AVN-HA1 Analogue Headphone Amp for AVN-PA8/D & AVN-PM8/D Portals
AVN-HA1 Analogue Headphone Amp for AVN-PA8/D & AVN-PM8/D Portals
The AVN-PA8/D and the AVN-PM8/D Portals can be combined with multiple AVN-HA1 headphone amplifiers to provide a headphone distribution system – the portal output connections can supply analogue power to satellite headphone amplifiers and the unit can be mounted using the CM-MNT1 desk mount panel.
The AVN-PA8/D and AVN-PM8/D can be combined with multiple Sonifex AVN-HA1 headphone amplifiers to provide 8 separate headphone signals where each headphone amplifier can be sent a separate feed, mixed from any physical or stream inputs.
- Front panel 6.35mm (1/4”) headphone socket and volume control knob.
- Front panel Mute/GPO push button.
- Analogue audio input on RJ45 (the connector provides power to the unit and a GPO back to the portal).
- Loop through audio output on RJ45 (power and GPO signal are not connected).
- Locking DC power connector (if a portal is not being used to supply the unit with power).
Note: The AVN-HA1 is an analogue input product taking an analogue audio feed from the AVN portals. It can be used independently of the portals by using the separate DC input for power and a separate analogue input.
Additional information
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LTN LEAF Flypack
A complete IP solution for any live event, Flypack features plug-and-play setup, offers encryption capabilities, and encodes and decodes at up to 20 mbps with SD and HD support.
By connecting Flypack to the LTN Network, you can manage and control video from origin to destination, monitoring every signal with LTN Portal and always-on NOC support.
Power the future of video with LTN Leaf Flypack.
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LTN LEAF
The Leaf appliance requires minimal on-site setup, and features a compact, integrated design that harnesses the flexibility of multiple encode and decode configurations. Leaf also enables seamless access to the LTN Network, with real-time monitoring and always-on support.
The Leaf encode mode supports interactive interviews and live shots with lower end-to-end latency than satellite. Leaf also features broadcast-quality encoding for fast-motion sports and complex tape playout content at bitrates up to 20 Mbps, with high-quality multi-codec decoding of up to four simultaneous feeds.
Create the future of video with LTN Leaf.
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LTN Network
With proprietary protocols for dynamic multicarrier routing and rapid error recovery, the LTN Network is engineered to outsmart congestion and preempt failure, ensuring consistent live video delivery.
Backed by the LTN Network, major media producers and distributors expand their reach, increase their revenue, and deliver transformative video experiences.
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Caton Prime
Caton Prime is designed with unmatched reliability for round-the-clock mission-critical live broadcast, making it ideal for telemedicine, remote production, enterprise and critical infrastructure sector's applications.
Key features:
- High quality up to 4K 60p 4:2:2
- Switchable between encoder and decoder function
- Reliable & Secure IP transmission with CTP
- Easy management and monitoring
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Nimbra 400
With its swiss-army-knife approach, the Nimbra 400 family is the very definition of versatility. Engineered for the harsh realities of the internet, the Nimbra 400 family supports a wide range of video and transport technologies. The combination of native support for both AVC and HEVC encoding/decoding, compressed video over IP and ASI and an open and flexible approach to IP networking in a super-efficient 1 RU form factor makes the Nimbra 400 family a perfect companion for space constrained remote sites and flight kits. With native support for RIST, SRT, Zixi and our own Edge Connect software means you can get your content anywhere. The integrated encryption and firewall paired with transport of any remote management traffic and comms also minimizes the need for third party equipment on site.
Paired with its standalone software counterpart, the Virtual Nimbra, means there are solutions ranging from highly redundant appliances for flight-kit and racks, to flexible software for datacenters and public cloud providers.
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Nimbra 300
Combining native video, audio and high quality ethernet transport with switching, the Nimbra 300 provides a high-quality multiservice solution for demanding media applications. Multi-service capabilities allow you to transport live and file-based content across the same network, and with native support for DTT (ATSC 3.0, DVB-T2, ISDB-T) transport, Broadcast Radio distribution and video contribution the Nimbra 300 series gives you all the flexibility you need.
With support for service admission control and guaranteed bandwidth, performance and characteristics different services can safely be mixed no matter the network load. With hitless 1+1 protection of media and data services you can be sure that the expected service quality will always be delivered.
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Nimbra 1060
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Nimbra Edge
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HOME
HOME is a management platform for IP-based media infrastructures. It is designed to connect, manage and secure all aspects and instances of live production environments. HOME provides the tools and centralized services for swift and effective interaction of engineers with their tools.
HOME is cloud-native by design and ready to run anywhere, irrespective of the system’s size. With HOME, the cloud starts on your campus, private and locally. It turns an array of devices, setups, sites, hubs and data centers into a powerful, agile network — quickly and in a perfectly secure way.
Inside HOME, discovery of devices is automatic, while registering and admitting them to the network is only a button press away.
With the adoption of IP well underway, the focus of operators has shifted from whether to adopt IP to how to use its potential with minimal effort and maximum effect. This is where HOME shines: it addresses all pressing issues real-world operators face today and tomorrow. In one place and via a single, platform-agnostic, intuitive user interface.
Lawo’s HOME platform is based on open standards such as ST2110, NMOS, IEEE802.1x and RADIUS and has been designed and built from the ground up using LUX—the Lawo Unified Experience, a framework for conceiving, designing, and building solutions that put you first and defines the standard for user experience and design across the Lawo portfolio.
Discovery and Registration: HOME solves IP complexity with automatic plug & play discovery of IP audio and video devices, which are registered with their name, location, status and type. This applies not only to Lawo products but to third-party solutions as well via NMOS. Discovered devices are managed in a central inventory list, ready for access and configuration.
