Dante-enabled Product Catalog

With an open-architecture design, twelve acoustic echo cancellers (AEC), 16 x 16 Dante® connectivity, and flexible signal processing, the ControlSpace EX-12AEC conferencing signal processor provides robust expansion for conference rooms using the ControlSpace EX-1280C processor. With an open-architecture design, twelve acoustic echo cancellers (AEC), 16 x 16 Dante® connectivity, and flexible signal processing, the ControlSpace EX-12AEC conferencing signal processor provides robust expansion for conference rooms using the ControlSpace EX-1280C processor. ControlSpace Designer software simplifies the setup process with intuitive drag-and-drop programming and conference-specific software tools that help reduce installation time and on-site errors. Features

• Channel AEC expansion with multiple references to complement ControlSpace EX-1280C processors in larger systems
• Adaptable noise cancellation with non-linear processing and comfort noise on each AEC channel enhances clarity
• Dante audio networking supports 16 x 16 audio channels for connection to an EX-1280C processor and native Dante-integrated microphones
• Front-panel interface features a large OLED display and rotary encoder for setting network parameters and monitoring channel activity

Applications

• Boardrooms
• Large conference rooms
• Distance learning
• Multi-purpose spaces
• Divisible Conference Rooms
• Operations Centers

The APB-112 OW-D is an active, on wall AudioPressBox unit with 1 channel DANTE input and 12 LINE/Mic outputs. The DANTE-enabled on wall unit features 1 channel DANTE input with high quality digital to analog converter. Powered via PoE from DANTE-network. The unit has 12 balanced, individually transformer isolated Line/MIC outputs. The unit is powered only via PoE from DANTE network. This is a PoE level 3 device

MIMO4040DN is a digital matrix with native processing of 40 inputs and 40 outputs (8 analogue inputs, 8 analogue outputs, 32 Dante™ inputs and 32 Dante™ outputs). MIMO4040DN is a digital matrix with native processing of 40 inputs and 40 outputs (8 analogue inputs, 8 analogue outputs, 32 Dante™ inputs and 32 Dante™ outputs). Incorporating audio over IP technology, MIMO4040DN allows signal transmission using a Local Ethernet network. It features accessories from WPNET Series connected by Ethernet. MIMO4040DN is also capable of executing EclerNet projects thanks to its processor acting as a UCP server. Dimensions 482,6x44x266,5mm. Weight 3.25 kg

Ecler PAGENETDN is a desktop paging console for voice notices with Dante™ audio over IP.  Ecler PAGENETDN is a desktop paging console for voice notices with Dante™ audio over IP. Features a 10” touch screen, a server client and a UCP server (User Control Panel). Since it’s a server/UCP client device, it’s possible to add other kind of control panels beside the ones destined for voice messages such as: volume control, sound source selection, presets recovery, communication via TCP/UDP with external devices, etc. Power over Ethernet or local power supply. Gooseneck microphone included. 

The 9821EMR-AG-HUB is a 1RU platform that is optimized for operation in 10GbE and 25GbE IP SDVN environments, supporting SMPTE ST 302M, AES67*, TDM, MADI and DANTE (up to 512x512 channels of Dante).  The 9821EMR-AG-HUB is a 1RU platform that is optimized for operation in 10GbE and 25GbE IP SDVN environments, supporting SMPTE ST 302M, AES67*, TDM, MADI and DANTE (up to 512x512 channels of Dante). With TDM connections it can externally support AA, AES, EQX embedded audio, HYDRA2 and ALINK. The 9821EMR-AG-HUB provides a gateway to link IP infrastructures with discrete audio IO directly and by using any of the existing Evertz TDM enabled Audio interfaces. It has a number of functional use cases including translating between AoIP standards including SMPTE ST 302M, AES67* and Dante, for audio bridging between devices supporting SMPTE ST 302M or AES67* to a common core IP audio format. AES67 to AES67 for profile grooming*. Bulk audio connection between audio systems such as connecting multiple mobile broadcast vehicles that are using EMR audio router modules.

