DELIVER CATALOG
DELIVER CATALOG
Broadcast-grade products for the professional delivery of audio media
!
y
r
na
io
t
u
l
vo
Re
Perfect Audio over Imperfect IP
From APT
APT Audio Codecs
Radio All In One
Audio Processors
RDS encoders
WorldCast Systems is a highly respected provider of professional, reliable and innovative broadcast systems
to the Radio & Television market worldwide. Encompassing the industry-leading brands of APT, Ecreso &
Audemat, WorldCast Systems offers a wide range of high performing products for audio and data delivery,
transmission and monitoring.
Deliver
Transmit
Monitor
APT codecs deliver broadcast-grade audio between locations. Our WorldNet and
WorldCast ranges offer reliable and cost-effective broadcast solutions delivering
high quality content over IP, T1, E1, ISDN & Leased Lines.
Designed for use in studio transmitter links, studio networking and remotes / OB
applications, the APT codec portfolio includes a wide range of stereo and multichannel units that can be deployed as a simple STL or a large-scale IP audio
network.
Ecreso transmitters offer efficient distribution of terrestrial Radio signal for many
FM transmission technologies such as FM, DAB, DAB+ and T-DMB.
With modular design for maximum flexibility and unequalled simplicity of use and
maintenance, our transmitters fit all low power and medium power requirements.
Audemat provides a range of professional monitoring and measurement tools for a
wide variety of broadcast technologies.
The Audemat portfolio includes RF and data monitoring equipment and mobile field
strength meters for analog and digital radio (HD Radio, DAB/DAB+/T-DMB, AM,
FM) and analog and digital TV (DVB-T/H/SH, DVB-T2, ATSC, PAL/SECAM, NTSC). In
addition, Audemat offers industry-leading digital test and measurement equipment,
audio processors, world-class RDS encoders and a range of facility remote control
solutions.
2
Complementing Audemat’s product range is an extensive range of professional
software solutions for the management, configuration and monitoring of broadcast
networks.
1
Deliver...
WorldCast Systems offers an extensive range of broadcast-grade products that ensure the most reliable delivery
of high-quality audio and data to the transmitter site and other locations.
In addition to our highly-regarded APT audio codec line, we also provide a range of associated equipment including
audio processors and RDS encoders from Audemat.
APT Audio Codecs
APT’s award-winning line of professional audio codecs for stereo and
multi-channel studio transmitter links and audio networking.
SureStream, Perfect Audio over Imperfect IP p 3
APT Stereo Audio Codecs p 5
Which APT Codec Solution is for you p 6
Stream-In Silver / Stream-Out Silver, Professional Audio over IP Networking p 7
WorldCast Horizon NextGen, Delivering Broadcast Quality Audio over Inexpensive IP Links p 8
WorldCast IP Decoder, Receiving Broadcast quality Audio over Inexpensive IP Links p 9
WorldCast Equinox Multi-algorithm IP, ISDN & X.21/V.35 audio codec p 10
WorldCast Astral Multi-algorithm IP audio codec p 11
WorldNet Oslo Multi-Channel Multiplexer for audio, voice and data over IP and T1/E1 p 12
Next Gen AOIP Module for the WorldNet Oslo platform p 17
APT Network Management Software p 19
Audio Processing
Digital audio processor that can be used to process audio for delivery.
Digiplexer p 22
Silver Audio Processor 4B-Mini FM p 23
RDS Encoding
Audemat’s industry-leading range of RDS/RBDS encoders offering versatility,
reliability and support for innovative new methods of revenue generation.
Audemat RDS Encoders FMB50, FMB80 & RDS Encoder Silver p 24
Perfect Audio over Low Cost IP
Watch the video at www.surestream.ws
Lose your Synchronous and ISDN Links and Save
Utilize inexpensive IP links (3G, 4G, LAN, WAN, WI-FI, Xdsl)
Always On Air
Protection from loss of connections and dropped packets
No Compromise to Audio Quality
Maintain consistent delay and audio quality
Now shipping on Oslo and Horizon NextGen Audio Codecs
SureStream
Perfect Audio over imperfect IP
SureStream technology is a revolutionary innovation from
APT that enables broadcasters to use inexpensive IP links and
still maintain professional broadcast-grade audio quality and
reliability. It delivers the audio quality and reliability you expect
from a T1/E1 link at a fraction of the associated cost.
The cost savings are so significant that it is possible to achieve a return on your investment in under 4 months!
What is more, the results from tests in the field have been simply outstanding! In one application, a broadcaster set up two ADSL
links which together suffered over 12 connection losses and over 5,000 dropped packets. Using SureStream, not a single packet
was lost in the resultant audio stream and the THD was 100% unaffected.
With SureStream technology, broadcasters wishing to move to IP no longer need to insist on MPLS networks or guaranteed
bandwidth. One or two public IP links are enough to ensure not only continuity of service but also continuity of audio quality.
SureStream ensures optimum quality in a wide variety of applications. For remote and outside broadcast applications, you can
run SureStream on an IP link from a single supplier and for a mission-critical STL, SureStream across two links from separate
suppliers will provide optimum redundancy.
The Benefits of SureStream - Broadcast Audio over
Imperfect IP
• Lose your Synchronous Links & Save
SureStream utilizes inexpensive IP links such as Xdsl, LAN,
WAN, Cable, 3G and 4G wireless, even Wi-Fi.
A combination of Links from different providers can be
used to increased redundancy.
• Always On Air
SureStream protects your audio content from loss of
connection & dropped packets
This diagram shows how SureStream re-constitutes two
imperfect component streams to produce an uninterrupted
output stream. Component streams are affected by packet
loss and LOC (Loss Of Connection) events.
• No Compromise to Audio Quality
SureStream does not tamper with your audio to ensure
continuity of service, nor does it switch between links
depending upon performance. It employs advanced
resequencing technology to offer you seamless streaming
of high quality audio.
SureStream’s innovative approach and its twin benefits of
high audio quality and cost savings have been recognised
throughout the industry and the technology was selected for
not one, but two, prestigious NAB awards in 2011.
• Fixed, Constant Delay
The user-configured latency buffer provides an absolute
fixed and constant delay.
SureStream Technology is available for the following
products:
4
•
WorldNet Oslo
•
WorldCast Horizon NextGen
•
WorldCast IP Decoder
APT Stereo Audio Codecs
APT codecs deliver broadcast-grade audio between locations.
Our WorldNet and WorldCast ranges offer reliable and costeffective broadcast solutions delivering high quality content
over IP, T1, E1, ISDN & Leased Lines.
Designed for use in studio transmitter links, studio
networking and remotes / OB applications, the APT codec portfolio includes a wide range of stereo and multi-channel units that
can be deployed as a simple STL or a large-scale IP audio network.
Studio Transmitter Links using Multicast IP
All WorldCast units provide support for both point-to-point and point-tomulti-point configurations.
In this example, a station is taking
advantage of the highly-efficient
multicasting technique to send
audio and data from a single source
to multiple destinations using the IP
infrastructure.
A unicast return stream can also be set up to provide a confidence
monitoring feed as is shown at Transmitter Site 1.
Studio To Studio Links Over Leased Lines
Permanent digital links such as X.21, V.35, T1/E1, fractional T1/E1, Satellite and Microwave are frequently chosen by broadcasters
for the reliability and guaranteed bandwidth they offer. In this application, the Worldcast Equinox is enabling the contribution
and distribution of audio between
two studio sites over a leased line.
In the case of network outages and
failures, the unit can be configured
to automatically fail over to the
secondary IP connection and
recover, again automatically, when
certain user-defined conditions
are met.
Remote Broadcast over ISDN
WorldCast units take the stress and pain out of setting-up Remote Broadcasts. With low latency Enhanced apt-X compression as
standard, WorldCast codecs enable true real-time remotes and talk-back applications with no annoying delays and ‘over-talk’.
A wide choice of other coding algorithms ensures that the units can connect to other manufacturers’ equipment in the field.
The WorldCast Equinox
provide flexible, highly
interoperable solutions
offering
maximum
connectivity
whether
you need to arrange a
simple mono feed on
the public internet or a
high quality broadcast of
live concert music over
bonded ISDN.
5
On Page Number
Rack Space
DSP-Based Architecture
Redundant Power Supplies
page 6
page 7
page 8
1/2U
1U
1U
page 9
1U
W
or
ld
N
O s et
lo
W
or
ld
As Ca
tr st
al
W
or
Eq ldC
ui as
no t
x
co
de
r
De
IP
W
o
Ho rld
N r C
ex iz as
tG on t
en
S
St tre
re a
m
Si am -I
lv -O n
er u
t
Which APT Audio Codec Solution is for you?
page 10
page 13
1U
3U
Network Interfaces
Ethernet / IP
X.21 / V.35
ISDN (B channels)
T1 / E1
(2)
(2)
(2 )
(2 )
(4)
Audio
Max Channels per unit
Duplex Connections
Balanced Analog I/O (XLR)
Digital AES/EBU I/O
Simultaneous AES/EBU & Analog Outputs
Voice-Grade Audio Transport (FXO/FXS/E&M)
2
2
2
2
2
24
RCA or XLR
Connectors
Algorithms
Linear Audio
Standard apt-X®
Enhanced apt-X®
MPEG 1/2 Layer II
MPEG 1/2 Layer III
MPEG-4 AAC LD & AAC ELD
MPEG-2/4 AAC LC & HE-AAC v1/v2
MPEG Layer III VBR
G.711
G.722
J.57 & J.41
*
(AAC HE only)
(AAC HE only)
Control
Front Panel Keypad Control & LCD Screen
Network Management Software Compatible
Web Browser
Comms Port
Headphone Monitoring &VU Meters
*
(AAC HE only)
*
IP
IP
IP
IP
IP
IP
(Headphones only)
Features
SureStream
SIP/SDP, SNMP & N/ACIP Compliance
Forward Error Correction
Aux Data & GPIO Alarms
Output Silence Detect
Audio Backup via SD card or USB
Automated Line Back-up
*
*
*
*
*
*
USB *
SD *
SD *
(Streaming)
IP
(Streaming)
*
*
(SD)
(SD)
(aux data only)
IP
(Streaming)
*
(Streaming)
ISDN, X.21/V.35
Standard
Optional
* Future Feature
6
*
(Streaming)
T1/E1, IP
Stream-In Silver / Stream-Out Silver
New !
