixed installations. The sign itself contains the control electronics, and RGB LEDs are
used, so the signs can be simply configured onsite for your particular requirements.
SignalLED Range of LED Signs
274
Custom Signs
276
Sonifex SignalLED Sign.
Photo courtesy of ABC Erina & OnAir Solutions.
273
Studio Illuminated Signs
SignalLED Range of LED Signs
Category: SignalLED Illuminated Signs.
Product Function: Signage for outside
studio/office doors and on studio walls
to indicate Mic Live, Recording, etc
Typical Applications: MIC LIVE, ON-AIR,
PHONE and RECORD signs for radio/TV
studios, TX-REH for production galleries,
MEETING IN PROGRESS for board rooms,
SILENCE PLEASE outside lecture theatres.
The SignalLED range of illuminated
RGB LED signs are a new range of signs
designed for use outside recording studios,
on-air & production studios, meeting
rooms, conference rooms and for fixed
installations. The sign itself contains the
control electronics, and RGB LEDs are
used, so the signs can be simply configured
onsite for your particular requirements.
Features:
• Low power consumption, only 4W.
• Maintenance free, LED lighting.
• Multi-coloured RGB LEDs offering 9
different colours.
• Sign can be made to flash, pulse, fade and
be on or off.
• Flush-mounting as standard or end/
ceiling-mounting with kits.
• 20cm (8”) or 40cm (16”) widths or 2 x
20cm (2 x 8”).
• Standard wording includes: MIC LIVE, ON
AIR, RECORD, PHONE, TRAFFIC FLAG ON,
AD BREAK, TX, REH, REHEARSAL, DOOR,
Different Display Modes
Five modes of illumination are available:
•
•
•
•
•
Constant illumination.
Flashing, regularly on/off.
Pulsing - flashing, then off, repeated
Fading
Off
•
•
•
•
•
OBIT, NO ENTRY, EXIT, SILENCE PLEASE,
MEETING IN PROGRESS, INTERVIEW IN
PROGRESS.
Custom signs can be etched with
company logos, different wording or
graphics.
2 x GPIs for control from a light switch,
or fader GPO (pull down to 0V).
Supplied as standard as red/green
switching.
Use separate remote control (LD-RPC)
to program the colour, switching mode
or brightness.
Supplied complete with fixings & DC
power supply.
RGB LEDs - Any Colours Possible
The illuminating LEDs in the unit are RGB
meaning that the colour of the sign can be
selected on installation.
You can configure the sign from one of 9
colours: white, green, red, blue, brick red,
yellow, orange, cyan and magenta.
9 different colours of the SignalLED Sign.
274
Studio Illuminated Signs
Single or Twin Signs
Each sign is 40cm or 20cm long. The 40cm
signs can be split into two 20cm sides which
can be separately, or jointly controlled, e.g.
you can have a 40cm ‘ON AIR’ sign, or twin
2 x 20cm signs, such as ‘ON AIR’ and ‘MIC
LIVE’.
Single 20cm Flush Mounting Sign
These are the available choices for the
smaller 20cm flush-mounting signs. Custom
signs can be made to order, if required.
LD-20F1REC Single Flush Mounting 20cm
‘RECORD’ Sign.
LD-40F1ONA Single Flush Mounting
40cm ‘ON AIR’ Sign.
LD-KC1 Ceiling Mounting Kit
The LD-KC1 mounting kit can be used to
attach a 40cm or 20cm flush mounting sign
to the ceiling or a desktop.
The LD-KC1 mounting kit used to attached the sign
to the ceiling.
Dual Control Inputs
The sign can be controlled by either, or both,
of 2 pull-low inputs, e.g. a single input can
be used to control a single ‘MIC LIVE’ sign,
or 2 inputs can be used to control a twin ‘TX’
and ‘REH’ sign with independent control of
each side of the sign.
