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: firstname.lastname@example.org UK Office: Whiterock Business Park 729 Springfield Road Belfast, BT12 7FP UK T: +44 (0) 28 90 677 200 E: email@example.com WorldCast Systems Inc 19595 NE 10th Avenue Suite A Miami, FL 33179 USA T: +1 305 249 3110 E: firstname.lastname@example.org www.WorldCastSystems.com © Copyright WorldCast Systems 2012. All rights reserved. Enhanced apt-X® is a registered trademark of APT Licensing Ltd.
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