Device Management: In today’s hectic live broadcast environments, operators rely on speedy, unified device configuration routines, especially when setting generic device parameters or configuring senders and receivers. The ability to save and recall configurations is key to speed up tasks. HOME provides a centralized “mission control” for these processes, providing fast and unified access to device parameters for easy tweaking, irrespective of the end point being controlled.
Operability: With its simple, user-friendly UI, HOME allows users to organize and access processing services. With all required facilities accessible in one place, operators can set up and change stream configurations, and route them across an infrastructure without the need for a separate controller. For large infrastructures HOME works seamlessly with a broadcast controller in the same set-up and helps to speed up configuration and operation. HOME is based on LUX, a UI language common to all Lawo devices and many of their functionalities. Through HOME’s user interface, operators can access and edit device parameters quickly utilizing integral mechanisms that help get the job done efficiently. With HOME, operators quickly get right to what they’re looking for, without distractions and complications, to focus 100% on the task at hand.
Security: The content created by a production crew and transported over a network is any operation’s most valuable asset and deserves strong protection. While a robust security system needs to cover all aspects of media infrastructure and content creation, the key lies in its simplicity. HOME provides a variety of security strategies, first of which is quarantining unknown devices when they come online. Only after being deliberately approved, via an intuitive IEEE802.1X-based routine, can they begin exchanging signals with the HOME network. Secondly, HOME uses an authentication strategy based on a centralized user management system, with dedicated user roles and groups. The LDAP based service allows users to authenticate either locally – within HOME – or via their own corporate IT infrastructure, e.g. Microsoft® Active Directory. Finally comes the arbitration of devices and individual streams based on pinpointed rights management. HOME’s architecture is prepared to manage services such as transport layer security, network segmentation and other IT security mechanisms such as RADIUS.
Scalable Architecture: Home is cloud-native by design, which means that its architecture is built to run detached from hardware constraints. This does not automatically mean that services must be outsourced to an external service provider whose meter is running 24/7; with HOME, the cloud starts on your campus, private and locally, on COTS hardware. The HOME platform is designed as functional blocks that provide microservices, which are self-contained and supply functionality to operators or other services.
HOME can be expanded with additional services at any time to increase its functionality — the platform scales on demand. Should there be a need for a larger RDS, because the installation grows, additional instances of the required resources can be added anytime. One of the core principles of HOME is its focus on the utilization of open standards wherever possible, for broadest compatibility and future-proof integration. With HOME, flexibility and resource utilization in IP media infrastructure maximizes.
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Caton Netscope
Unlike general NPMD tools, Caton Netscope is meticulously engineered for Live IP production, providing unparalleled network visibility and real-time detection for a superior, error-free streaming experience.
Key Benefits:
1.) Unrivaled Network Insight
Empowering Proactive Network Management
2.) Real-time Event Analysis
Precise Pinpointing of Network Failures
3.) Superior Compatibility
Integrating Seamlessly with Any Protocol
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INTRAPLEX® IPCONNECT
IPConnect solves the problem of reliable transport across complex networks that traditionally suffer data loss through dropped packets. IPConnect provides an added layer of reliability for many types of IP traffic such as HD Radio, DAB/DAB+, or DRM by employing a combination of packet protection schemes with network/time diversity and packet-level forward error correction. Additionally, IPConnect can bridge local area network (LAN) segments across wide-area networks with seamless tunneling to enhance reliability of program and signal transport across large geographical regions with multiple receiving and transmitting sites.
With its three network interfaces, IPConnect’s data gateway provides extra protection through packet encapsulation, which encloses external IP data packets in a GatesAir protocol wrapper as it moves across IP networks. IPConnect provides this protection for studio-generated data, as well as IP data from external sources. While the former might include RDS data or SNMP control signals to trigger a command at the transmitter site, the latter may incorporate program or control data coming from a network operations center, satellite feed or advertising service.
As the next-generation member in GatesAir Intraplex family of IP codecs, the award-winning IPConnect leverages the strengths of previous IP Link codecs—including GatesAir’s patented Dynamic Stream Splicing (DSS) technology to mitigate IP packet loss, eliminate off-air time, and optimize stream reliability.
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Caton Relay Server
Deploy and manage complex video with Caton Relay Server
Caton Relay Server is a TS streams management platform that manages your input streams and relays them seamlessly to multiple locations and devices, while monitoring the status of the stream transport.
Caton Transport Protocol (CTP) multipathing is an enhanced feature for the Caton Relay Server. The CRS can now set up 8 x 8 network connections with up to 64 paths for a single transmission. The intelligence of CTP multipathing with traffic shaping capabilities improves the resilience of the transmission over the public internet, even in the most challenging network conditions. In addition, the CRS customises and prioritises the network for different streams based on business needs like priority of service and costing. The newly introduced auto-failover feature within the CRS automatically switches from the primary CRS to the backup CRS in the event of a breakdown. Thus eliminating the need for physical intervention by the network engineer, resulting in a reduced downtime and support resources.
Supporting major industry standards protocols (including CTP and SRT) and with native support for CatonNet Video Platform (CVP), the Caton Relay Server is designed to handle complicated video routing, real-time status, built-in TS analyzers, and stream-level monitoring alert that provides constant information about bandwidth, jitter, up-time and connection status for network operators.
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INTRAPLEX® HD LINK™
Our challenge: to design a studio-to-transmitter link (STL) for your most demanding 950 MHz applications — one as reliable and robust as Intraplex® T1 and IP audio links. Digital STL manages much more than audio, so installation and configuration need to be straightforward, not a science project. Data should not be optional, requiring additional boxes and complexity. Your STL should be ready, out of the box, for AM, FM and HD Radio™, as well as future multimedia applications.