Available Now !
Professional Audio over IP Networking
Stream-In Silver
Stream-In Silver is a highly affordable IP audio encoder designed
for the reliable and robust encoding of audio content over IP
networks. A large range of algorithms including Enhanced
apt-X, MPEG 4 AAC-HE and linear audio are available on the
compact ½ x 1U unit which is fully compatible with all other APT
and N/ACIP –compliant codecs.
Stream-Out Silver
Low on cost but rich in features, the Stream-Out Silver is an IP Audio decoder that decodes a wide range of professional and
consumer audio formats. Perfect for broadcast, retail, hospitality and other applications, the Stream-out Silver offers headphone
monitoring, audio back-up courtesy of either an on-board USB port or ShoutCast server#, embedded auxiliary data and autodetection of an incoming IP stream.
Technical Specification
Size
2 codecs fit into 1U 19” rack space,
½ U each
Dimensions
44mm x 220mm x 160mm
1.73” x 8.66” x 6.3”
Weight
< 1.5 kg (3.3 lbs)
Power Supply
External 12v DC supply with locking
connector
Power Consumption
< 15W
Environmental
0°C to +55°C, 95% humidity
IP
Ethernet 1 x RJ45, RTP/UDP, SIP/
SDP#
Aux Data
9 pin D type DTE, RS232 level. Baud
rates (embedded) 1200 to 9600. Baud
rates (over IP) 1200 to 19200
Control
Web interface, SNMP
Analogue Audio I/O
RCA or XLR connectors
Backup
USB slot for audio backup
Connector
DC power supply connector with locking capability
Analogue Audio
Bandwidth
10Hz through to 22.5kHz mono &
stereo
Sample Rate Converter
8:1
Standard Coding
Eapt-X16/24, Linear 16/20/24
Optional Coding
MPEG 4 AAC HE(v1/v2)
Compression Ratio
Eapt-X 4:1
Others: variable
Coding Delay
Enhanced apt-X 1.9ms
@48kHz Fs
Dynamic Range
> 85dB
Phase Response Linear
DC to Fs/2
Pass Band Ripple
< 0.2dB
Low cost separate encoder and decoder
•
Large range of algorithms as standard (Eapt-X, Linear),
AAC HE v1/v2*#
•
Compact 1/2U design – fit two units in 1U rackspace
•
Compatible with all APT & NACIP-compliant codecs
•
VLAN tagging
•
Fast boot time
•
Intuitive Web interface
•
Quiet, no-fan operation
•
Aux data
•
Headphone monitoring
Interfaces
•
Physical
Key Features Stream-In Silver & Stream-Out Silver:
•
•
•
•
•
Studio Transmitter Links
Confidence Monitoring
Commercial IP audio distribution
IP music distribution systems in hotels, hospitals, campuses etc..
In-store audio distribution applications
Stream-Out : Back Panel View (RCA Version)
Ethernet Port
Audio Output
Aux Data
Port
DC In
Audio
Typical Applications:
*Cost Option - # Contact us for availability of these features
USB Port
for audio backup
7
w!
o
!
N
w
Ne ailable
Av
WorldCast Horizon NextGen
Delivering Broadcast Quality Audio over Inexpensive IP Links
The WorldCast Horizon NextGen boasts the most complete set of IP
features ever included in APT’s extensive range of professional IP codecs
and is the first to feature our revolutionary “SureStream” technology.
Key Features:
•
See page 4 for details
Professional & Affordable
IP audio codec
SureStream technology
enables broadcast-grade
audio over inexpensive IP
links*
Dual IP ports
configurable for back up#
DHCP for automatic
configuration of IP
connection settings
Large range of
algorithms supplied as
standard#
Non-Destructive,
Cascasde-Resilient
Coding with Enhanced
apt-X
•
In addition, this next generation IP audio codec also offers a wide range
of algorithms as standard on a solid DSP platform with dual IP ports and
redundant power supplies. Despite the high feature density, the WorldCast
Horizon NextGen is extremely competitively priced.
•
•
Perfect for STLs and mission-critical applications, the WorldCast Horizon
NextGen provides extensive control and monitoring capabilities to manage
both your audio, data and network conditions and other equipment located
at the transmitter site.
•
•
Delivering Exceptional Audio Quality
• Professional duplex stereo IP audio codec at affordable price
• Wide range of algorithms: Enhanced apt-X algorithm, Linear PCM
& MPEG Layer 3 CBR#
• Support for MPEG 4 AAC HE V1/2*, MPEG 2/4 AAC LC*#, MPEG 4
AAC LD/ELD*#
Enabling Professional Audio over IP Networking
• Support for Unicast, Multiple Unicast & Multicast applications for
flexible IP configuration
• EBU N/ACIP Compliant with support for SIP/SDP protocols
• Robust connection under stressed network conditions and super
fast, automatic reconnection if a link is dropped
• Allows management of network conditions such as packet size,
buffers and QoS levels for optimum audio performance
Providing a Robust Professional Platform
• Audio Back up from SD card - single file#
• Adjustable Silence Detection with alarm output
• DSP-based architecture for 24/7/365 reliability
• Fast Boot-up time for mission-critical environment
• Redundant Power Supplies*
Physical
Audio
Full Control & Monitoring
• Front panel operation with LCD screen & Keypad*#
• Highly Intuitive Network Management Software (NMS)#
• Embedded Web Server for access & control from any location
• Support for SNMP, Alarm & Event Logging#
• User-definable access levels and logins maintain network
security
*Cost Option - # Contact us for availability of these features
8
•
•
•
•
•
•
Fully N/ACIP compliant#
VLAN Tagging#
Auto-detection of
incoming IP stream#
Embedded Auxiliary data
for transmission of RBDS
/ RDS or PAD#
Up to 4 Opto-coupled
Inputs and up to 4 Relay
Outputs
Audio Backup from SD
card
Fast boot time for
mission-critical
environments
Support for SNMP, Alarm
& Event Logging#
Intuitive web interface
Technical Specification
Interfaces
Ensuring Extensive Connectivity
• High quality audio transport over IP networks using RTP/UDP and
SIP/SDP
• Dual IP ports configurable for back up#
• Embedded Auxiliary data for transmission of RBDS / RDS or PAD#
• Up to 4 Opto-coupled Inputs and up to 4 Relay Outputs
• USB port for Remote Control of Legacy Serial Equipment*#
•
•
•
Size
1U x 19” Rackmount
Dimensions
44mm x 480mm x 160mm - 1.73” x 19” x 6.3”
Weight
<1.5 Kg / < 3.35 lbs
Power Supply
100-250VAC, 50-60Hz
Power Consumption
<20 W
Environmental
0°C to +55°C, 95% humidity
IP
Ethernet 2 x RJ45, RTP/UDP, SIP/ SDP, SHOUTcast, Icecast
Aux Data
9 pin D Type, RS232 level
Data Rates (embedded)
1200, 2400,4800, 9600 Baud
Data Rates (via IP)
1200, 2400, 4800, 9600, 19200 Baud
Control
Web Interface, SNMP
Digital Audio I/O
AES/EBU, Balanced XLR-3, Impedance 110 Ω
Digital Ref Input
XLR-3
GPIOs
15 way D Type, NO/NC contacts
Digital Operation
32 kHz, 44.1 kHz, 48 kHz, 96 kHz
Digital Audio Bandwidth
10Hz through to 22.5kHz mono & stereo
Sample Rate Converter
8:1 (with bypass modes)
Standard Coding
Enhanced apt-X 16-Bit & 24-Bit,
Linear PCM 16, 20 & 24 Bit , MPEG Layer 3 CBR#, MPEG 4
AACHEv1/2*
Compression Ratio
apt-X: 4:1 - others: variable
Coding Delay
Enhanced apt-X 1.9ms - @48kHz Fs#
Dynamic Range
16 Bit > 85dB - 24 Bit > 120dB
Phase Response Linear
DC to Fs/2
Pass Band Ripple
< 0.2dB
WorldCast IP Decoder
Receiving Broadcast Quality Audio over Inexpensive IP Links
The WorldCast IP Decoder is a fully featured IP Audio Decoder
designed for use at transmitter sites and stereo drop-off points. It
also features our revolutionary “SureStream”technology.
Key Features:
See page 4 for details
Designed upon a robust DSP-based platform with dual IP ports and
redundant internal power supplies*, the IP Decoder provides precisely
the reliability and peace of mind that broadcasters require in a remotely
located unit.
The IP Decoder is compatible with all APT codecs as well as all N/ACIP
compliant codecs so can fit into any existing network as an affordable
alternative to a full codec. As the unit is typically located at a remote
location, the IP Decoder provides extensive control and monitoring
capabilities to manage both your audio, data and network conditions as
well as other equipment located at the remote site.