The 9821EMR-AG-HUB is a 1RU platform that is optimized for operation in 10GbE and 25GbE IP SDVN environments, supporting SMPTE ST 302M, AES67*, TDM, MADI and DANTE (up to 512x512 channels of Dante).  MDi Series is the newest solution from RAM Audio for fixed installations, where state-of-the-art amplification technology has been used to manage acoustic systems in the most efficient and greenest way possible, without diminishing quality of sound and dynamics that have always typified RAM Audio amplifiers. A new generation of QuantaPulse switching mode power supplies with Power Factor Correction to improve efficiency, with a schematic thought to give a very high capacity of instantaneous power, together with very evolved high efficiency audio modules in class D with an excellent quality of sound, are the basis on which RAM Audio has developed this new series that is capable of offering the most optimal solution to the very exigent fixed-installations market. Series models:

• MDi2-1K4
• MDi2-2K7
• MDi2-6K
• MDi4-2K4
• MDi4-6K
• MDi4-12K
• MDi8-2K7
• MDi8-6K

The AMP203 is a revolutionary small amplifier, featuring network input through Dante providing a complete media networking solution to distribute uncompressed audio via standard Ethernet networks with near-zero latency, while allowing all other data to be connected on the same network. The AMP203 is a revolutionary small amplifier, featuring network input through Dante providing a complete media networking solution to distribute uncompressed audio via standard Ethernet networks with near-zero latency, while allowing all other data to be connected on the same network. It features various DSP functionalities, including Low-pass, Band-pass and High-pass filters. All configurations can be made via RS485 and Ethernet. The output connector has been implemented using a 4-pin terminal block connector, featuring an output power of 30W per channel and 60W when bridged. Using PoE (Power over Ethernet) the amplifier receives both power and signal through a single networking cable, this way flexibility is maximized while needed cabling is kept at a minimum. The compact convection cooled enclosure eliminates any humm or buzz otherwise caused by a fan. Various optional mounting brackets are available making it ideal for mounting under a desk, in a closet or on the wall, 19” rack mounting hardware is available as well.

The “Swiss Army Knife” of small amplifiers, the RevNet™ 2140-D gives you the power, versatility and quality you need for both commercial and high-end residential installations The “Swiss Army Knife” of small amplifiers, the RevNet™ 2140-D gives you the power, versatility and quality you need for both commercial and high-end residential installations. One RevNet amp will power two speakers. Units may be daisy-chained for larger applications, and the RevNet 2140-D has the added power of full Dante® compatibility with AES67 support. Quality Loads of headroom with 26 dB gain, coupled with the lowest distortion you’ll find on any comparable pro audio amp. Audiophile-quality RevNet amps are suitable for both pro audio and high-end HiFi operation. Powerful 140 watts per channel (280 W in mono) will power nearly any installed speaker in the room. RevNet amps can be used in 4/8/16 Ohm or 70/100V installations Easy to Install Fast-wiring Phoenix connectors allow you to quickly pre-wire sources and then just plug them into the amp. Versatile Control Direct RVC connections along with USB, TCP/IP, WiFi or Dante make it easy to manage your audio sources and volume. Onboard DSP for Easy Configuration A built-in sound masking generator allows you to quickly configure the amp for corporate uses, while 31 band EQ and RMS compressor bring you complete control over the sound of the room. Specifications

• 2 x 140W@16/8/4 , 100V/70V @280W
• Phoenix connectors for Inputs/Link Output, Amp Out and RVC
• Dante® (with AES67 support) and USB interfaces along with Integrated WiFi for remote control and set up
• Built-in sound masking generator
• 31 band fully customizable DSP
• In-Room AutoEQ and mobile app support

For the touring market, a new 1U X-Light Bridge interface designed to be deployed within stage I/O racks, features a fully redundant pair of X-Light connections using Neutrik opticalCON Quad format connectors on the front panel.  For the touring market, a new 1U X-Light Bridge interface designed to be deployed within stage I/O racks, features a fully redundant pair of X-Light connections using Neutrik opticalCON Quad format connectors on the front panel. On the rear panel there are two redundant pairs of Neutrik etherCON format Dante connections for audio and control distribution. Up to eight SB 32.24 stageboxes can now be connected via a pair of network switches to a single X-Light Bridge redundantly at 96 kHz with gain control of each input. There are no big workflow changes either! X-Light Bridge provides a seamless I/O boost for the user, as the routing process in the console interface is exactly the same via the X-Light Bridge as it is via the existing local Dante Expander Module. This includes the ability to configure the X-Light channel capacity as Virtual Tie Lines, for routing in Dante Controller, or as Dual Domain Routes for seamless routing and control from console channels.

For fixed installations where ruggedised connections are not required, the BL II Bridge also offers 256 channels of 96 kHz audio between a Live console and a Dante network by utilising SSL’s proprietary high bandwidth Blacklight II protocol. For fixed installations where ruggedised connections are not required, the BL II Bridge also offers 256 channels of 96 kHz audio between a Live console and a Dante network by utilising SSL’s proprietary high bandwidth Blacklight II protocol.