Earlier digital STLs were not designed for IP data transport. Adding IP data to them required optional modules and external add-ons, and many engineers working on HD Radio installations reported spending much time and money trying to eliminate glitches.
Our solution: HD Link™ is designed to manage all HD Radio™ transport scenarios, regardless of where you place your importer and exporter. Its two prioritized Ethernet paths give preference to HD Radio™ data over control and other LAN/WAN data. It supports both UDP and TCP, and even handles the switching of TCP return packets over asymmetric IP paths with plug-and-play simplicity.
HD Link offers RF power to spare, an integrated IP gateway with sophisticated data handling capabilities, and multiple channels of audio. The intuitive front panel and remote interfaces tap into the most complete feature set of any microwave STL, yet take less time to configure. It can even operate on both RF and IP simultaneously, and allows automatic backup of all services from one to the other.
Intraplex is broadcasting’s first choice for rock-solid, full-time operation of T1 STLs. With HD Link, you can now count on the same dependable performance, superior support and long-term value for your microwave links.
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INTRAPLEX® IP LINK
By incorporating three IP Interfaces that can be used for streaming and management, the IP Link systems can provide a level of reliability not seen in comparably-priced codecs.
As the latest additions to the Intraplex line of data transport products, the IP Link family of audio codecs bring legendary Intraplex reliability to the IP codec market.
IP Link 100/100p: Single bidirectional stereo audio channel IP Link 200: Two bidirectional stereo audio channels IP Link 200A: Two bidirectional stereo audio channels with one channel being AES67 Standard: Linear, AAC-LC, Opus and G.722 audio coding Optional: AAC-HE, AAC-HEv2, AACELD, MPEG2, MPEG3 and Enhanced aptX audio coding Optional: Automatic audio loudness leveling and metering compliant with EBU R-128 and ITU-R Optional: IPConnect capability to reliably transport external IP packets Other transport modes: Transparent AES up to 192 kbps to support composite FM multiplex signal over AES Protocol Encapsulation: RTP, SHOUTcast/Icecast, MPEG-TS Three independent IP interfaces for redundant network operation Optional redundant power supply: 12VDC or 48VDC Built-in silence sensor with optional stream switch over Automatic backup to audio playout from USB drive or external audio source Multicoding can encode the same audio source in multiple formats for STL, backup, and web streaming Optional Dynamic Stream Splicing providing “hitless” operation and T1/ E1 circuit like performance on less predictable IP networks Prioritized stream sources at the output with automatic switch over and switch back between primary and secondary streams and backup sources (streams, USB, external audio source) Programmable RTP level Forward Error Correction (FEC) scheme Programmable time diversity and interleaving of streams to combat burst packet losses Integrated with Intraplex IP Link Scheduler for automated scheduled program switching Integrated with Intraplex LiveLook (network analytics and monitoring software) N+1 redundancy with integrated control of external switching equipment IP Link 200/200A/100p: SynchroCast™ option provides dynamically managed precision delay for Single Frequency Network (SFN) broadcasting and simulcasting Support for IP multicast and multi-unicast Web browser user interface and SNMP network management Eight multipurpose contact closure inputs and outputs provide: Transport of logic signals with time-alignment to audio Stream control Alarm notification
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DARK1616S
In total there are 16 channels of audio sent from the Dark1616S into the network. The Dark1616S has 8 of AES3 inputs and 16 of analogue inputs. Using the GlenController App it is possible to set which inputs (AES3 or Analogue) are being sent to the Dante/ AES67 network, an auto mode is available that sends AES3 (when a valid signal is detected) in preference to the analogue input.
Simultaneously there are 16 channels of audio being received from the network by the Dark1616S and these incoming circuits are provided as outputs from the Dark1616S in both AES3 and analogue.
The AES3 inputs have sample rate converters on them and can accept input frequencies up to 192kHz, the incoming AES3 circuit is always sample rate converted to match the Dante network frequency.
The AES3 outputs have sample rate converters on them and can be locked to the sample frequency of the Word clock input, the first AES3 input or to the Dante network.
A Word clock output is also provided, this is clocked at the same sample rate as the Dante network and can be used for locking external equipment to the network's sample rate.
The unit can be remote controlled using our GlenController Windows 10 App. The App allows such things as the dBFs levels to be set, clock masters and also provides input/ output metering. See the App tab for further details.
For ease of cabling audio I/O is presented on D25 sockets to the AES59 standard (Tascam wiring convention) for which there are a number of reasonably priced break out cables available from multiple suppliers.
Being designed for resilient broadcast applications, the Dark1616S features both redundant power supplies and redundant Dante network links with link status GPOs (general purpose outputs (solid state relays)). Both primary and secondary network links are provided with both magnetic (copper RJ45) and fibre (SFP) interface connections. The Dante system itself provides a completely transparent redundant link system which means that if the Dark1616S lost its primary link circuit the secondary link would automatically take over with no loss of audio.
The primary and secondary network interfaces are routed internally via a network switch. It is possible to set this switch to work as a traditional network switch instead of the default redundant mode. This means that there would be just one link to the Dante network and the other connections of the switch could have other Dante or network devices connected to them. As with all Dante devices, once set up, Dark1616S units can be directly connected with each other with no external network hardware.
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DARK1616D
Dante network audio is a common protocol for distributing high quality linear audio over standard IP networks and it is widely used by many audio equipment manufacturers. The Glensound Dark1616D Dante audio interface will be compatible with any other manufacturers Dante audio interface.