7V-36V DC Input Option
For a small additional cost, the SignalLED
main circuit board can be modified to accept
a DC input in the range 7V-36V, for example,
if you have a 12V or 24V distributed DC
power network.
Please note that for 7V – 36V ‘LDD’ orders, a
DC power supply is not included as standard.
LD-20F1ONA Single Flush Mounting 20cm
‘ON AIR’ Sign.
LD-40F2ONA-MCL Twin ‘ON AIR’
& ‘MIC LIVE’ sign.
The LD-KC1 mounting kit used to attached the sign
to the desktop.
When placing your order, or requesting a
quotation, please either:
Change the ‘LD’ of the order code to ‘LDD’,
so for example, ‘LD-F1ONA’ becomes ‘LDDF1ONA’, or:
LD-20F1MCL Single Flush Mounting 20cm
‘MIC LIVE’ Sign.
Add the item LD-DC to your order.
The LD-KC1 mounting kit.
275
Studio Illuminated Signs
Flush or End Mounting
The sign is supplied as standard so that it
can be mounted flush to a wall, e.g. for the
‘TX’ & ‘REH’ sign shown below. However, a
mounting kit is available, LD-KE1, to mount
the sign perpendicular to a wall, e.g. above a
door, such as the ON AIR sign shown.
Simple Installation
The signs are supplied with a 6V DC power
supply. Simply wire up your control signal(s)
and DC input to the ‘screw terminal block’
inside the sign, route the cables through the
integral cable clamp and mount the unit on
the wall.
An installation tool, the LD-IT can be used
to hold the sign in place on the wall mount
whilst you wire it up.
LD-KE1 End Mounting Kit For 40cm Or 20cm Flush
Mounting Signs.
The display mode and colour options are
set-up using a separate remote control, the
LD-RPC.
LD-RPC SignalLED Remote
Programming Controller
The SignalLED signs are preconfigured to
display certain colours when the control
inputs are active. Information on the colours
shown can be found on the Sonifex website.
To vary the colours and switching modes,
the LD-RPC Remote Programming Controller
must be used.
This allows a simple way of programming
any sign using a hand-held remote control.
Custom Signs
Although a range of standard signs is
supported, custom signs, for example
with your company name or logo, can be
produced for a nominal setup fee. Contact
Sonifex with your requirements.
Please note
that the
control is sold
separately and one
LP-RPC can program
multiple signs.
Examples Of Custom Signs
End Mounting 40cm Single Sign (Created From
Single Flush Mounting LD-40F1ONA Sign & LD-KE1
kit).
LD-IT LED Sign End Mounting Installation Tool.
LD-40F2TX-REH Flush Mounting 40cm
Twin Sign.
Standard DC Power Supply with Plug Adapters.
LD-40F2ONA-SIL Twin ‘ON AIR’ & ‘SILENCE’ sign.
276
Custom Single Flush Mounting 40cm Sign.
Studio Illuminated Signs
Specification For SignalLED
Power Input:
5-7V DC
40cm Sign: 500mA max
Input Connector: 4 way screw terminal block
How to Order
Just select the name of the sign from the
list below.
LD-40F1TRF Single Flush Mounting 40cm ‘TRAFFIC
FLAG ON’ Sign.
LD-40F1NOE Single Flush Mounting 40cm
‘NO ENTRY’ Sign.
LD-40F1ONA Single Flush Mounting 40cm
‘ON AIR’ Sign.
Control Inputs:
2 x pull-down to 0V
Dimensions
(Perspex):
Small Single: 20cm (W) x 8cm (H)
Large Single: 40cm (W) x 8cm (H)
Dimensions
(Boxed 20cm):
40cm (W) x 20cm (D) x 11cm (H)
15.7” (W) x 7.9” (D) x 4.3” (H)
Dimensions
(Boxed 40cm):
60cm (W) x 20cm (D) x 11cm (H)
23.6” (W) x 7.9” (D) x 4.3” (H)
Weight (20cm):
Nett: 0.35kg Gross: 0.85kg
Nett: 0.8lb Gross: 1.9lb
Weight (40cm):
Nett: 0.55kg Gross: 1.15kg
Nett: 1.2lb Gross: 2.6lb
Equipment Type
Single Flush Mounting Signs (40cm):
LD-40F1REC
LD-40F1ADB Single Flush Mounting 40cm
‘AD BREAK’ Sign.