•
Professional stereo IP audio decoder at an affordable price
Wide range of algorithms: Enhanced apt-X algorithm, Linear PCM &
MPEG Layer 3 CBR#
Support for MPEG 4 AAC HE V1/2#
•
Auto-detection of incoming
IP stream#
•
Professional & Affordable IP
audio decoder
•
•
SureStream technology
enables broadcast-grade
audio over inexpensive IP
links*
Embedded Auxiliary data for
transmission of RBDS / RDS
or PAD
•
Up to 4 Opto-coupled Inputs
and up to 4 Relay Outputs
•
Audio Backup# from SD
card
•
Fast boot time for missioncritical environments
•
Support for Audemat’s
ScriptEasy software
suite enabling scripting,
telemetry and SNMP
control of all IP-enabled
units at remote site#
•
Support for SNMP, Alarm &
Event Logging
•
Intuitive web interface
•
Dual IP ports configurable
for back up
•
DHCP for automatic
configuration of IP
connection settings
•
Large range of algorithms
supplied as standard#
•
Non-Destructive, CascasdeResilient Coding with
Enhanced apt-X
•
Fully N/ACIP compliant#
•
VLAN Tagging#
Delivering Exceptional Audio Quality
•
•
! le Now !
w
e
N ailab
Av
Ensuring Extensive Connectivity
•
•
•
•
Decodes high quality audio over IP networks using RTP/UDP and SIP/
SDP
Dual IP ports configurable for back up#
Embedded Auxiliary data for transmission of RBDS / RDS or PAD
Up to 4 Opto-coupled Inputs and up to 4 Relay Outputs
USB port for Remote Control of Legacy Serial Equipment*#
Technical Specification
Physical
•
Enabling Professional Audio over IP Networking
•
•
•
•
•
Support for Unicast & Multicast applications for flexible IP
configuration
EBU N/ACIP Compliant with support for SIP/SDP protocols#
Auto-detection of incoming IP stream#
VLAN Tagging #
Robust connection under stressed network conditions and super fast,
automatic reconnection if a link is dropped
Allows management of network conditions such as packet size, buffers
and QoS levels for optimum audio performance
Interfaces
•
Providing a Robust Professional Platform
•
•
•
•
•
Audio Back up from SD card - single file
Adjustable Silence Detection with alarm output
DSP-based architecture for 24/7/365 reliability
Fast Boot-up time for mission-critical environment
Redundant Power Supplies*
Full Control & Monitoring
Front panel operation with LCD screen & Keypad*#
Highly Intuitive Network Management Software (NMS) #
Embedded Web Server for access & control from any location
Support for SNMP, Alarm & Event Logging
User-definable access levels and logins maintain network security
Remote Control of SNMP or Legacy Serial Equipment
Audio
•
•
•
•
•
•
WorldCast IP Decoder: Back Panel View
9
Size
1U x 19” Rackmount
Dimensions
44mm x 480mm x 160mm - 1.73” x 19” x 6.3”
Weight
<1.5 Kg / < 3.35 lbs
Power Supply
100-250VAC, 50-60Hz
Power Consumption
<20 W
Environmental
0°C to +55°C, 95% humidity
IP
Ethernet 2 x RJ45, RTP/UDP, SIP/ SDP*, SHOUTcast, Icecast
Aux Data
9 pin D Type, RS232 level
Data Rates (embedded)
1200, 2400,4800, 9600 Baud
Data Rates (via IP)
1200, 2400, 4800, 9600, 19200 Baud
Control
Web Interface, SNMP*
Digital Audio Output
AES/EBU, Balanced XLR-3, Impedance 110 Ω
Digital Ref Input
XLR-3
GPIOs
15 way D Type, NO/NC contacts
Digital Operation
32 kHz, 44.1 kHz, 48 kHz, 96 kHz
Digital Audio Bandwidth
10Hz through to 22.5kHz mono & stereo
Sample Rate Converter
8:1 (with bypass modes)
Standard Coding
Enhanced apt-X 16-Bit & 24-Bit,
Linear PCM 16, 20 & 24 Bit , MPEG Layer 3 CBR, MPEG 4 AAC
HEv1/2#
Compression Ratio
apt-X: 4:1 - others: variable
Coding Delay
Enhanced apt-X 1.9ms - @48kHz Fs
Dynamic Range
16 Bit > 85dB - 24 Bit > 120dB
Phase Response Linear
DC to Fs/2
Pass Band Ripple
< 0.2dB
*Cost Option - # Contact us for availability of these features
WorldCast Equinox
Professional Audio over IP, X.21 & ISDN
WorldCast Equinox is a multi-algorithm, fully duplex,
stereo audio codec offering IP, ISDN and X.21 / V.35 connections.
Designed primarily for studio to transmitter links and inter-studio networking applications, the WorldCast Equinox provides a
reliable platform for the transport of broadcast grade audio with 24/7/365 reliability.
Technical Specification
Delivering Exceptional Audio Quality
•
•
•
•
Fully duplex stereo audio codec with available audio bandwidths from 10Hz to 22.5kHz
Analog and AES/EBU I/O (AES3), Digital Reference In
Simultaneous Analog and Digital Outputs
Standard algorithms include: Linear PCM (16 bit and 24 bit resolution),Enhanced aptX algorithm (16 bit & 24 bit), MPEG Layer II, MPEG Layer III, G.711 and G.722
Advanced Algorithm Pack* includes: MPEG 4 AAC LC, AAC LD, AAC ELD & AAC HEv1/
v2
1U x 19” Rackmount
Dimensions
44mm x 480mm x 360mm
1.75” x 19” x 14.2”
Weight
3.5Kg / 7.7lbs
Power Supply
100-250VAC, 50-60Hz
(optional 48VDC Supply)
Power Consumption
<25W
Environmental
0°C to +55°C, 95% humidity
IP
Ethernet 1 x RJ45, (optional 2nd port)
RTP/UDP, SIP/ SDP
X.21/V.35
15 Way D Type DTE, RS422 levels
Rates 64-576 kbit/s
ISDN
2x RJ45 - standard
Aux Data
9 pin D Type, RS232 level
Data Rates (embedded)
1200, 2400,4800, 9600 Baud
Data Rates (via IP)
1200, 2400, 4800, 9600, 19200 Baud
Control
NMS Software & SNMP
Optional web browser & front panel
Digital Audio I/O
AES/EBU, Balanced XLR-3,
Impedance 110 Ω
Digital Ref Input
XLR-3
Providing a Robust Professional Platform
Analog Audio I/O
•
•
•
•
Balanced XLR-3,
Input Impedance >10k / 600 Ω
Output Impedance <50 / 600 Ω
SD
Support for SD & SDHC Cards
USB
Type A USB Connector
Alarms
8 Alarms on 25 pin D Type, NO/NC
contacts
Opto-Coupler
25 way D Type, 8 Inputs driving
4 Relay Outputs
Digital Operation
32kHz, 44.1kHz, 48kHz
Digital Audio Bandwidth
10Hz through to 22.5kHz mono & stereo
Analog Operation
8kHz-48kHz
Analog Audio Bandwidth
10Hz through to 22.5kHz mono & stereo
Sample Rate Converter
3:1 (with bypass modes)
A/D Converter
24-bit/96kHz sigma-delta
Standard Coding
Linear, Enhanced apt-X 16-Bit & 24-Bit,
ISO/MPEG 1/2 Layer III, , G.711, G.722
Algorithm Options
ISO/MPEG 4 AAC LC
ISO/MPEG4 AAC-LD & AAC-ELD
AAC HEv2
Compression Ratio
apt-X: 4:1 - others: Variable
Coding Delay
Enhanced apt-X 1.9ms @48kHz Fs
Dynamic Range
16 Bit > 85dB, 24 Bit > 120dB
Phase Response Linear
0 to Fs/2
Pass Band Ripple
< 0.2dB
•
Physical
Size
Ensuring Extensive Connectivity
• High quality audio transport over IP networks using RTP/UDP and SIP/SDP
• Dual IP Ports for separate streaming & control*
• ISDN interface with Mucas Bonding & L3 Telos Bonding
• CCS, H221 & Hitachi ISDN Bonding
• X.21/V.35 interface enables easy connection to permanent digital links
• Embedded Auxiliary data for transmission of RBDS / RDS or PAD
• Up to 8 Opto-coupled Inputs and up to 8 Relay Outputs
•
USB port for Remote Control of Legacy Serial Equipment*
•
•
•
EBU N/ACIP Compliant with support for SIP/SDP protocols
“AutosyncTM” feature of Enhanced apt-X ensures robust connection under stressed
network conditions and super fast, automatic reconnection if a link is dropped
Allows management of network conditions such as packet size, buffers and QoS
levels for optimum audio performance
Interfaces
Enabling Professional Audio over IP Networking
• Support for Unicast & Multicast applications for flexible IP configuration.
Redundant Power Supplies*
Bandwidth Flexing (keeps link alive using minimal bandwidth when silence detected)
Adjustable Silence Detection with alarm output
Static Audio Back-Up from SD Card
Full Control and Monitoring
Front panel operation with level meters and headphone socket*
Highly Intuitive Network Management Software (NMS) (see page 19)
Embedded Web Server for access & control from any location*
Support for SNMP, Alarm & Event Logging
User-definable access levels and logins maintain network security
*Cost Option #Future Feature
WorldCast Equinox: Back Panel View
Analog In
L
R
Audio
•
•
•
•
•
AES/EBU Digital
Alarm USB Port X.21 / V.35 Debug
Port
Port
In
Reference In Ports
L
R
Analog Out
AES/EBU
Out
Opto-Inputs
ISDN
SD Card
IP Ports Slot Aux Data
Port
10
Fused Power
Sockets
WorldCast Astral
Flexible & Reliable IP STL Platform
The industry’s most versatile and professional low cost IP
audio codec, the WorldCast Astral is a rock-solid IP STL
platform offering a full complement of professional audio delivery algorithms. Thanks to the unit’s modular architecture, broadcasters
can then bolt-on additional functionality to ensure that the unit meets their broadcast needs.
Options include additional algorithms for optimum compatibility, dual IP ports, redundant Power Supplies, embedded Webserver,
remote control of multiple third party units at Studio and /or Transmitter Site, and a sophisticated Audio Back-Up Suite using SD
card or SHOUTcast streaming.