Being designed for live on-air broadcast applications the Glensound Dark1616D has been designed with multiple redundancy capabilities. It has 2 mains power sources and it also has fully redundant network connections for both Copper & Fibre circuits.
The Dark1616D provides 8 balanced AES3 inputs and 8 balanced AES3 outputs to the Dante network on rear panel D25 connectors wired to AES59 (also known as the Tascam standard).
As per our other Dante equipment 0dBu = -18dBFs
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GS-FW012 ip
There are still four inputs for the top panel loudspeaker or headphone monitoring, each with their own level controls and each being derived from the Dante network. There are still four talkback outputs each being routed to the Dante network.
Both the inputs and outputs are presented via a single rear panel Neutricon RJ45/CAT5 connection that is Dante/AES67 compliant.
The four talkback outputs are identical. Each has a 3 position lever key talk switch that is either off, locked on, or in a sprung push to talk mode. The output of each circuit can be off unless the talkback button is pressed, looping the the associated Dante input input, or outputting the local cue input.
The local cue input is presented on XLR and has it's own level pot for the speaker or headphone monitoring.
A good quality microphone amplifier with adjustable gain followed by a compressor limiter is built into the unit making its talkback outputs to the Dante network crystal clear. Conversion of the audio signals to/ from the Dante network are by low noise, high bandwidth 48K 24bit converters providing superb performance.
Audio I/O can be routed via the digital Dante router. The free Dante Controller software configures all routes and can be downloaded by clicking here.
There is an internal switch mode power supply, or the GS-FW012 ip can be powered via PoE via the CAT5 connection. If both supplies are present then they will act as redundant supplies.
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DARK16O
In total there are 16 channels of audio being received from the AoIP network and converted via low noise DACs (Digital to Analogue Converters) to balanced analogue outputs.
For ease of cabling audio outputs are presented as balanced circuits on Neutrik XLRs.
Being designed for resilient broadcast applications the DARK16O features both redundant power supplies and redundant Dante network links with link status GPOs (general purpose outputs (solid state relays)). Both primary and secondary network links are provided with both magnetic (copper RJ45) and fibre (SFP) interface connections. The Dante system itself provides a completely transparent redundant link system which means that if the DARK16O lost its primary link circuit, the secondary link would automatically take over with no loss of audio.
The primary and secondary network interfaces are routed internally via a network switch. It is possible to set this switch to work as a traditional network switch instead of the default redundant mode. This means that there would be just one link to the Dante/ AES67 network and the other connections of the switch could have other Dante or network devices connected to them. As with all Dante devices, once set up, DARK16O units can be directly connected with each other with no external network hardware.
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DARK16I
In total there are 16 audio input channels, all being simultaneously converted to the AoIP network via high quality low noise analogue to digital converters.
For ease of cabling audio inputs are balanced circuits on Neutrik XLRs.
Being designed for resilient broadcast applications, the DARK16I features both redundant power supplies and redundant Dante network links with link status GPOs (general purpose outputs (solid state relays)). Both primary and secondary network links are provided with both magnetic (copper RJ45) and fibre (SFP) interface connections. The Dante system itself provides a completely transparent redundant link system which means that if the DARK16I lost its primary link circuit the secondary link would automatically take over with no loss of audio.
The primary and secondary network interfaces are routed internally via a network switch. It is possible to set this switch to work as a traditional network switch instead of the default redundant mode. This means that there would be just one link to the Dante/ AES67 network and the other connections of the switch could have other Dante or network devices connected to them. As with all Dante devices, once set up, DARK16I units can be directly connected with each other with no external network hardware.
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DARK88 MKII
In total there are 8 channels of audio sent from the DARK88 MKII into the network. The DARK88 MKII has 8 of analogue electronically balanced audio inputs on Neutrik XLRs.
Simultaneously there are 8 channels of audio being received from the network by the DARK88 MKII and these incoming circuits are provided as outputs from the DARK88 MKII in analogue.
Network sample rates of up to 192KHz are accommodated seamlessly within the DARK88 MKII.
Being designed for resilient broadcast applications the DARK88 MKII features both redundant power supplies and redundant Dante network links. Both primary and secondary network links are provided with both magnetic (copper RJ45) and fibre (SFP) interface connections. The Dante system itself provides a completely transparent redundant link system which means that if the DARK88 MKII lost its primary link circuit the secondary link would automatically take over with no loss of audio.
The primary and secondary network interfaces are routed internally via a network switch. It is possible to set this switch to work as a traditional network switch instead of the default redundant mode. This means that there would be just one link to the Dante network and the other connections of the switch could have other Dante or network devices connected to them. As with all Dante devices, once set up, DARK88 MKII units can be directly connected with each other with no external network hardware.
On the front panel, LEDs indicate the status of the 2 power supplies and the 2 network links. GPO status outputs are also provided for external indication of the power supply & network status.
Network connections are placed on the front panel of the DARK88 MKII in order that the network cables (or fibres) match those of a rack mounted professional network switch, making installation and tracing interconnecting cables easy. Fibre connections are via SFP slots, meaning that users can select their own preferred fibre type & connector style by installing their own fibre SFP modules (a selection of modules is available from Glensound if preferred).
The DARK88 MKII features the Brooklyn module from Audinate which is AES67 compliant.
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DARK8ADI
The DARK8ADI (Analogue & Digital Input) is a very powerful Dante®/ AES67 network audio interface, in a robust 1RU 19" subrack It has a 8 of low noise analogue line inputs and 4 of transformer balanced AES3 digital audio inputs. These inputs are transmitted on a fully redundant Dante/ AES67 network interface.