The SB 16.12 is a 3U ruggedised enclosure featuring dual redundant power supplies, 16 mic/line inputs, 12 analogue line outputs and 4 digital inputs and outputs on 2 AES3 input/output pairs.  The SB 16.12 is a 3U ruggedised enclosure featuring dual redundant power supplies, 16 mic/line inputs, 12 analogue line outputs and 4 digital inputs and outputs on 2 AES3 input/output pairs. It has a pair of redundant RJ45 Dante network connections in addition to a user configurable SFP port that can be fitted with RJ45 or optical connectors. These can be used for network extension or to provide network separation for the gain-compensated Dante “split”, for connection to a second Dante-equipped console or appropriately equipped device on a different network. It has individual signal present, clip and phantom power LED’s as well as global indication of PSU, Network A and B and Hardware status. SB 16.12 can operate at 96kHz or 48kHz sample rates.

Like all SSL consoles the secret to the success of the L350 is SSL’s dedication to great sound. Part of the success is attributable to exemplary hardware performance throughout the signal chain: bringing SSL Studio grade mic pre’s to the stage, outstanding audio conversion, 96kHz operation and magnificent summing. Since their introduction in 2013 SSL Live consoles have met widespread praise from many of the world’s leading live engineers. SSL Live consoles are designed to help highly skilled operators deliver outstanding performances in the most demanding applications in Touring, Installation, Houses Of Worship and Theatre. Like all SSL consoles the secret to the success of the L350 is SSL’s dedication to great sound. Part of the success is attributable to exemplary hardware performance throughout the signal chain: bringing SSL Studio grade mic pre’s to the stage, outstanding audio conversion, 96kHz operation and magnificent summing. There is a sublime transparency, definition and detail in the fundamental sound of the L350 that is difficult to describe but is immediately recognisable as the SSL signature. An equal, perhaps even greater part of the secret of the power of the L350 is however the way it enables operators to perform at their very best. The deceptively simple ergonomic excellence of the combination of multi-touch screen control with assignable hardware controls, the use of colour coding, the open architecture which lets engineers configure the console layout to suit their own way of working, the arsenal of processing options, the precision and depth of those processing tools… and so much more… it all combines to let the operator get where they can hear they want to be extremely quickly. Like all great creative audio technology, SSL Live consoles are instruments for engineers. Control The key to a great control surface is a clear view of everything in your audio environment and finding exactly the control you need at your fingertips when you need it. With the Live SSL’s obsession with ergonomics and workflow has delivered an intelligent, elegant and comfortable layout. All of the most commonly used functions are carefully arranged so that they sit within reach where your hands naturally fall on the console. There is a wealth of visual feedback with carefully considered use of colour change technology that will not fatigue the user during long periods of operation. The Live control surface delivers a genuinely intuitive combination of gestural touch screen & hardware control and a whole collection of innovative features designed to streamline workflow. Power The power within SSL Live consoles is provided by our new ‘Tempest’ platform, developed specifically to meet the demands of live production. We are of course very proud of our clever new processing technology, but at SSL our focus is always on what really matters and that is you having the power to do your job well. Live harnesses Tempest’s power in a sensibly flexible way to let you balance allocation of resource between signal processing and console architecture to suit each project. Thanks to the intrinsic flexibility of our approach, no matter how you configure it, when you compare the numbers, Live gives you more Inputs and Outputs, more Channels, Stem Groups, Auxes, VCA’s and Masters, more processing tools and more signal processing power than many consoles with much bigger price tags. Tempest Live consoles are powered by Tempest, a new fourth generation digital audio platform developed specifically to meet the demanding requirements of the live environment. Learning from more than 25 years of digital console development, Tempest is a ground up development. All legacy code – the key to the sonic excellence of previous generations of SSL digital consoles – has been rewritten in the latest software languages. The high capacity signal processing engine at the heart of Tempest uses SSL’s patented Optimal Core Processing (OCP) to deliver highly efficient and reliable performance with minimal latency. Tempest delivers all of the power required to equip Live with all of the mixer architecture, processing power and connectivity to handle large scale events. The Live signal processing engine is integrated entirely within the control surface.