Technical Specification
Delivering Exceptional Audio Quality
• Fully duplex stereo audio codec with available audio bandwidths from 10Hz to 22.5kHz
• Analog and AES/EBU I/O (AES3), Digital Reference In
• Simultaneous Analog and Digital Outputs
• Standard algorithms include: Linear PCM (16 bit and 24 bit resolution),Enhanced apt-X
algorithm (16 bit & 24 bit), MPEG Layer II, MPEG Layer III, G.711 and G.722
• Advanced Algorithm Pack* includes: MPEG 4 AAC LC, AAC LD, AAC ELD & AAC HEv1/v2
1U x 19” Rackmount
Dimensions
44mm x 480mm x 360mm
1.75” x 19” x 14.2”
Weight
3.5Kg / 7.7lbs
Power Supply
100-250VAC, 50-60Hz
(optional 48VDC Supply)
Power Consumption
<20W
Environmental
0°C to +55°C, 95% humidity
IP
Ethernet 1 x RJ45, (optional 2nd port)
RTP/UDP, SIP/ SDP
Aux Data
9 pin D Type, RS232 level
Data Rates (embedded)
1200, 2400,4800, 9600 Baud
Data Rates (via IP)
1200, 2400, 4800, 9600, 19200 Baud
Control
CMS Software & SNMP
Optional web browser
Digital Audio I/O
AES/EBU, Balanced XLR-3,
Impedance 110 Ω
Digital Ref Input
XLR-3
Analog Audio I/O
Providing a Robust Professional Platform
• Redundant Power Supplies*
• Bandwidth Flexing (keeps link alive using minimal bandwidth when silence detected)
Balanced XLR-3,
Input Impedance >10k / 600 Ω
Output Impedance <50 / 600 Ω
SD
Support for SD & SDHC cards
•
•
USB
Type A USB Connector
Alarms
8 Alarms on 25 pin D Type, NO/NC
contacts
Opto-Coupler
25 way D Type, 8 Inputs driving
4 Relay Outputs
Digital Operation
32kHz, 44.1kHz, 48kHz
Digital Audio
Bandwidth
10Hz through to 22.5kHz mono & stereo
Analog Operation
8kHz-48kHz
Analog Audio Bandwidth
10Hz through to 22.5kHz mono & stereo
Sample Rate Converter
3:1 (with bypass modes)
A/D Converter
24-bit/96kHz sigma-delta
Standard Coding
Linear, Enhanced apt-X 16-Bit & 24-Bit,
ISO/MPEG 1/2 Layer III, , G.711, G.722
Algorithm Options
ISO/MPEG 4 AAC LC
ISO/MPEG4 AAC-LD & AAC-ELD
AAC HEv1/v2
Compression Ratio
apt-X: 4:1
others: Variable
Coding Delay
Enhanced apt-X 1.9ms @48kHz Fs
Dynamic Range
16 Bit > 85dB, 24 Bit > 120dB
Phase Response Linear
0 to Fs/2
Pass Band Ripple
< 0.2dB
Physical
Size
Enabling Professional Audio over IP Networking
• Support for Unicast & Multicast applications for flexible IP configuration.
• EBU N/ACIP Compliant with support for SIP/SDP protocols
• Autosync TM” feature of Enhanced apt-X ensures robust connection under stressed
network conditions and super fast, automatic reconnection if a link is dropped
• Allows management of network conditions such as packet size, buffers and QoS levels
for optimum audio performance
Interfaces
Ensuring Extensive Connectivity
• High quality audio transport over IP networks using RTP/UDP and SIP/SDP
• Dual IP Ports for separate streaming & control *
• Embedded Auxiliary data for transmission of RBDS / RDS or PAD
• Up to 8 Opto-coupled Inputs and up to 8 Relay Outputs
• USB port for Remote Control of Legacy Serial Equipment*
Adjustable Silence Detection with alarm output
Static Audio Back-Up from audio files on SD Card
Full Control & Monitoring
• Front panel operation with level meters and headphone socket*
• Highly Intuitive Network Management Software (NMS) (see page 19)
• Embedded Web Server for access & control from any location*
• Support for SNMP, Alarm & Event Logging
• User-definable access levels and logins maintain network security
*Cost Option #Future Feature
Analog In
L
R
AES/EBU Digital
Alarm USB Port
In
Reference In Ports
L
R
Analog Out
AES/EBU
Out
Opto-Inputs
IP Ports
SD Card
Slot
Audio
WorldCast Astral: Back Panel View
Debug
Port
Fused Power
Sockets
Aux Data
Port
11
WorldNet Oslo Audio Multiplexer
The Professional Approach to Audio and Data Networking for Broadcast
The WorldNet Oslo is the industry’s most robust, professional
platform for high quality FM, HD Radio/DAB and Surround
Sound audio contribution & distribution.
Multiple channels of both linear and compressed audio can be
combined with data and voice channels onto a single digital
link for cost-effective networking.
With a modular, single-platform approach, a wide variety of plug-in modules, multiple layers of redundancy and a highly acclaimed
Control Interface, it is the perfect solution for STL, TSL, RPU, backhaul and studio linking applications.
Outstanding Audio Quality
Round the Clock Reliability
With choices including pure linear audio or high quality Enhanced
apt-X coding, the WorldNet Oslo will make your station sound simply
outstanding!
The WorldNet Oslo has no single point of failure and can be configured
to provide multiple layers of redundancy ensuring your station stays on
air even under the most stressful network conditions (see page 17 for
more details).
•
•
•
•
•
Available audio bandwidths from 10Hz through to 22.5kHz, for
FM and HD Radio/DAB applications
Analog and AES/EBU I/O (AES3) Audio
Low Delay, Cascade-resilient Enhanced apt-X coding as standard
Optional pure, uncompressed Linear audio for uncompromised
audio quality
Optional coding & companding options including MPEG I/II Layer
2, J.57 and J.41
Professional Platform, Advanced
Technology
The WorldNet Oslo is a sophisticated, professional solution
offering a wealth of advanced features for many different broadcast
environments.
•
•
•
•
•
•
Highly acclaimed Codec Management Software enables
extensive monitoring ability and intuitive control over network,
equipment and audio variables
SIP/SDP/RTP protocols are supported according to the Tech 3326
standard for IP compatibility defined by the N/ACIP workgroup
within the EBU
Cross-connect functionality on T1 and E1 links enables advanced
network features such as drop and insert, drop and copy, and
backup schemes utilizing powerful Time Slot Management
techniques
LAN extension allows use of surplus bandwidth to connect
unnetworked sites
In-band Management eliminates need for overlay control
network
Phase-lock audio option enables true Surround Sound image
across multiple channels
•
•
•
•
•
•
•
DSP-based architecture for 24/7/365 operation
Redundant Power Supplies with power condition detector &
backup switch
Passive Backplane with no active components
Automatic switching to secondary transport link and/or spare
audio card with automatic restore on user-defined conditions,
rules and presets
Hot-swappable cards enable uninterrupted audio Independent
Master Controller Card safeguards system configuration
User-configurable suite of audio, link, sync and PSU alarms
SureStream
“Perfect Audio over Imperfect IP”
•
•
•
•
•
SureStream is an exclusive technology from APT
SureStream technology delivers the audio quality and reliability
you expect from a T1/E1 link at a fraction of the associated cost
SureStream utilizes inexpensive IP links such as wireless 3G and
4G IP networks, LAN, WAN, Wi-Fi and simple ADSL.
SureStream protects your audio content from loss of connection &
dropped packets
SureStream employs advanced resequencing technology to offer
you seamless streaming of high quality audio.
Highly Flexible, Highly Customizable
There is no need for major capital investment when adding additional
programming, migrating from synchronous to IP or layering in new
back-up networks; simply add new modules to the existing WorldNet
Oslo framework.
•
•
•
•
Over 15 different plug-in audio modules including analog, digital,
duplex, encoder, decoder and 5.1 phase-locked options
Up to 4 channels per audio module and up to 6 audio modules
per frame enabling from 2 duplex channels to 24 simplex
channels in a single frame
Selection of transport modules include T1 (1.5Mbit/s), E1 (2Mbit/
s) and Ethernet/IP
Auxiliary Data Contribution modules
12
See page 4 for further details
Architecture & Options
Based around a 19-inch 3U high standard rackmount chassis, the
WorldNet Oslo is modular in design and built to customer requirements.
Users can choose from a variety of “hot swappable” modules to fit their
particular audio, voice, data and transport needs.
The modular design together with inherent and optional failsafe
mechanisms ensure that the WorldNet Oslo provides the ability to
deliver audio under extreme circumstances.
Primary
Transport
Module
Audio / Aux Data / Voice / Video Transport Modules
Secondary
MCU /
Transport
System
Module
Controller
As the unit is designed around a mid-plane architecture, each module
is supplied as a pair with a front panel indicator plug-in card and a
corresponding interface module.
Redundant
AC/DC
Power
Supplies
Master Controller (MCU) Module
The MCU module is responsible for the configuration and monitoring of the complete WorldNet Oslo system.
•
•
•
•
•
System Control - The MCU module controls the operation of all other modules in the system,
allocating them time slots on the backplane and sending configuration information.
Status Monitoring - The processor constantly polls the status of individual transport modules and
makes failsafe decisions based on this status. The module also monitors the frame temperature and
the power supplies of the system.
Hardware Alarms - The alarm port on the MCU module provides seven GPIO relay contact closures in
order to indicate critical status and alarm conditions.
Hot-Swappable - Once a WorldNet Oslo is configured, it is able to continue to operate in the event of an MCU card failure as each card
will hold its local settings. The MCU can then be “hot swapped” and will recover the current system settings from the individual modules
In-band Management - The IP interface on the MCU module is the communication interface for the NMS (Network Management
Software) software. It also provides the ability to control multiple remote WorldNet Oslo frames throughout a network using in-band
management.
T1/E1 Transport Module
IP Transport Module
The T1/E1 module is one of the most
common choices for audio transport
due to the reliability and widespread
availability of T1 and E1 circuits. There
are two physical slots to accommodate
transport modules on the WorldNet Oslo:
one for the main circuit and a second for
backup, bridging or crossover of up to
four lines. The E1 module is available with both RJ45 and
BNC connectors.
The IP interface module is the other main
alternative for audio and data transport.
Again, two cards can be fitted, one for the
main circuit and a second for backup or
a second link.
•
•
•
•
Line Redundancy - Each module comprises two T1/E1 framer
devices and two line interfaces enabling line redundancy.
Drop & Insert - The T1/E1 module provides a failsafe circuit that,
in the event of total failure of the module, connects the upstream
T1/E1 circuit to the downstream T1/E1 circuit, thereby preserving
the integrity of the data path to any other devices on the network
ring
Hot Standby Module Backup - The T1/E1 module also provides an
intercard high speed connection to indicate its health condition
to the second stand-by multiplexer (if fitted). In the event of bad
status, the backup module takes over the whole payload from the
active device, minimizing disruption.