LD-40F1EXIT Single Flush Mounting 40cm
‘EXIT’ Sign.
LD-40F1MCL Single Flush Mounting 40cm
‘MIC LIVE’ Sign.
LD-40F1REH Single Flush Mounting 40cm
‘REHEARSAL’ Sign.
LD-40F1SIL Single Flush Mounting 40cm ‘SILENCE
PLEASE’ Sign.
LD-40F1PHN Single Flush Mounting 40cm
‘PHONE’ Sign.
40cm ‘RECORD’ Sign
LD-40F1ONA
40cm ‘ON AIR’ Sign
LD-40F1MCL
40cm ‘MIC LIVE’ Sign
LD-40F1PHN
40cm ‘PHONE’ Sign
LD-40F1TRF
40cm ‘TRAFFIC FLAG ON’ Sign
LD-40F1ADB
40cm ‘AD BREAK’ Sign
LD-40F1REH
40cm ‘REHEARSAL’ Sign
LD-40F1DOR
40cm ‘DOOR’ Sign
LD-40F1OBT
40cm ‘OBIT’ Sign
LD-40F1NOE
40cm ‘NO ENTRY’ Sign
LD-40F1EXIT
40cm ‘EXIT’ Sign
LD-40F1SIL
40cm ‘SILENCE PLEASE’ Sign
LD-40F1MET
40cm ‘MEETING IN PROGRESS’ Sign
LD-40F1INT
40cm ‘INTERVIEW IN PROGRESS’ Sign
Twin Flush Mounting Signs (2 x 20cm):
LD-40F2TX-REH
2 x 20cm ‘TX’ & ‘ REH’ Sign
LD-40F2ONA-MCL 2 x 20cm ‘ON AIR’ & ‘MIC LIVE’ Sign
LD-40F1DOR Single Flush Mounting 40cm
‘DOOR’ Sign.
LD-40F1MET Single Flush Mounting 40cm
‘MEETING IN PROGRESS’ Sign.
Single Flush Mounting Signs (20cm):
LD-20F1REC
20cm ‘RECORD’ Sign
LD-20F1ONA
20cm ‘ON AIR’ Sign
LD-20F1MCL
20cm ‘MIC LIVE’ Sign
Mounting Kits:
LD-40F1OBT Single Flush Mounting 40cm
‘OBIT’ Sign.
LD-40F1INT Single Flush Mounting 40cm
‘INTERVIEW IN PROGRESS’ Sign.