 Like all SSL consoles the secret to the success of the L550 is SSL’s dedication to great sound. Part of the success is attributable to exemplary hardware performance throughout the signal chain: bringing SSL Studio grade mic pre’s to the stage, outstanding audio conversion, 96kHz operation and magnificent summing. In just a few years SSL Live consoles have met widespread praise from many of the world’s leading live engineers. SSL Live consoles are designed to help highly skilled operators deliver outstanding performances in the most demanding applications in Touring, Installation, Houses Of Worship and Theatre. Like all SSL consoles the secret to the success of the L550 is SSL’s dedication to great sound. Part of the success is attributable to exemplary hardware performance throughout the signal chain: bringing SSL Studio grade mic pre’s to the stage, outstanding audio conversion, 96kHz operation and magnificent summing. There is a sublime transparency, definition and detail in the fundamental sound of the L550 that is difficult to describe but is immediately recognisable as the SSL signature. An equal, perhaps even greater part of the secret of the power of the L550 is however the way it enables operators to perform at their very best. The deceptively simple ergonomic excellence of the combination of multi-touch screen control with assignable hardware controls, the use of colour coding, the open architecture which lets engineers configure the console layout to suit their own way of working, the arsenal of processing options, the precision and depth of those processing tools… and so much more… it all combines to let the operator get where they can hear they want to be extremely quickly. Like all great creative audio technology, SSL Live consoles are instruments for engineers... Control The key to a great control surface is a clear view of everything in your audio environment and finding exactly the control you need at your fingertips when you need it. With the Live SSL’s obsession with ergonomics and workflow has delivered an intelligent, elegant and comfortable layout. All of the most commonly used functions are carefully arranged so that they sit within reach where your hands naturally fall on the console. There is a wealth of visual feedback with carefully considered use of colour change technology that will not fatigue the user during long periods of operation. The Live control surface delivers a genuinely intuitive combination of gestural touch screen & hardware control and a whole collection of innovative features designed to streamline workflow. Power The power within SSL Live consoles is provided by our new ‘Tempest’ platform, developed specifically to meet the demands of live production. We are of course very proud of our clever new processing technology, but at SSL our focus is always on what really matters and that is you having the power to do your job well. Live harnesses Tempest’s power in a sensibly flexible way to let you balance allocation of resource between signal processing and console architecture to suit each project. Thanks to the intrinsic flexibility of our approach, no matter how you configure it, when you compare the numbers, Live gives you more Inputs and Outputs, more Channels, Stem Groups, Auxes, VCA’s and Masters, more processing tools and more signal processing power than many consoles with much bigger price tags. Tempest Live consoles are powered by Tempest, a new fourth generation digital audio platform developed specifically to meet the demanding requirements of the live environment. Learning from more than 25 years of digital console development, Tempest is a ground up development. All legacy code – the key to the sonic excellence of previous generations of SSL digital consoles – has been rewritten in the latest software languages. The high capacity signal processing engine at the heart of Tempest uses SSL’s patented Optimal Core Processing (OCP) to deliver highly efficient and reliable performance with minimal latency. Tempest delivers all of the power required to equip Live with all of the mixer architecture, processing power and connectivity to handle large scale events. The Live signal processing engine is integrated entirely within the control surface.

Kramer's AFM-20DSP is a 20-port audio matrix with DSP and interchangeable inputs & outputs. AFM‑20DSP provides configuration flexibility and eliminates I/O limitations with its 20 interchangeable analog audio ports, letting AV professionals get more functionality for less real estate. Kramer's AFM-20DSP is a 20-port audio matrix with DSP and interchangeable inputs & outputs. AFM‑20DSP provides configuration flexibility and eliminates I/O limitations with its 20 interchangeable analog audio ports, letting AV professionals get more functionality for less real estate.  AFM-20DSP also packs 8 AEC channels, auto feedback suppression, Dante™ support, built-in Hi-Z and Lo-Z 120W power amplification, and complete DSP functionality powered by Xilica – who has been a prominent name in the development of class-leading configurable DSP systems since its establishment in 2002. Not overlooking video requirements, AFM-20DSP provides HDMI™ embedding and de-embedding at 4K60 4:2:0. AFM‑20DSP even boasts both fixed and open architecture, enabling AV professionals to use predefined sound configurations or configure a system from scratch.    AFM-20DSP can be easily integrated in existing AV installations but is ideal for the most demanding boardrooms, multipurpose spaces, auditoriums, sports bars, houses of worship, live events, broadcast facilities, and recording studios.