Contribution Data & Clocking - Any data received by the X.21 port
is then placed onto the T1/E1 transport link. In combination with
the external clock input, this module provides versatile system
timing options to connect to almost any network.
•
•
•
•
•
•
13
Powerful Processing - offers a
powerful high performance Network
Processing Engine capable of supporting many complex IP routing
infrastructures.
Multiple Streams to Multiple Destinations - Up to 24 real-time
audio streams to 24 individual destinations can be set up on the
IP module’s streaming table. The unit can be configured for pointto-point, multiple unicast and multicast applications.
Low, Stable Latency - An on-board VCXO on all audio modules
enables clock synchronization and stable data flow across IP
links. Alternatively, the IP module can be connected to an external
AES/EBU or GPS reference clock to ensure synchronized delivery
without introducing delay.
Quality of Service - The module supports QoS settings to ensure
optimum audio performance over packetized networks.
IP Backup - The system offers automatic backup using either
the second IP port of the IP interface module or a redundant IP
interface module.
Bridging IP Networks and T1/E1 lines - The WorldNet Oslo
enable progressive migration from synchronous to asynchronous
networks including bridging content across both worlds.
WorldNet Oslo Contribution Modules
Audio Modules
There are over 15 different varieties of pre-configured, plug-in audio modules for the WorldNet Oslo. All modules provide an
interface I/O board with standard XLR connectors. With up to 6 physical slots, a maximum of 24 simplex and 12 duplex connections
are possible on a single frame.
Flexible Analog or AES/EBU
Configurations
Each module offers four audio
channels which can be either four
encode, four decode or two encode
and two decode. Offering a high
level of flexibility and channel
density, the simplex modules can
operate either as dual stereo or
quad mono cards.
Card Variations: Fully Duplex Audio Modules
Channel Routing
On both synchronous and IP networks, individual channels can
be enabled and disabled or routed to different destinations.
For example, on a T1/E1 link, stereo A can go to the main
T1/E1 link and stereo B on the secondary / drop and insert
circuit. On an IP link with 4 simplex channels, channels A&B
could be sent to one transmitter location and channels C &D
to a local studio. On IP networks, the inputs of audio cards can
also be sent together to multiple destinations in multicast or
multiple unicast applications.
Broadcast Grade Audio
Exceptional audio quality can be delivered using either
Enhanced apt-X audio coding or the pure linear PCM/
uncompressed option. The Enhanced apt-X audio format
provides an operational audio resolution depth of either
16 bits or 24 bits while the digital circuits of the audio module
run at a resolution of 24 bits (A/D – D/A converters, audio
DSP etc). Using low delay, Enhanced apt-X at high quality
16-Bit resolution, the WorldNet Oslo can combine up to 6
fully duplex 15kHz stereo programs on a single T1 line or 7
on an E1 line.
Digital
Duplex
Analog
Duplex
Simplex Audio Modules
Analog
Encoder
Analog
Decoder
Digital
Encoder
Digital
Decoder
The diagram below shows how the Duplex Module with Analog
Input Back-up ensures a seamless switchover should the main
audio input fail.
Extended Interoperability
Other coding and companding options available include
G.711, MPEG 1/2 Layer 2, J.57 and J.41. These options provide
the ability to employ the WorldNet Oslo in networks equipped
with third-party codecs.
Phase-Matching for Surround Sound
6 duplex audio channels or 12 simplex audio channels,
provided by three audio cards of a kind, can be phase locked
in order to provide a true surround sound image for 5.1 Radio
Broadcasts.
Audio Back-Up
Additional variants of audio module are also available to
provide increased levels of redundancy and back-up. These
include a Digital Duplex card with Simultaneous Analog
Outputs and a Digital Duplex card with Automatic Input Backup with Analog Inputs.
14
Aux Data Modules
Voice Transport Modules
The Auxiliary Data module, available in several different
configurations, is designed to be plugged into any of the
contribution module slots in the WorldNet Oslo frame.
Several Voice Transport modules are available enabling
broadcasters to combine voice channels with both audio and
data on a single digital link.
The Wideband Voice Communications module (pictured) offers
four discrete full duplex channels utilizing 16 bit Enhanced
apt-X coding at a sample rate of 16kHz (7.5kHz audio
bandwidth – 64kbps per channel).
There are two builds of this card – one offering 10 serial data
ports and no GPIO connections and the second offering 5
serial data port and a set of 8 relay contact closures and optoisolated switch inputs. Both builds allow up to 9600 bauds per
data port.
To enable OPX (Off-Premises extension) applications, 2-wire
FXO, 2-wire FXS and 4-wire E&M modules are also available.
In addition, the Auxiliary Data Card provides all logical devices
and processor capacity to support the entire data subsystem
including all optical switch inputs. The main board is designed
to handle both asynchronous and synchronous data in different
interface formats, such as RS232 and RS422.
The 2-wire modules offer four discrete duplex channels on 4
x RJ11 with support for G.711 A-Law, µ -Law (3.4kHz audio
bandwidth), Ground start or Loop start and E1 CAS / T1 RBS.
The 4-wire module presents 4 x RJ45 and supports G.711 ALaw, µ -Law, Type I, II & V signalling and E1 CAS / T1 RBS.
Applications
The flexibility of the WorldNet Oslo’s modular architecture means that it can be deployed throughout many different network
topologies and architectures. Whether you require AM, FM, HD RadioTM or Surround Sound broadcasts, packetized or synchronous
transport, point-to-point or drop and insert configurations or bidirectional or simplex connections, the WorldNet Oslo can be
tailored to your exact requirements.
Designed to provide extreme resilience for mission-critical applications and quick recovery from network disruptions and power
failures, the WorldNet Oslo is the ideal choice for linking studios and transmitter sites over T1, E1 and IP links. In addition to
bidirectional audio transport, the unit can combine serial and LAN data, RBDS and other traffic on digital links for increased
network efficiency. For HD Radio™ applications, WorldNet Oslo can also deliver HD Exporter streams and Program Associated
Data (PAD) across the digital link.
In
this
application,
the
regional studio is using the
station’s private LAN to deliver
contribution audio to the main
station.
This bi-directional
connection also enables the
distribution of the main station’s
content via satellite to the
Regional FM Transmitter.
The main station’s transmitter
network is operating over a
conventional T1/E1 ring network
in drop and insert mode. This
configuration provides a failsafe
circuit to safeguard against
network and system failures.
15
Redundancy on the WorldNet Oslo
Designed for mission-critical broadcast applications, the WorldNet Oslo offers a
suite of back-up and redundancy options to protect against system failures and
keep your station on air 24/7/365.
3)
Transport Card Redundancy
As there are two transport slots on the WorldNet Oslo, a second transport card
can be fitted for additional redundancy and main circuit protection. In a crossmedia configuration, this can also be used to bridge data from an IP network to
a TDM line and vice versa.
4)
PSU Redundancy
Provision is made for a fail-safe power system using redundant power supplies
in each WorldNet Oslo frame. The WorldNet Oslo provides a power condition
detector and backup switch, in order to initiate the second PSU in the case of
any failure.
Physical
Telecom
2)
Line Back Up
Both the T1/E1 and IP transport modules have dual ports on each module
enabling an automatic switchover to a secondary line in the case of failure on
the main link. This fail safe option will also ensure an automatic switch back
to the main line when it has been resurrected for a user-specified amount of
time.
6)
Backup Monitor
An optional feature of the NMS, Network Management Software (see page 19),
the Backup Monitor is a highly sophisticated and customisable feature which
enables a broadcaster to continually monitor the status of both transport and
audio circuits across multiple WorldNet Oslos throughout a network. For each
defined alarm condition, the user will specify a combination of corresponding
rules and priorities that will trigger the reconfiguration of units and switching
to an appropriate back-up scenario using the inbuilt preset system.
16
Audio
The AC power supplies will share the load continuously, the redundant DC power
supply functions as a hot standby. Any combination of AC/AC, DC/DC or AC/DC
PSU types is possible.
5)
MCU Preset System
A preset system within the MCU module will store and recall up to thirty different
system configurations. The Preset System reads the configuration of all cards
and all parameters like a snap shot of the entire WorldNet Oslo frame and
stores it on the MCU memory. The MCU also allows a user to store (backup) the
entire preset system on a PC hard drive. A Preset can be restored locally or can
be applied to a remote WorldNet Oslo via an inband link.
3U x 19” Rackmount
Dimensions
133mm x 482mm x 430mm
5.25” x 19” x 17”
Ancillary Data & Control
1)
Hot Swappable Cards
Based on a modular architecture, all WorldNet Oslo modules are “hot swappable”
and can be changed without causing a glitch on any of the audio channels or
affecting them in any way. Once the system is configured, it is able to continue
to operate in the event of an MCU card failure as other cards in the frame will
each hold their local settings. The MCU can then be “hot swapped” and will
recover the current system settings from the other cards. This prevents system
down time and ensures that there is no single point of failure on the WorldNet
Oslo.
Technical Specification
Size
Weight
9Kg / 19.8lbs
AC Power Supply
90 - 260 VAC, 47 - 60Hz
DC Power Supply
36 to 72 V DC
Power Consumption
<200W
Environmental
-10°C to +45°C
Humidiity
Up to 95%
E1
G.703, 2.048Mbit/s, 32 Duplex
DS0s
RJ45 Balanced, 120/100 Ohms
Termination or BNC, 75 Ohms
T1
T1.102, 1.544Mbit/s, 24 Duplex
DS0s
RJ45 Balanced, 120/100 Ohms
Termination
IP
10BaseT, RJ45, RTP / UDP, QoS
Aux Data
Up to 10 channels (1 channel/
stereo pair)
D type, RS232 / RS422
Data Rates
1200, 2400,4800, 9600, 19200 Baud
Control I/O
8 TTL switch inputs / 8 Contact
Closures
Alarms
15 pin D type, 7 Relays,
3 Contacts Per Relay
Control In
10/100BaseT Ethernet (RJ45),
Network Management System,
SNMP (optional)
Audio Input / Output
Analog, AES/EBU
Sampling Frequencies
32, 44.1 & 48kHz
Audio Bandwidth
10Hz-22.5kHz
Analog Mode
Balanced XLR-3
I/P Impedance
>24k/600 Ohms, Symmetrical
O/P Impedance
<100/600 Ohms, Symmetrical
Digital Mode
Balanced XLR-3 / Unbalanced BNC
Impedance
110 Ohms / 75 Ohms
Digital Ref In
Balanced XLR-3
Source Coding
Enhanced apt-X 16-bit & 24-bit, Linear,
J.57, J.41, G.711, MPEG L2
Compression Ratio
Enhanced apt-X: 4:1
Others: variable
Coding Delay
Enhanced apt-X: 2ms @ 48kHz
Dynamic Range
16bit>85dB, 24bit >110dB
AoIP NextGen Module
For the WorldNet Oslo Platform
The new AoIP NextGen card for the WorldNet Oslo platform
combines audio, IP transport, management and auxiliary
data onto a single, plug-in module. This enhances the Audio
over IP performance of the Oslo unit as well as increasing its
scalability and flexibility.