In total 16 audio circuits are sent to the Dante/ AES67 network. Eight of these circuits are always derived from the 4 x AES3 inputs, the other eight are automatically switched between the AES3 and analogue inputs. If the DARK8ADI detects valid AES3 signal on a channel then this will be routed to the network output. If no valid AES3 is detected then the analogue input will be routed to the network output instead of the AES3.
The rugged design, redundant mains powering & redundant network facility of the unit means that it can easily be placed and left unattended wherever audio sources are required.
Eight Analogue Audio InputsThe DARK8ADI has 8 electronically balanced analogue line level audio inputs. Each input is on its own Neutrik 3 pin XLR socket.Quality Analogue To Digital ConvertersTo get the best possible results from your analogue audio inputs, the very best widest range analogue to digital converters (ADCs) currently available are used to make sure the digital audio on your network is as good as it possibly can be.Four AES3 Audio InputsThe DARK8ADI has 4 transformer balanced digital AES3 audio inputs. Each input is on its own Neutrik 3 pin XLR socket. These inputs can accept sample rates up to 192kHz.
Analogue/ AES3 Auto SwitchingEight of the audio outputs are derived from both the 4 x AES3 inputs and the 8 analogue inputs, whereby the AES3 inputs take priority to the analogue circuits. Our input circuitry looks for valid AES3 data streams on the AES3 inputs. If one is detected then the pair of audio outputs (AES3 is two audio channels) associated with that circuit will be routed to the Dante/ AES67 output and if no valid AES3 signal is detected then the analogue input will be routed to that output.
AES3 Network OutputsAs well as the switched outputs, the outputs of the 4 x AES3 input circuits are always routed to 8 audio channels of the AoIP network.
Sample Rate ConvertersThe AES3 audio inputs are fed through sample rate converters so that they match the AoIP networks sample frequency. The AoIP network supports up to 192kHz sampling and the AES3 inputs can be between 32 & 192kHz.
Network InterfaceThere are 4 network interfaces on the DARK8ADI. There are 2 x Neutrik Ethercon (RJ45) connectors and there are also 2 x SFP slots for customers to fit their own preferred fibre interfaces.
Redundant Network InterfaceWhen using the Dante protocol it is possible to set the DARK8ADI to have a fully redundant network interface whereby a completely glitch free automatic redundant audio network link is provided across 2 of the network interfaces.
PowerEach DARK8ADI can be powered from two independent sources to provide a multi-redundant power option.Two wide range switch mode power supplies are fitted as standard to provide redundant mains power supplies.
Alarms & LEDsThe front panel features two LEDs (one indicating OK and the other fault) for both power supplies and both network ports. This data OK/ Fault information is also provided on solid state relay outputs on front panel D connectors.
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DARK1616M
Inputs & OutputsThere are 16 digital inputs and 16 digital outputs, available on 8 x AES connections. There are also 16 analogue inputs and 16 analogue outputs, available in parallel to the digital. The analogue and digital outputs are always available, and the inputs work either on AES input priority or can be selected via the control app.
The AES3 inputs have sample rate converters and can accept input frequencies up to 192kHz. The incoming AES3 circuit is always sample rate converted to match the Dante network frequency. The AES3 outputs are locked to the sample frequency of the Dante network.
ConnectionsFor ease of cabling, audio I/O is presented on D25 sockets to the AES59 standard (Tascam wiring convention) for which there are a number of reasonably priced break out cables available from multiple suppliers.
Remote Mic AmpsThe analogue inputs are switchable between mic, line and 48v phantom power. These are controllable across the network via the Dark Controller Windows 10 app. This allows remote control by an engineer of the input selection, gain adjust and input on/off control. A meter is also provided for each input on the app, to monitor the input level and help with setting the gain.
Network InterfaceBeing designed for resilient broadcast applications the Dark1616M features both redundant power supplies and redundant Dante network links with link status GPOs (general purpose outputs (solid state relays)). Both primary and secondary network links are provided with both magnetic (copper RJ45) and fibre (SFP) interface connections. The Dante system itself provides a completely transparent redundant link system which means that if the Dark1616M lost its primary link circuit the secondary link would automatically take over with no loss of audio.
The primary and secondary network interfaces are routed internally via a network switch. It is possible to set this switch to work as a traditional network switch instead of the default redundant mode. This means that there would be just one link to the Dante network, and the other connections of the switch could have other Dante or network devices connected to them. As with all Dante devices, once set up, Dark1616 units can be directly connected with each other with no external network hardware.
AlarmsOn the front panel 4 bright LEDs indicate the status of the 2 power supplies and the primary and secondary local network links. In parallel to these 4 indicating LEDs there are 4 solid state relay outputs for connecting to external alarm systems for failure notification of a power supply or link fault.
';
DARK1616
In total there are 16 channels of audio sent from the Dark1616 into the network. The Dark1616 has 8 of AES3 inputs and 16 of analogue inputs, but they cannot be used at the same time. The AES3 inputs take priority over the analogue (if an AES3 input is receiving a valid AES3 signal then it will turn off the equivalent analogue input pair and route its output to the network).
Simultaneously there are 16 channels of audio being received from the network by the Dark1616 and these incoming circuits are provided as outputs from the Dark1616 in both AES3 and analogue.
The AES3 inputs have sample rate converters on them and can accept input frequencies up to 192kHz. The incoming AES3 circuit is always sample rate converted to match the Dante network frequency.
The AES3 outputs are locked to the sample frequency of the Dante network.
For ease of cabling audio I/O is presented on D25 sockets to the AES59 standard (Tascam wiring convention) for which there are a number of reasonably priced break out cables available from multiple suppliers.