LD-KE1
End Mounting Kit For 40cm Or 20cm
Flush Mounting Signs
LD-KC1
Ceiling Mounting Kit For 40cm Or
20cm Flush Mounting Signs
LED Sign End Mounting Installation
Tool
LD-IT
277
278
Index
Index
Stock Code
Description
Page No
3
3G, HD & SD-SDI Embedders & De-Embedders
Description
Page No
35
208
Index
279
IP Streamers
265
K
Key to Symbols:
C
Stock Code
Description
RB-AEC Acoustic Echo Canceller
I
A
Alternative S2 Button Text
Stock Code
70
RB-AES4B3 Quad 3 Way Passive AES3ID Splitter BNC Connectors
112
RB-AES4X3 Quad 3 Way AES/EBU Splitter XLR Connectors
113
RB-BL2 Single Stereo Bi-Directional Matching Converter
RB-BL4 Dual Stereo Bi-Directional Matching Converter
i
Page No
RB-DA4x5 4 Input, 4 x 5 Output Distribution Amplifier/Mixer
90
91
100
RB-DA6 6 Way Stereo Distribution Amplifier
95
RB-DA6G 6 Way Stereo Distribution Amplifier With Output Gain Control
96
RB-DA6P 6 Way Stereo Distribution Amplifier With Phoenix Connectors
97
RB-DA6R 6 Way Stereo Distribution Amplifier With RJ45 Connectors
98
CM-AESB3 Single 3 Way Passive AES3ID Splitter With BNC Connectors
154
M
CM-AESX3 Single 3 Way AES/EBU Passive Splitter With XLR Connectors
153
Mentor Time-Server
173
CM-BH4W Belt-Pack 4-Wire Headphone Amp
238
Monitor Speaker Stands
221
CM-BHA Belt Pack Headphone Amplifier With Limiter & Loudspeaker
239
CM-CU1 Commentator Unit 1 Commentator & Line Input
230
CM-CU21 Commentator Unit 2 Commentators & 1 Guest
234
CM-GM1 Professional Gooseneck Condenser Microphone
229
P
CM-MS3 Single 3 Way Passive Microphone Splitter
155
PC-AD2 2 Stereo Analogue I/O & 2 Stereo AES-3 I/O PCIe Half Height
Sound Card
158
PC-AM6-32 AM Radcap PCle Card (6 to 32 Channels)
165
RB-DDA6A-2P 6 Way Stereo AES/EBU Digital Distribution Amplifier with
Dual Power Supplies
103
PC-AUR44 Auricon 4.4 PCIe Analogue Sound Card
162
RB-DDA6A3 6 Way Stereo AES3ID Digital Audio Distribution Amplifier
107
CM-TB8 8 Channel Talkback Control Unit
224
CM-TB8G Talkback Control Unit, CM-TB8 with CM-TBG Dual GSM Interface
228
N
Net-Log Network Audio Logger
168
RB-DA6RG 6 Way Stereo Distribution Amplifier With RJ45 Connectors
& Output Gain Control
RB-DA24MD 24 Way Mono Audio Distribution Amplifier
99
110
RB-DAC1 Digital To Analogue Converter
60
RB-DD4 4 Channel Digital Audio Delay
68
CM-TB8T Talkback Control Unit, CM-TB8 with CM-TBT Dual PSTN
Telephone Hybrid
227
PC-DAB1-4 Multi-Ensemble DAB+/DAB Radcap PCle Card
163
RB-DDA6A 6 Way Stereo AES/EBU Digital Distribution Amplifier
102
CM-TBU Line Powered Telephone Balance Unit
248
PC-DIG4 Digitorc 4, 4 Stereo AES-3 I/O PCIe Sound Card
161
RB-DDA6S 6 Way Stereo S/PDIF Digital Distribution Amplifier
104
CM-TLL Line Powered Telephone Line Listen Unit
249
PC-FLS8 Flashlog 8 Audio & Radio Capture Logging Software
174
PC-FLS8LT1 Flashlog 8 Single Stereo Channel Logging Software
176
RB-DDA6W-2P 6 Way Word Clock Distribution Amplifier with Dual
Power Supplies
106
PC-FLS8LT2 Flashlog 8 Dual Stereo Channel Logging Software