About the WorldNet Oslo:
The WorldNet Oslo is the industry’s most robust,
professional platform designed to transport multiple voice,
data, high-fidelity compressed or uncompressed audio
and other types of payload data within IP networks.
Fully compatible with the
many hundreds of existing
Oslo units already deployed
worldwide,
each
AoIP
NextGen card can deliver
four independent audio IP
streams or many more using
multiple unicast.
Based around a 19 inch, 3U high standard chassis, the
Oslo’s modular, card-based approach provides substantial
flexibility to enable customization to current network
requirements and scalability for future expansion.
The AoIP NextGen module
offers the entire range of
audio formats and modes
meeting the audio industry’s
requirements:
simplex,
duplex, AES/EBU, AES/
EBU with analog backup,
analog with HI/LO or 600Ω
impedance.
Utilizing either linear audio or the Enhanced apt-X®
algorithm ensures that the Oslo platform delivers audio
content with exceptional musical fidelity. Many additional
audio codecs are also supported to enable interoperability
with other manufacturers’ equipment.
The WorldNet Oslo has no single point of failure and can
be configured to provide multiple layers of redundancy
ensuring your station stays on air even under the most
stressful network conditions.
The module offers the same broadcast-grade audio for which
the Oslo platform is renowned with support for a wealth of
standards such as: MPEG-I/II L2*, MPEG-4 AAC LC/LD/ELD*,
MPEG-4 AAC HEv1/v2*#, Enhanced ap t-X®, and Linear PCM
24Bit/48kHz.
FEATURES
BENEFITS
•
•
•
•
•
•
Transports up to four audio channels in a single stream
•
Individual clock domain per audio channel allows
seamless point-to-point and anywhere-to-anywhere
streaming
Offers a configurable delay-jitter buffer for each receive
IP stream (5 ms to 5000 ms)
•
•
Deliver up to 24 IP audio streams per module
Simplex and duplex operational modes
Point-to-Point and Point-to-Multipoint operation
Four Independent Clock Domains per module
Supports a variety of protocols including: UDP RTP/
RTCP, SIP/STUN#, SAP#, DHCP#, IGMP for Multicasting
mode, ICMP, VLAN Tagging, SNMP v2c, SNMP v3#
Supports “Diffserv” Quality of Service (QoS) on variable
DSCP values & Forward Error Correction (FEC)#
User selectable packet size for each IP route
•
•
•
•
Supports APT’s award winning SureStream Technology
for High Quality Audio over Open Internet Links
Enables Performance monitoring on each individual IP
stream
Can perform Diagnostics with PING on streaming via
both ports
Available on the the AoIP NextGen card, SureStream technology is an innovative and
award-winning new approach to transporting audio over contended IP networks. It
enables you to obtain the audio quality and reliability you expect from a T1/E1 link
at a fraction of the associated cost. See page 4
17
Technical Specification
Independent Channel Formats
Electronically balanced, capacitive isolated on 37 pin D-Type
connector
Audio channels
Simplex Mode: 4x Input or 4x Output
Duplex Mode: 2x Input and 2x Output (analog outputs are
simultaneously available in digital mode)
I/O impedance
High >10 kΩ/Low <50 Ω or 600 Ω
Modes of operation
Stereo, Dual Stereo, Quad Mono
Digital Interfaces:
AES-3, transformer balanced* on 37 pin D-Type connector
I/O impedance
AES-3: 110 Ω (on D-Type connector)
Audio channels
Simplex Mode: 2x AES Input or 2x AES Output
Duplex Mode: 1x AES Input and 1x AES Output
Modes of operation
Stereo or Dual mono
I/O sampling rates
32 / 44.1 / 48 kHz
Digital Ref Input
AES-11 reference input
with decoder mode: on second AES input
Audio Bandwidth
10 Hz – 22.5 kHz (-3 dB)
Digital Range
Up to >110 dB @ 24 Bit
Audio Formats &
Algorithms
Linear PCM, Transparent AES#, Enhanced apt-X®,
MPEG 1 L2#, MPEG 2 L2#, MPEG 2 L3#, MPEG 4 AAC LC#,
MPEG 4 AAC LD#, MPEG 4 AAC HEv2#
Aux Data
Up to 2 channels per module RS232
Aux Data Mode
Embedded# or Non embedded (depending on format/algorithm)
Data Rates
300/600/1200/2400/3600/4800/9600/19200/38400Baud
GPIO
4x opto-isolated switch inputs: ±7.5-30V DC
Relay Contacts
4x relay contacts via 2 pin switches
Physical IP Interface
Dual Ethernet, RJ45
IP Protocols Supported
IPv4, VLAN, DHCP#, ICMP, IGMP, TCP/IP, UDP, RTP/
RTCP, FTP, HTTP, HTTPS#, SNMP, SMTP#, VoIP#
Modes
Unicast, multiple unicast, multicast
Clock
Master, Slave, internal, external,
4 independent clock domains per module
De-Jitter Buffer
4x buffers per module (5 - 5,000ms) independently
adjustable
Streams per Module
4x audio channels x N streams (up to 24 streams per
module)
Quality of Service
DiffServ, with separate DSCP values per stream
FEC#(SMPTE 2022-1/3), SureStream Technology
Independent Clocking
Management
Codec Management Software, SNMP, API#
The four clock domains also eliminate the issues of clock
drift associated with streaming multiple channels over IP
to a single decoder and enable an anywhere-to-anywhere
streaming method on the network on a per channel basis.
Performance Monitoring
Performance Monitor per Stream. Data includes: Name of
Stream, Number of sent/received packets, Number of sent/
received bytes, Number of dropped packets, Percentage of
Re-Sequenzer activity, Loss of Connection events, Duplicated
packets, De-Jitter buffer size and actual level, SIP status#,
SIP errors#, SIP Redial counts#, RX port number, RX/TX
packet sizes, Physical port (ETH 0/1), Source IP Address,
Source IP port
On the AoIP NextGen module, the audio channels are
independently clocked by four separate clock domains, which
supports the sending and receiving of audio in different
formats.
Analog
Analog Interfaces:
IP
Data
Audio
With six cards per Oslo and 4 channels per card, the unit
is therefore able to decode up to 24 streams, even discrete
monos with independent bitrates and algorithms, from
independent locations.
Digital
This opens up several new applications: a program can be
sent in high quality with the return feed carrying voice-grade
audio or one program sent in broadcast quality with a second
program set to the highest quality for content contribution.
Independent Channel Management
Control
AoIP NextGen Front & Rear Modules
*Also compatible with most 75 Ω unbalanced interfaces
#
Future Feature
The IP performance is increased yet further thanks to the
fact that each AoIP NextGen card handles its own IP traffic,
avoiding any bottlenecks in the system. On the network
receiving end, four de-jitter buffers allow independent stream
management on a per channel basis.
18
Multiple Routes
Each AoIP NextGen module can generate multiple
streams per stereo or mono signal. On multiple unicast
or multicast many different destinations can be supplied
with program audio from a single module. Currently up to
24 routes per card are supported.
Network Management Software
Supplied as standard with many WorldCast units, APT’s Network Management Software (NMS)
package is a powerful and intuitive graphical user interface which enables extensive remote
monitoring and management capability of your units deployed throughout a network.
Mode / Status View
Click on any unit to see detailed configuration and connection settings. Graphical
representation shows live status view of level bars and enables configuration of primary and
back-up connections as well as Master / Slave status.
Network Overview
At-a-glance view of multiple units
throughout your audio network
enables easy monitoring of multiple
sites and provides immediate
notification of alarms, alerts and
other network issues.
Web Browser
When a user choses the dual IP ports port
option, they automatically receive access
to the APT web browser. It is provided
as standard on the WorldCast Horizon
NextGen, IP Decoder and Stream In /
Stream Out Silver.
The web browser application delivers the
full functionality of the NMS but with the
added benefit of being accessible from
any location using only a web browser.
Connection Settings
Configure your IP network settings to
transmit or receive unicast, multicast
or multiple unicast streams. You
can also set packet size, QoS and
buffer levels to compensate for
network jitter. Store up to 150 fully
configured IP Speed Dials for simple,
fast connection to frequently-used
locations. X.21/V.35 and ISDN settings
can also be extensively managed
through this screen.
Audio Settings
This screen enables you to
configure audio alarm levels
and time-outs, set algorithms
and data rates and establish
pre-set configurations (audio
profiles) which can be assigned to
connection speed dials for ‘oneclick connection’.
SIP Dialer
Alarm and Event Logs
View and save logs of all connection
activity and alarms through this
screen. Logs can be filtered by event
category e.g. major, minor or critical or
by individual codec.
Enabling ‘one click’ connection, the
SIP Dialer is an intuitive interface
enabling the simple setup and
teardown of SIP calls. It allows all
commonly used destinations and
settings to be stored in an address
book for easy access by non-technical
users.
19
Network Management Software for WorldNet Oslo
APT’s Network Management Software (NMS) for the WorldNet Oslo provides for control of the entire network from a single seat.
Audio Settings
Clicking on any audio module enables configuration of the algorithm,
resolution and bandwidth on that module. Level bars on each module also
provide easy indicators of level status.