Being designed for resilient broadcast applications the Dark1616 features both redundant power supplies and redundant Dante network links with link status GPOs (general purpose outputs (solid state relays)). Both primary and secondary network links are provided with both magnetic (copper RJ45) and fibre (SFP) interface connections. The Dante system itself provides a completely transparent redundant link system which means that if the Dark1616 lost its primary link circuit the secondary link would automatically take over with no loss of audio.
The primary and secondary network interfaces are routed internally via a network switch. It is possible to set this switch to work as a traditional network switch instead of the default redundant mode. This means that there would be just one link to the Dante network, and the other connections of the switch could have other Dante or network devices connected to them. As with all Dante devices, once set up, Dark1616 units can be directly connected with each other with no external network hardware.
On the front panel 4 bright LEDs indicate the status of the 2 power supplies and the primary and secondary local network links. In parallel to these 4 indicating LEDs there are 4 solid state relay outputs for connecting to external alarm systems for failure notification of a power supply or link fault.
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Livelink
Livelink is a robust IP delivery solution for transporting live linear and OTT content from point to single-point or multi-point. It allows users to self-manage their live content for delivery to any destination worldwide by utilizing cloud environments available in every region. Livelink is cloud agnostic and compatible with any transport protocol including Zixi, RIST, SRT, HLS and RTMP.
Livelink’s expand on demand infrastructure ensures that teams can quickly scale their requirements within minutes. This flexible approach enables stakeholders managing live broadcasts and events to adapt to changing circumstances. Livelink does not rely on traditional infrastructure which makes it extremely cost-effective. Its intuitive self-service interface enables users to easily transport, manage and extract value from live content. The platform integrates rights management, automatic booking updates and customisable monitoring and alerts so that customers retain control of all aspects of delivery.
Livelink offers broadcasters the ability to deploy and deliver live feeds with minimal on-site presence. The platform offers global reach and is the ideal solution to reduce live event overheads and also supports current social distancing measures which the broadcasting industry has implemented. The self-management interface monitors availability, routing and scheduling for all feeds, as well as tracking cloud environment costs. Cerberus Tech can fulfil hardware requirements, direct to the end user if required. However, in many instances no additional hardware is required for set-up, just an internet connection.
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Network 1
Network 1 is a reliable and cost-effective internet-based platform, for the primary distribution of linear content. It offers content owners 24/7/365, point to multi-point channel distribution to global audiences. Network 1 empowers customers to focus on the content, not the distribution.
From public service broadcasters, to global media giants, the UK's oldest commercial network and premier league football teams, Network 1 customers are increasingly utilising global connectivity to transport linear video content.
Network 1 offers significantly quicker implementation than traditional fibre and satellite distribution methods, with set-up normally measured in hours rather than months. Network 1 is multi-cloud compatible, protocol agnostic and delivers over managed or unmanaged networks. Infinite scalability ensures on-boarding new channels is easy and if no hardware is required, customers can deliver to new affiliates or broadcast platform operators straight away.
Network 1 is a fully managed service, with network management and monitoring capabilities, offering complete reliability.
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EDGE
It can either be used standalone through a web interface or in most cases connects over websockets to our control and monitoring platform called CORE (deployed on prem or in the cloud).
Typical use-cases are secure and reliable Internet transit, a monitoring probe, 2022-7 protection, Input protection and (De)Scrambling.
Functions that set itself apart from a native SRT implementation are:
• SMPTE 2022-7 seamless input with per path statistics
• SRT RTP header passthrough allowing 1+1 paths followed by downstream 2022-7 seamless protection
• BISS-2 scrambling and descrambling with seamless odd/even key transition
• Thumbnailing (based on ffmpeg library)
• ETR290 Priority 1, Priority 2 & Priority 3 (fully written in house)
• UDP and RTP output with packet pacing and launch delay offset
• Threshold based input redundancy with support passive or active backup sources
• CORE (CORE is the name of our control platform) - Centralised control and monitoring capable of controlling and monitoring through firewalls and presenting a northbound REST control and monitoring API
• CORE - High availability horizontally scalable cluster based control and monitoring capable of managing tens of thousands of EDGEs.
• CORE has a REST API with existing integrations with ATOS BNCS and Skyline Dataminer.
• CORE supports local, LDAP, SAML and Azure AD authentication.
• CORE - Grafana native driver for correlated monitoring of all software components
• Floating centralised license model with both OPEX and CAPEX options
• EDGE Debugging tools - top talkers & PCAP
• HLS input and output
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Versal ™ Prime Series Gen 2 adaptive SoCs
Designed for a broad range of applications including 8K video processing, the Versal Prime Series Gen 2 devices offer expanded hardened IP to complement the programable logic and processing system. This new IP includes hardened video encode & decode, DDR5/LPDDR5X memory controllers, 100G Ethernet, PCIe® Gen5, and an integrated Arm Mali™-G78AE GPU, enabling next-generation applications within tight area & power constraints.
From capturing content and enabling virtual production to creating live events, Versal Prime Series Gen 2 enables Broadcast and Pro AV equipment suppliers to deliver ever more compelling experiences thanks to a unique combination of dedicated acceleration, flexible interfaces, and adaptive computing that’s ideal for real-time video & audio processing.
Each video codec unit (VCU) tile offers:
- Hardened bidirectional encoding, decoding and transcoding
- Support for HEVC & AVC up to 4K60, 4:4:4, 12-bit
- Up to two VCU tiles per device; aggregate both tiles for limited 8k30 support
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MDX Core & Aggregate Media over IP Switching
- Fault tolerant switching hardware
- TICO and JPEG compression for bandwidth conservation
- Fast OpenFlow switching
- Non-blocking switch architecture
- Integrated provisioning and network management
- SMPTE 2110 enabled
- NEBS Certified
Available Models:
- MDX32C Core Switch comes equipped with 32 x QSFP28 switch ports. Each port supports 1 x 40/100GbE, 2 x 50GbE, or 4 x 10/25GbE.