176
RB-DDA6W 6 Way Word Clock Distribution Amplifier
105
PC-FM6-32 FM Radcap PCle Card (6 to 32 Channels)
164
RB-DDA22 Digital Audio Distribution Amplifier With Multiple Outputs
108
PC-RACK Industrial PC Racks for Audio Logging
177
RB-DEDD8 Dolby® E & Dolby Digital Decoder 8 Channel, Digital Outputs
50
Portable Radio Studio Package
220
RB-DEED8 Dolby® E Encoder 8 Channel, Digital Inputs
48
Pro Audio Streamers
265
RB-DHD6 Digital 6 Way Stereo Headphone Distribution Amplifier
Professional Sound Cards
158
RB-DMA2 Dual Digital Microphone Amplifier
138
PS-AMP IP to Speakers Streaming Decoder
270
RB-DMX4 4 x 4 Channel Digital Audio Mixer/Router
128
PS-PLAY IP to Audio Streaming Decoder
268
RB-DS2R Remote Switch Panel For RB-DS2
63
PS-SEND Audio to IP Streaming Encoder
266
RB-DS2 Stereo Delay Synchroniser & Time-Zone Delay
64
RB-DSD1 Digital Silence Detection Unit
76
Contents
iii
D
DHY-03, DHY-03S & DHY-03T Digital Telephone Hybrids
252
DHY-03EC Automatic Digital Telephone Balance Unit Eurocard
255
DHY-04G Single Automatic GSM Hybrid, AES/EBU & Analogue I/O
With Ethernet
258
DHY-04HD Single Automatic HD Voice & GSM Hybrid, AES/EBU &
Analogue I/O With Ethernet
260
DHY-04 Single Automatic Digital TBU, AES/EBU & Analogue I/O
With Ethernet 256
F
Free Sonifex Switchboard Software - Remote Control of DHY-04/G/HD
Hybrids
R
263
H
HY-03, HY-03S & HY-03T Analogue Telephone Hybrids
250
Radio Capture Cards
163
Radio Studio Packages
216
RB-ADDA2 A/D and D/A Converter, 24 bit 192kHz
54
RB-ADDA Combined A/D and D/A Converter
52
RB-DSD8 8 Channel Silence Switcher
RB-DSS10 10 Way Stereo Digital Source Selector
120
79
124
RB-FS42 Audio Failover Switcher, 4 Main I/O, 2 Standby I/O
82
RB-FS82 Audio Failover Switcher, 8 Main I/O, 2 Standby I/O
84
RB-HD1 Stereo Headphone Amplifier With VCA Volume Control
117
279
Index
Stock Code
Description
RB-HD2 Dual Stereo Headphone Amplifier
Page No
Stock Code
Description
Page No
Stock Code
Description
Page No
RB-HD6 6 Way Stereo Headphone Distribution Amplifier
119
RB-VHCMD16 3G/HD/SD-SDI Embedder & De-Embedder 16 Channel
Digital I/O
RB-IPE IP Extender for GPIO & Analogue Control Signals
148
RB-VHDA2x4 3G/HD/SD-SDI 2 Input, 8 Output Video Distribution Amplifier
32
RB-LC3 3 Way Light/Power Controller
144
RB-VHDA8 3G/HD/SD-SDI 1 Input, 8 Output Video Distribution Amplifier
34
RB-VHDDD8 3G/HD/SD-SDI De-Embedder Dolby® E & Dolby Digital Decoder
& Re-Embedder
RM-MC51L Monitor Controller, 5.1 Stereo Inputs & Outputs With
Light Control
26
46
RM-MCL Monitor Light Controller, 3 Button, TX-STANDBY-REH
27
RB-VHDMA8 3G/HD/SD-SDI De-Embedder, 8 Channel Analogue Output
36
RB-LI2 Stereo Line Isolation Unit
RB-LU4 Quad Stereo Balanced To Unbalanced Converter
118
93
89
RB-MA1 Single Microphone Amplifier
135
RB-MA2 Dual Microphone Amplifier
136
RB-MA2G Dual Microphone Amplifier Gain Controls
137
RB-ML2 Stereo Microphone & Line Level Limiter
140