Audio Level thresholds can be set for each of the four channels and other
alarms such as Loss of Digital Input and Autosync can also be activated.
Network Overview
The Network overview offers an at-a-glance indication of the condition
of all units throughout the network. The unit’s appearance reflects its
current status according to a suite of configurable alarms and alerts
and any unit can be selected for control and configuration.
IP Connection Settings
The NMS makes setting up complex
multicast applications as quick and painless
as a simple point-to-point connection.
Extensive configuration of IP parameters
such as packet size, jitter buffers and
QoS levels allows maximum control over
network conditions. Audio performance
can be monitored on a ‘per route’ or unit
basis.
For multi-transport networks, audio can be
bridged from T1 or E1 networks onto the
IP link.
T1/E1 Connection Settings
The T1 / E1 screen enables control
and configuration of both the main and
secondary (Drop & Insert) link as well as
the external clock. Users can configure
back-up options and bridge between the
T1/E1 and IP Links.
T1/E1 Timeslot Allocation
The NMS provides a powerful Timeslot
Allocation tool which enables audio and
data channels to be allotted space on both
the transmit and receive paths.
Drop and Insert applications can be easily
set up and amended as required.
New remote management functionality
employs contact closures to trigger
switching between national and localized
content using selected timeslots.
20
Alarms and Logs
The NMS enables all audio, unit
and network alarms that have been
set on the individual modules to be
grouped into Minor, Major and Critical
categories.
When an alarm condition occurs, the
alarm or alert status will be reflected
by a change in colour or flashing unit
on the network overview screen. The
user can also view, filter and export
extensive logs of network and unitrelated events.
Network Management Software
Designed specifically for the management of larger and more complex audio networks, APT’s Network Management System
(NMS) offers a complete solution to simplify the control & configuration of any mix of WorldNet Oslo or WorldCast
audio codecs.
The NMS allows you to manage extensive networks of codecs
by gathering them into smaller logical groups in a hierarchical
structure and provides additional tools which enable the user to
automatically configure multiple codecs.
Configuration Uploader
Eliminating the need to repeat timeconsuming tasks on individual codecs
throughout
large
networks,
the
configuration uploader enables the user
to quickly and easily upload firmware,
IP speed-dials, ISDN Speed-dials and
audio profiles to a large group of multiple
codecs.
Upload can be manually controlled or
scheduled for a specific date and time
to minimize disruption. The information
to be uploaded is retrieved from an SQL
database allowing multiple configurations
such as audio profiles to be uploaded to
individual codecs or a single configuration
to be uploaded to many codecs.
Hierarchical View*
All audio codecs can be managed in groups based on a hierarchical structure.
This allows for codecs to be grouped by city, by state and by country and
monitored from a single location.
Alarms at a codec level are propagated up through all of the levels allowing
early detection of faults developing throughout a network. A user can easily
drill-down through the hierarchical map to locate the source of the alarm and
take remedial action where appropriate.
The hierarchical map views can be easily constructed with user-generated
bitmaps, associating base IP addresses with each codec group.
Journalist Panel #
Connection Wizard
The NMS provides a userfriendly tool enabling nontechnical staff to quickly and
easily establish & clear IP &
ISDN connections.
Audio Profile Editor#
Tools for editing the speed-dials and audio
profiles in the database are also provided.
This tool guides the user through a logical sequence of entering a cost code
(optional), selecting a destination (either manually, from speed-dials or from
an SQL database lookup), choosing from a selection of pre-set audio profiles
(covering audio algorithm and bit rate), establishing scheduling and timing of
each connection.
Dual Destination calls (ISDN) are also managed with this feature.
*Cost option - # Contact us for availability of these features
21
Digiplexer
HQSound Audio Processor
The Digiplexer is the first HQSound® audio processor. Using
the latest multi-band DSP technology, it provides up
to 2.8 gigaflops of processing power for FM and HD format
FEATURES
What is the secret of great Digiplexer sound?
HQSound! According to our test users, it is the most incredible dynamic
processing engine ever made…
Why does HQSound give such good results?
The results of our extensive tests prove that HQSound, the new algorithm
engine specifically developed for our range of products, offers on average 20
times more power than those commonly used in competitors’ products. So,
with 4 bands and no other peripherals in front or behind it, HQSound® can
rival and even surpass processing chains made up of several processors in
series.
• Higher sampling rates 192 kHz / 1.5 MHz
• Extended FM bandwidth to 17 kHz
• Low delay: <6ms on any output
•
•
•
•
•
•
•
•
•
•
•
•
•
HQSound Processing
Gated Automatic Gain Control
4 Parametric Equalizers
Dynamic Bass and Trebble Enhancers
Stereo Enhancer and Limiter
Multiband compression
Multiband Gating and Expander
Sound Impact System
Multiband Limiters
1.5MHz FM limiter
Virtual MPX Limiter
Simultaneous HD Limiter
MPX Power Limiter (ITU412-BS)
1st HQSound PROCESSOR
• More processing power built-in
• Sampling rates never achieved before
• Factory and user defined presets, and easy Fine-Tuning
• Low delay ~5ms
MORE RELIABLE
• Proven Hardware reliability
• Auto-switch, Audio Backup, Hardware Bypass, Flash
memory, 2s reboot…
EVOLUTION CAPABILITIES
• Change your processor by simple licence activation
• Add software modules
Three HQSound versions, single compact hardware:
SOFTWARE
UPGRADE
Additional Features:
HQSound® 1-band version:
• Incorporates the AGC plus a Clipper
• To replace an old FMX410/480 (Original
Digiplexer)
HQSound® 2-band version:
• As a main audio processor for soft and
medium formats (Classical, Voice, Jazz…)
• As a secondary processor to finalize the
audio at each transmitter site
SOFTWARE
•
Stereo Encoder: Using high speed DSPs, the Digiplexer
includes a highly over-sampled stereo encoder that offers
amazing stereo performance and perfect synchronization.
•
RDS Encoder: The Digiplexer includes a RDS encoder
fully compliant with the RDS and RBDS standards, the
UECP communication protocol and the new RT+ (Radiotext Plus) standard. Optional Full RDS encoding enables
RDS features equivalent to Audemat’s industry-leading
FMB80 encoder.
•
Audio Backup: The Digiplexer offers an optional 80Gb
hard drive for audio backup with configurable crossfades
during input auto-switches. Basic hard-drive back-up
and advanced audio backup with playlist management
and SHOUTcast/ICEcast server are both available on this
unit.
•
I/O Remote control and ScriptEasy V2 Software
•
Digital audio Input/Output
UPGRADE
HQSound® 4-band version:
• The sound that competes with the best
processors on the market
• For stations who need loudness.
• As a main audio processor for all formats
22
Audio Processor Silver 4B-mini FM
The most cost effective audio processor
The Audio Processor Silver 4B-mini FM is a powerful 4 band
digital audio processor that can be used to process audio for
FM. Using the latest multi-band DSP technology, the Audio
Processor Silver offers cost-effective, versatile and powerful
tools that enable broadcasters to create their sonic signature
and to attract and keep a loyal audience.
FEATURES
•
•
•
•
•
Multi-band audio processing
Analog and AES/EBU audio Inputs
Ethernet and RS232 ports
Preset trigger port and scheduler
Security access codes
BENEFITS
•
•
•
•
•
•
Compact 1U rack Audio Processor & Stereo
Encoder
Complete multiband processing architecture
Secured unit with 3 s reboot after power fail
Easy to install and configure using PC software
or front panel display
Remote TCP/IP access
Many User and Factory Presets
Software interface
•
Input selection and conditioning: The Audio Processor
Silver 4B-mini FM offers the user input selection between
analog and digital. The audio is then routed through optional
pre-emphasis filters.
•
Distortion Controlled Clipper: The Audio Processor Silver
4B-mini FM’s main clipper uses sophisticated algorithms
to produce tightly peak controlled output and maximum
distortion control
•
Bass Enhancement: The Audio Processor Silver 4B-mini
FM offers bass enhancement via a peaking filter that can be
set to provide up to 6 dB of gain on one of four frequencies
with a choice of 4 Q’s.
•
Stereo Encoder: The stereo encoder is highly oversampled and offers superb stereo performance. A
composite clipping function is provided for those who
wish to use it.
•
Multi-band AGC: The Audio Processor Silver 4B-mini FM
processes each band with RMS based levelers. Each band,
gain control, and processing function can be configured in
different manners to provide different effects. Adjustable
timing constants, drive and silence gating afford the user
full control of this important stage of the processor.
•
Multi-Band Limiters: Each band has its own dynamic
peak limiter. Multiple time constant based detectors are
adjustable as well as input drive levels.
•
Mixer: The Audio Processor Silver 4B-mini FM includes a
4 band equalizer. It allows you to subtly color your sound
after the Multi Band Limiter stage, and before the audio
goes into the peak processing path.
•
Bass Clipper* :The Audio Processor Silver 4B-mini FM peak
limits (clips) and linear phase filters the low frequencies
before these are fed to the final clipper stages.
Software Interface
PC software enables easy local or remote
configuration of the Audio Processor Silver.
Connection is made through RS232 (serial
or USB port) or via TCP/IP. All parameters
can be configured with the user friendly
interface and the results are displayed on
the LED meters, on the front panel and on
the configuration screen of the software.
23
RDS Encoders FMB50 & FMB80
FMB50/FMB80
FEATURES
•
•
•
•
•
•
•
•
•
•
•
•
Fully Compliant with RDS/RBDS standards
Support for RT+
ODA support for TMC & Emergency Alert*
Embedded Scheduler
Command Triggering on RDS Signal or internal/
external events
Firmware upgrade available from FMB50 to FMB80
Embedded Webserver
Multi communication links to connect with
automation software
Support for multiple protocols : ASCII , UECP,
ASCII+UECP
Remote monitoring of external devices via status
inputs & control outputs
RDS Viewer allows Remote Monitoring of
broadcasted data
Full customization of alarms and email alerts
BENEFITS
•
•
•
•
•
•
Send timely & relevant information to your listener
base
Offer interactive radio to engage your listeners &
improve loyalty
Change Scrolling PS & RT messages based on time
& date
Access and control the encoder securely from
anywhere via the web browser
Simple configuration with standard UECP protocol
or Audemat ASCII commands
See what your listeners can see with the FMB RDS
Viewer
APPLICATIONS:
RDS/RBDS can be used:
•
•
•
•
•
•
to identify the Radio Station
to automatically retune between
transmitters
of
the
same
program
to display song titles and artist
information on receivers
for interactive radio (RT+ ODA
application)
for traffic Announcements & Traffic Message Channel
(TMC)*
as an emergency alert system*
*FMB80 only
Audemat are widely considered as leaders in the field of RDS/
RBDS with over 12,000 units installed worldwide. The FMB50
and the FMB80 are the key players within the RDS encoding
range thanks to their unrivalled reputation for reliability,
quality and functionality.