- MDX48x6C Aggregation Switch comes equipped with 48 x SFP28 (1G/10G/25G) + 6 QSFP28 (100G), a switching capacity of 1.8 Tbps, plus OpenFlow and Media Links’ Non-blocking Technology.
Applications & Use Cases:
- Carrier Class Media Delivery Networks
- High Performance Studio Interconnects
- Flawless Contribution Video Transport
- Contribution Video over Terrestrial and Satellite Networks
DOWNLOADS:
- DOWNLOAD Datasheet for MDX-32C
- DOWNLOAD Datasheet for MDX-48X6C
- WATCH 100G Eco-System Video
- DOWNLOAD Media Links Solutions Brochure
- DOWNLOAD North America Case Study (Telecom)
- DOWNLOAD Metropolitan Distribution Network Application Note
- DOWNLOAD Live Sports Production Application Note
- DOWNLOAD Centralized IP Switching Application Note
- DOWNLOAD WAN IP Network Application Note
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L-GRID
L-GRID tackles the problem of speed mismatches at the source and avoids an extra investment in large buffer core switches.
L-GRID will configure the network card to only allow bursts of the configured bandwidth. Data flows to the preconfigured destinations are sent out of the network card perfectly shaped to the given bandwidth. As the L-GRID eliminates bursts before they enter the network, no exhaustive buffer space is required from the connected switches. The traffic is controlled/managed at the source. The cause for congestion is remediated before it can happen.
L-GRID enables the user to configure specific bandwidths for specific/configured clients. L-GRID runs as a user-space app on a Linux server and is configured via a config file or a REST interface. A certified network card is required
L-GRID will thereby improve the efficiency of your installed network.
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GRID
GRID encompasses a technology and a methodology to enable guaranteed data flows with effectively-enforced QoS in any IP network: a studio network, a post-production network, a data center network, etc.
GRID’s software transforms the IP network and the data flows to be fully predictable, non-blocking and 100% reliable. As a result, your network will be used much more cost-efficiently. GRID also provides several wizards to make configuration and management of the network easy and intuitive. The software enables a 100% service guarantee with a specific bandwidth, low latency and no packet-loss per service or flow.
GRID is compatible with the hardware of multiple vendors.
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RouteMaster
RouteMaster uses Rascular’s proven router control and emulation technology to form a powerful yet flexible router control system. Built on tried and tested software modules, RouteMaster can be used with a wide range of video and audio routers from all major manufacturers.
It’s equally suitable for new router installations or increasing the capabilities – and lifetime – of existing systems. You can add third-party hardware panels to existing systems, add IP connectivity to older serial routers, or simply increase the number of control ports available.
The integrated web server means operators can control routers from any web browser, using custom control panels built with Helm Designer. And because it’s built on HTML5 standards, there’s no need for browser plugins.
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AVN-HD1 Digital Headphone Amp for AVN-PD8/D Portal
The AVN-PD8/D can be combined with multiple Sonifex AVN-HD1 headphone amplifiers to provide 8 separate headphone signals where each headphone amplifier can be sent a separate feed, mixed from any physical or stream inputs.
- Front panel 6.35mm (1/4”) headphone socket and volume control knob.
- Front panel Mute/GPO push button.
- AES3 digital input on RJ45 (the connector provides power to the unit and a GPO back to the portal).
- AES3 digital output on RJ45 (power and GPO signal are not connected).
- Locking DC power connector if a portal is not being used to supply the unit with power.
Note: The AVN-HD1 is an AES3 digital input product taking an AES3 audio feed from the AVN portals. It can be used independently of the portals by using the separate DC input for power and a separate AES3 audio input.
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AVN-HA1 Analogue Headphone Amp for AVN-PA8/D & AVN-PM8/D Portals
The AVN-PA8/D and AVN-PM8/D can be combined with multiple Sonifex AVN-HA1 headphone amplifiers to provide 8 separate headphone signals where each headphone amplifier can be sent a separate feed, mixed from any physical or stream inputs.
- Front panel 6.35mm (1/4”) headphone socket and volume control knob.
- Front panel Mute/GPO push button.
- Analogue audio input on RJ45 (the connector provides power to the unit and a GPO back to the portal).
- Loop through audio output on RJ45 (power and GPO signal are not connected).
- Locking DC power connector (if a portal is not being used to supply the unit with power).
Note: The AVN-HA1 is an analogue input product taking an analogue audio feed from the AVN portals. It can be used independently of the portals by using the separate DC input for power and a separate analogue input.
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Oracle Acme Packet Platforms
Technical and Business Benefits
- Wide range of platforms to meet different customer needs
- Consistent set of features across platforms provides the extra flexibility in deployment
- Scalable network performance and capacity
- Advanced DoS/DDoS protection, utilizing hardware-assisted, dynamic, fine-grained controls
- Choices in network interfaces and DSP modules for customized solutions
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Quarra PTP Ethernet Switches
With the transition to IP networks for all aspects of the signal processing path, accurate timing becomes more difficult, due to the fundamentally asynchronous, non-deterministic nature of packet-based networks. Fortunately, a solution is available, in the form of IEEE 1588 Precision Time Protocol (PTP). When used properly, this technology can synchronize device clocks to within nanoseconds across a large network with many hundreds of nodes. When these clocks are derived from GPS (Global Positioning System) signals, PTP can provide a very accurate and stable timebase for all types of signals within modern media operations.