RB-MM1 Mix-Minus Generator
145
RB-MS4X3 Quad 3 Way Passive Microphone Splitter
114
RB-MSP6 6 Way +48V Phantom Power Supply
116
RB-MTV1 Contribution Voiceover Monitor With Talkback
146
RB-OA3 3 Studio On-Air Switcher
132
RB-OA3C Expansion Unit Cable For RB-OA3
134
RB-OA3R Remote Switcher Panel For RB-OA3
134
RB-PA2 Dual Stereo RIAA Phono Amplifier
92
RB-PD2 Stereo Profanity Delay
66
RB-PLI6 6 Way Mono Passive Line Isolation Unit
94
RB-PMX4 10 Input, 4 Output Analogue Pre-set Mixer
126
RB-SC1 Sample Rate Converter
56
RB-SC2 Dual Sample Rate Converter
58
RB-SD1IP Silence Detection Unit With Ethernet & USB
74
RB-SD1 Silence Detection Unit
72
RM-MC1L Monitor Controller, Single Stereo Input & Output With Light Control 24
42
RB-VHEDD8 3G/HD/SD-SDI De-Embedder, Dolby® E Encoder & Re-Embedder 44
RB-VHEMA8 3G/HD/SD-SDI Embedder, 8 Channel Analogue Inputs
38
Redbox Audio & Video Interfaces
30
Reference Monitor Controllers
23
Reference Monitor Meters
18
Reference Monitor Rackmount Audio Monitors
2
RM-2S4 Reference Monitor, 2 LED Meters & 4 Stereo Inputs
4
RM-2S10 Reference Monitor, 2 LED Meters, 10 Stereo Analogue &
10 Stereo Digital Inputs
5
RM-4C8-HD1 Reference Monitor, 4 LED Meters, 8 Channel Inputs &
Dual Source Selectors With RM-HD1 3G/HD/SD-SDI De-Embedder Card
RM-4C8-HDE1 Reference Monitor, 4 LED Meters, 8 Channel Inputs &
Dual Source Selectors With De-Embedder & Dolby Decoder Card
RM-4C8 Reference Monitor, 4 LED Meters, 8 Channel Inputs &
Dual Source Selectors
9
10
7
RM-CA2 Confidence Monitor, 2 LED Meters & 2 Analogue Stereo Inputs
14
RM-CAD8 Confidence Monitor, 2 LED Meters, 2 Analogue & 6 Digital
Stereo Inputs
16
RM-E1B Reference Monitor Dolby E & Dolby® Digital Decoder BNC
Expansion Card
13
RB-SL2 Twin Mono, Or Stereo, Limiter
141
RM-E1X Reference Monitor Dolby E & Dolby® Digital Decoder XLR
AES Expansion Card
13
RB-SM1 Single Stereo To Mono Converter
142
RM-HD1 Reference Monitor 3G/HD/SD-SDI Expansion Card
11
RB-SM2 Dual Stereo To Mono Converter
143
RM-HDE1 Reference Monitor HD-SDI & Dolby® E & Dolby®
Digital Expansion Card
12
RB-SP1 Digital Splitter and Combiner
61
RB-SS10 10 Way Stereo Analogue Source Selector/Mixer
122
RM-M1F53 1 Stereo 53-Segment Meter, Free-Standing
19
RB-SSML1 Mic/Line Source Selector with Compressor/Limiter
130
RM-M1F106 1 Stereo 106-Segment Meter, Free-Standing
21
RB-TGHD(B or X) Multi-Channel HD Tone Generator
150
RM-M1R53 1 Stereo 53-Segment Meter, Rack-Mount
19
21
RB-UL1 Single Stereo Unbalanced To Balanced Converter
86
RM-M1R106 1 Stereo 106-Segment Meter, Rack-Mount
RB-UL2 Dual Stereo Unbalanced To Balanced Converter
87
RM-M2F53 2 Stereo 53-Segment Meters, Free-Standing
20
RB-UL4 Quad Stereo Unbalanced To Balanced Converter
88
RM-M2R53 2 Stereo 53-Segment Meters, Rack-Mount
20
RB-VHCMA4 3G/HD/SD-SDI Embedder & De-Embedder 4 Channel
Analogue I/O
RM-M2R106 2 Stereo 106-Segment Meters, Rack-Mount
22
40
RM-M4R53 4 Stereo 53-Segment Meters, Rack-Mount