Both units have been designed for professional broadcast
use and are fully compliant with the
RDS/RBDS standards. The FMB50
has been designed specifically for
broadcasters who wish to exploit the
interactive functionality offered
by RT+ / Song tagging while the
top of the range FMB80 offers
not only RT+ but also support
for Traffic Message Channel,
Emergency Alert applications and all
other ODAs.
Why is Audemat a World Leader in RDS Encoding?
Experience
Audemat has over 20 years experience in the field having been
an active participant in the RDS forum since its inception.
We have worked together with many major broadcasters
throughout the world to install extensive RDS networks and
have an installed base of over 12,000 units.
Versatility
We have a complete range of RDS encoders which means we
have a solution to meet your needs and requirements both at
the studio and at the transmission site. Whatever your brand
of automation software, and no matter what data path you use,
one of our RDS encoders will suit your needs and will enable
you to achieve your goals.
High Quality and Spectral Purity
All of our RDS encoders use a superbly designed composite/
MPX board, which allows us to keep high frequency signal paths
very short, and provides inherent shielding. The result is onair performance that is measurably better than that offered by
other types of product architecture. All our encoders construct
their data digitally; with no analog pass band filters, there is no
possibility of drifting or degradation of the audio signal.
Reliability
Audemat encoders are designed for continuous on-air
operation and are known for their reliability. All of our units
feature solid state memory, have no moving parts and retain
their configuration through power outages. Every parameter
is controlled through software: there are no potentiometers or
trimmer capacitors that may age or need realignment.
24
FMB50 and FMB80 RDS Encoders
Key Features of the FMB50/FMB80 include:
RT+ : Catapult your FM radio station into the digital
age!
FMB50 / FMB80 Network Services
Today’s listeners can access content via multiple platforms
such as the internet, social networks, mobile apps and digital
broadcasting.
•
•
•
With RT+ on your Audemat RDS encoder, you can now make
exciting interactive content from these new platforms available
on your analog FM broadcast. Podcasts, Streaming, Radio on
Demand, SMS, MMS and EPG are just some examples of the
rich new functionality that can be offered to better engage with
your listeners and encourage greater loyalty.
•
•
•
•
•
Listeners with suitably equipped receivers will not only know
the name of the song they’re listening to
but also be able to access related web data,
a music store or any internet based music
search service. They can use an Electronic
Program Guide (EPG) to schedule program
recording, directly contact the station by
phone, email or SMS and gain easy access
to additional information on commercials.
The FMB50/FMB80 simply allows radio
stations to offer the interactivity today’s
listeners have come to expect.
•
•
HTTP with embedded web server for operator control
SMTP e-mail notification on selected events
F T P (server and client) for firmware, configuration,
data and log file.
Telnet command line interface
UECP encapsulation in UDP
SNMP V1, MIB II, MIB IP2 system
Optional Specific Service Operator MIBs
Operating system: Audemat IP2 Communication protocols
ASCII: For standard computer terminal
UECP: Universal encoder protocol
TCP/IP
You can control your Audemat RDS encoder through TCP/IP.
Simply plug into a local Ethernet network, WAN, or even the
internet for secure operation from anywhere in the world!.
Its built-in, password-protected server is compatible with
FTP, Telnet, SNMP and HTTP and with UECP standard RDS
protocols.
The FMB50 / FMB80 includes three independent high-speed serial
ports for direct wired local control and data communications.
SCROLLING PS
Scrolling PS enables you to scroll dynamic messages (song
titles, artist information and much more) and mix these
messages with the static PS name or call letters.
FMB RDS Viewer
The FMB RDS Viewer, a software
tool included with the FMB50/
FMB80,
simulates
a
RDS
compatible receiver and enables
you to check the encoder has
been properly set.
LABELLING
Song titles and artist information coming from the automation
software can now be automatically framed with text stored
in the encoder. The text can be customized and configured
through embedded web pages in the FMB50 and FMB80. Note
that the units can now manage a large number of commands
such as the music genre, the type of program, Internet URLs,
even contact information for the radio station.
ODA (Open Data applications)*:
There are many applications that can use the RDS/RBDS
technology, bringing more service to your listeners and more
revenue to the Radio station/Network. These applications cannot
be deployed with just any RDS/RBDS encoder as they require
ODA. The FMB80 RDS encoder offers this capability
as standard.
COMMAND INTERPRETER
The FMB50/FMB80 can be configured to interpret whatever
labels and commands are coming from the automation
software. This is useful for older systems that do not have
sufficient flexibility or systems in other languages.
TMC: Traffic Message Channel*: TMC delivers up-to-
date traffic information and voice messages direct to a user’s
satellite navigation system.
SCHEDULER
The scheduler enables you to execute any command, including
changing the scrolling PS and RT messages based on time and
date.
Emergency Alert System*: The FMB80 is compatible with
services that relay data to emergency providers.
25
*FMB80 only
Group supported
RDS Encoder Silver
Basic
B8
0
FM
Full
0A, 2A and FFG
From 0A to 15A except
14A & 14B
From 0A to 15A
Fixed
Configurable
Configurable
8 PSN
9 PSN
PS
PS1/PS2
4 DSN
6 DSN
10 DSN
2 DSN
PI
PI1/PI2
4 DSN
6 DSN
10 DSN
2 DSN
Group sequence
EON
RDS Features
In addition to the FMB80 and
FMB50, Audemat offers several
other RDS encoders to suit the
requirements and budgets of a
wide variety of customers:
B5
0
RDS Encoder
Comparison Chart
FM
RD
S
Si Enc
lv o
e r de
r
Di
gi
pl
ex
er
Which RDS Encoder is for you?
0A to 15A
0A, 2A, 4A,
FFG
Configurable
PTY
RDS/RBDS
RDS/RBDS
RDS/RBDS
RDS/RBDS
AF
25 AF No method B
method A and B
method A and B
method A and B
with software
by command or
contact closure
by command or
contact closure
by command or contact
closure
TP / TA
PTYN
with software
CT
* Subject to applicable regulations
RT+ only
Scrolling PS
Dynamic PS
Sequencing speed
Scrolling by character
Adjustable in sec
Adjustable in sec
Adjustable in sec
Adjustable in sec
from 1 up to 8
from 1 up to 8
from 1 up to 8
from 1 up to 8
1 message
10 messages
10 messages
10 messages
Group Sequence
Group Sequence
Group Sequence
Scrolling by word,
8 character block,
Automatic centering,
Truncate long words
Repetition, Labeling,
Delay before display
Radiotext
RDS Encoder Silver is the most
affordable encoder in our product
line. It can be connected to
your automation software and
supports Radio Text messages,
Alternative Frequencies, Traffic
Announcements and scrolling
PS* by word or characters. The
compact and reasonably-priced
RDS Encoder Silver comes with
a
user-friendly
configuration
wizard and a simulation mode.
ODA: TMC, EWS, EPP
PAGING, RT+...
Radiotext
RT Rate Adjustment
Formatted Radiotext
RT+
Scheduler
ScriptEasy
Configuration Software
Digiplexer
Communication
Audemat’s Digiplexer includes RDS
encoding functionality in addition
to sound processing, audio backup,
remote control, FM transmitter
etc.
Embedded web server
Embedded web server
with RS232
with RS232 or TCP/IP
with RS232 or TCP/IP
with RS232 or TCP/IP
Simulation
Real time
Real time
Real time
for messages
for configuration and
messages
for configuration and
messages
for messages
Partially compliant
Compliant
Fully compliant
8 inputs + 4 relays
8 inputs + 4 relays
16 in / 8 out Option
3 RS232 (two R S - 2 3
2 75 to 115,200 baud,
one 9600 baud) + 1
RS485 + 1 TPC/IP
3 RS232 (two R S - 2 3
2 75 to 115,200 baud,
one 9600 baud) + 1
RS485 + 1 TPC/IP
3 RS232 + 2 ethernet + 1 USB
with command
with command
with command
Password protection
History Log
Connection with automation software
Command translator
Remote Display
ASCII protocol
TCP/IP port - HTTP FTP - TELNET - SNMPSMTP- UDP- TCP
UECP standard
See pages 22 for more information
on this product.
RDS Hardware
Inputs/Outputs
Communication port
1 RS232 (RS232-USB
cable) - 1200 to 9600
baud
Synchro. Monitoring
Side Chain Mode, Loop
through mode, Bypass
feature
Integrated RDS Decoder
26
Fully
compliant
Compliant
WorldCast Systems
Head Office:
20, av Neil Armstrong
Parc d’Activités J.F. Kennedy
33700 Bordeaux-Mérignac
France
T: +33 557 928 928
E: contact@worldcastsystems.com
UK Office:
Whiterock Business Park
729 Springfield Road
Belfast, BT12 7FP
UK
T: +44 (0) 28 90 677 200
E: info@aptcodecs.com
WorldCast Systems Inc
19595 NE 10th Avenue Suite A
Miami, FL 33179
USA
T: +1 305 249 3110
E: ussales@worldcastsystems.com
www.WorldCastSystems.com
© Copyright WorldCast Systems 2012. All rights reserved.
Enhanced apt-X® is a registered trademark of APT Licensing Ltd.
Was this manual useful for you? yes no
Thank you for your participation!

* Your assessment is very important for improving the work of artificial intelligence, which forms the content of this project

Download PDF

advertising