The Quarra line of PTP IP solutions compliment Artel's media delivery portfolio offering end users an excellent solution set for delivering real-time media IP network. As the broadcast industry continues to migrate toward all-IP networks, providing reliable solutions to manage IP-based workflows becomes paramount to maintaining efficiencies and ensuring the delivery of mission-critical multimedia streams.
The Quarra PTP Ethernet switches provide leading-edge precision timing protocol switch technologies and have become de-facto standards for the AES67/ SMPTE ST-2059-2 (IEEE1588) audio market. Designed for Carrier Class applications where accurate timing and control is required, Quarra switches are aimed at the professional markets of Audio/Video Broadcast, Defense & Security, Finance, Utilities, Telecom, and Enterprise IT. The Quarra switches are designed as either a standard 1RU rackmount unit with dual power supplies, or a half-width 1RU unit with a single power supply, the switches are available in several configurations of 1 Gigabit and 10 Gigabit interfaces, supporting a best-in-class layer 2 IEEE 1588v2 algorithm. A 1pps external reference input from GPS is included as standard, and options include an internal layer 1 Synchronous Ethernet module. The Quarra switches support SMPTE Standards-based Timing ST 2110-10 and ST 2059-2 and are RAVENNA AES67 approved. The Quarra switches also support IGMP v2/3, QoS, protection switching, VLAN, MEF service delivery and network OAM.
Ideal for:
• Live video/audio broadcast/production over IP utilizing IEEE1588 timing, QoS and Multicast filtering.
• Broadcast DTT/DTV IP distribution, utilizing IEEE1588 and SyncE timing to replace or backup GPS timing at base stations.
• Telecom networks, providing high level of network timing.
• Financial/trading networks, utilizing our IEEE1588 precise nanosecond time synchronization.
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SMART Media Delivery Platform
•Integrated Switching Capability - A non-blocking layer 2/3 switching and routing engine has been incorporated into the SMART platform to interface with the network eliminating the need for an external switch or router. The integrated switching capability allows for deep packet inspection, traffic management, flow control, protection switching, and link aggregation that can be easily configured through an SNMP interface.
•Control and Management - Simplifying the migration to an IP infrastructure, the SMART platform’s control and management interface allows the ease of integration, operation, problem detection and isolation.
•Designed for Service Provider and Broadcast Networks - The SMART platform is designed to support transport and data services, and broadcast network video, audio, and IT network applications. With a single module, multiple ports of media may be aggregated and injected into the network without the need for different physical elements. The SMART supports SMPTE ST 2022-7 hitless switching, VLAN tagging, QoS, and traffic management. These functions along with greater port density drastically reduces power consumption, size and cost.
•Software Defined Functionality - The migration to IP has increase the need for more agile solutions. The SMART platform is designed to update functionality and add new applications via software download providing a solution to manage multi-platform content delivery.
The SMART Media Delivery Platform’s small footprint, true data networking and transport versatility, software-defined functionality, and high density opens up new opportunities for media transport and broadcast applications and allows for a seamless transition as IP networks evolve.
A finalist for the 2018 IABM BaM Awards in the Connect category, the SMART Media Delivery Platform™ is a carrier-grade, software-defined platform with integrated nonblocking Layer 2/3 switching and routing capabilities. Designed to attach seamlessly to the IP network without the need for external network elements, the SMART platform supports SMPTE ST 2022: 1, 2, 5, 6, and 7 hitless switching; QoS; VLAN tagging; and traffic management. The platform features four video ports for transport of video, audio, and ancillary data and four GigE data ports bridged to a 10G interface. The SMART solution is software-enabled, providing an easy and efficient platform for adding or upgrading functionality via software download.
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MPA1000
The Macnica MPA1000 4K AV over 1GbE Module merges the promise of AV over IP with the reliability and performance of ST 2110 in a module that has that has what it takes to deliver encrypted UHD 60P 4:4:4 content and control over a standard 1GbE connection.
The MPA1000 Module is field configurable for transmit or receive. Only one part number is all that is needed for delivering routable UHD video, audio, USB, and HDMI control signals (EDID, HDCP, HPD, CEC, etc) over IP.
Once connected to the network, the module’s traffic is discoverable through the RESTful NMOS protocol - an open standard for publishing and discovery of content streams in broadcast. Simple configuration and control is the core principle in the design of this module. Every module includes a built-in web server with a RESTful API and a web application that is ready for customization. Host access is provided through an optional dedicated standard interface (Ethernet or serial).
Additionally, Macnica offers the MPA1000 Development Kit for test and development. Design files available upon request (currently for limited alpha customers only). Please contact Macnica directly for price and availability.
- 4K AV over 1GbE based on SMPTE ST 2110 standards
- Capable of transporting HDMI2.0 4K60p 4:4:4 HDMI 2.0 AV and Control Signals (Configurable as Rx or Tx) over standard 1Gigabit Ethernet network
- Encrypted USB 2.0
- Sub-frame Latency
- JPEG-XS based Visually-Lossless Compression
- Synchronous Sources
- Full-stack API with Web Interface and Scripting Built In
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TS Splicer
Applications include Ad Insertion, Slate insertion (covering content) or live content replacement.
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TS Switch
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ME10 SoC
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ME10/MPA1000 Launch Kit
Everything you need to begin developing 4K AV-over-IP products with IPMX interoperability, all preconfigured so you can begin using it right away.
Use the Launch Kit to evaluate how standards-based video will work with your current products, to develop new products, and as a reference to the connections and configurations you will need as you go into production.
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