20
280
RM-MC4L Monitor Controller, 4 Stereo Inputs, 1 Stereo Output With
Light Control
25
S
S0 Radio Broadcasting On-Air Mixer
180
S1 Radio Broadcast On-Air Mixer
184
S2-05 Chassis
209
S2-7SS Script Space
210
S2-10 Chassis
209
S2-10SS Script Space
210
S2-15 Chassis
209
S2-20 Chassis
209
S2-25 Chassis
209
S2-30 Chassis
209
S2-BI Bus Interlink Cable
212
S2-C6SS 6 Way Stereo Source Select Channel with EQ
197
S2-CB Blank Channel
200
S2-CDAS Dual Stereo Line Channel &
196
S2-CDASE Dual Stereo Line Channel with EQ
196
S2-CDS Digital Dual Stereo Channel
195
S2-CDSE Digital Dual Stereo Channel with EQ
196
S2 Chassis Sizes
209
S2-CML Mic/Line Channel
192
S2-CMM Mic/Mic Channel
192
S2-CS Dual Stereo Line Channel & S2-CSE Dual Stereo Line Channel with EQ 193
S2-CSG Stereo Line & Gram Channel & S2-CSGE Stereo Line & Gram
Channel with EQ
194
S2-CSMM Stereo Mix Minus Channel
198
S2-CT Telco Channel
199
S2 Digital I/O Analogue Radio Broadcast Mixer
190
S2 Input Channels
191
S2-M6SS Meterbridge 6 Way Source Select Panel
207
S2-MB1-5 Meterbridge Blanking Plates 1-5
208
S2 Meterbridge Modules
205
Index
Stock Code
Description
Page No
S2 Mixer Placement In Furniture
211
S2-ML53 Meterbridge LED Meter Panel
206
S2-ML Meterbridge Loudspeaker Monitor Panel
207
S2-MPH Meterbridge Phase Meter Panel
206
S2-MPPM Meterbridge PPM Meter Panel
206
S2-MSB1 S2 Meterbridge Switch Panel With 1 Button
207
S2-MSB3 S2 Meterbridge Switch Panel With 3 Buttons
207
S2-MTB6 Meterbridge 6 Way Talkback Panel
208
S2-MTBS Meterbridge Station Master Talkback Panel
208
S2-MT Meterbridge Dual Timer Panel
207
S2-MVU Meterbridge VU Meter Panel
206
S2-ODA Digital AUD Output Channel &
204
S2-ODAF Digital AUD Output Channel with Master Fader
204
S2-ODP Digital PGM Output Channel &
203
S2-ODPF Digital PGM Output Channel with Master Fader
203
S2-OMC Control Room Monitor Channel
201
S2-OMS Studio Monitor Channel
202
S2 Output Channels
201
S2-PG Penny & Giles Conductive Plastic Fader
204
S2-PSU Power Supply
212
S2-PSUS Dual Power Supply Switcher
213
S2 Solutions Dimensions
222
S2 Solutions Single and Split Mixer Furniture
218
S2 Solutions Technical Specification
217
S2 Without Meterbridge
205
SignalLED Range of LED Signs
274
Sonifex Switchboard Software - Remote Control of DHY-04/G/HD Hybrids
263
Synchronisation Add-On Boards
Sonifex Production - Manufactured in the United Kingdom.
Sonifex Product Testing.
62
T
TB-6D & TB-6R 6 Way Talkback Intercoms
240
The Best Telephone Hybrid in the World Just Got Better!
257
What Is A Telephone Hybrid?
247
281
www.sonifex.co.uk
7-1HſEG
5QPKHGZ.VF
61 Station Road, Irthlingborough,
Northants, NN9 5QE, UK
Tel: +44 (0) 1933 650700
Fax: +44 (0) 1933 650726
Email: [email protected]
#WUVTCNKCP1HſEG
5QPKHGZ2V[.VF
12/6 Leighton Place,
Hornsby NSW 2077, Australia
Tel: +61 (2) 9987 0499
Fax: +61 (2) 9476 4950
Email: [email protected]
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