 The Intelli-Series range of DSP-based system management and loudspeaker processing devices from Apex has gained a reputation for being both innovative and intuitive, always one step ahead of the game. The all-new Intelli-Power 10 is no exception.  The Intelli-Series range of DSP-based system management and loudspeaker processing devices from Apex has gained a reputation for being both innovative and intuitive, always one step ahead of the game. The all-new Intelli-Power 10 is no exception. Featuring dual DSP internal processing and high-end, studio-grade analogue circuit designs for premium sonic performance, Intelli-Power 10 combines a powerful 4 x 2500W amplifier with Apex’s renowned Intelli-Series DSP to set new standards in powered system management and loudspeaker processing design. Fully plug-and-play with integrated Dante audio networking capability plus fast and intuitive remote configuration, monitoring and control via Apex’s much-vaunted Intelli-Ware software, Intelli-Power 10 is suited to a wide range of applications. Equally at home as a standalone system amplifier configured using the front panel controls, or as a comprehensive wireless networked system for stadium or arena-sized tours, Intelli-Power 10 offers a range of features to easily integrate and optimise any system, starting with the DSP; like the rest of the Intelli-Series, IP-10 provides all the system management tools necessary for comprehensive system alignment and EQ through to provision of IIR and custom FIR-based crossovers. I/Os can be selected (in pairs) to be either analogue, AES/EBU or Dante, enabling a mix of audio formats to be used simultaneously. Internal three-way Ethernet switches enable the device to interface easily into any existing or bespoke Ethernet network. Both automatic IP (DHCP) and static IP configuration is provided as standard for straightforward plug-and-play connectivity. A fully redundant Dante module enables easy integration into a Dante IP audio network. In terms of synchronisation and redundancy, the IP-10 provides a three-tiered redundancy approach which ensure that if any one of the four sync sources (Dante, AES A/B, AES C/D and Internal) is lost, the device will automatically revert to a different source ensuring no loss in audio.

The DI-4.1000 has an output power of 1000 W / 4 ohms per channel, four input- and four output channels. It has a compact 2U / 19“ enclosure and weighs just 9.4 kg. The amplifier works extremely quietly and its fans are not used while idling.  The new DI-Series digital amplifiers have been developed and manufactured in house by Fohhn in Germany. At every stage of development, new intelligent technologies have been implemented. The DSP-controlled amplifiers perform at the highest level and guarantee maximum operational reliability, highest power density and excellent sound. Since 2002, Fohhn has been developing extremely high quality, reliable digital electronics that have been consistently incorporated into the company’s own product lines. The DI-Series amplifiers represent 15 years of know-how in digital electronics, with a great many customer wishes brought to fruition as a result. Developed for mobile sound reinforcement applications and fixed installations, the amplifiers offer all the essential features plus virtually unlimited connectivity with regard to audio signal inputs. Even if you opt right now for an AES/EBU, Dante™, Optocore or Analog option, you are not permanently committed: The optional input interfaces can be changed, as required, customizing your amplifier to suit the relevant digital or analogue network. Extensive DSP input and output processing enables you to intelligently adapt your loudspeaker systems to the particular sound reinforcement situation and room acoustic. All DSP functions can be controlled in real time using FOHHN AUDIO SOFT. The DI-4.1000 has an output power of 1000 W / 4 ohms per channel, four input- and four output channels. It has a compact 2U / 19“ enclosure and weighs just 9.4 kg. The amplifier works extremely quietly and its fans are not used while idling.  Features

• developed for mobile applications and fixed installations
• network compatible, remote control, remote monitoring
• extensive DSP input and output processing
• input options: AES/EBU (terminal block or XLR), Dante™, Optocore or Analog
• output options: terminal block or Speakon connectors
• microprocessor-controlled universal power supply, 100-240 V AC, with Power Correction Factor (PFC) and sensory functions for mains voltage, power consumption and temperature
• Auto Power Save for fan-free idling, temperature-controlled fans
• two configurable switching contacts, one fault contact
• performance presets for all relevant Fohhn loudspeakers
• up to 10 seconds peak power

The DI-2.2000 has an output power of 2000 W / 4 ohms per channel, four input- and two output channels. It has a compact 2U / 19“ enclosure and weighs just 9.4 kg. The amplifier works extremely quietly and its fans are not used while idling.  The new DI-Series digital amplifiers have been developed and manufactured in house by Fohhn in Germany. At every stage of development, new intelligent technologies have been implemented. The DSP-controlled amplifiers perform at the highest level and guarantee maximum operational reliability, highest power density and excellent sound. Since 2002, Fohhn has been developing extremely high quality, reliable digital electronics that have been consistently incorporated into the company’s own product lines. The DI-Series amplifiers represent 15 years of know-how in digital electronics, with a great many customer wishes brought to fruition as a result. Developed for mobile sound reinforcement applications and fixed installations, the amplifiers offer all the essential features plus virtually unlimited connectivity with regard to audio signal inputs. Even if you opt right now for an AES/EBU, Dante™, Optocore or Analog option, you are not permanently committed: The optional input interfaces can be changed, as required, customizing your amplifier to suit the relevant digital or analogue network. Extensive DSP input and output processing enables you to intelligently adapt your loudspeaker systems to the particular sound reinforcement situation and room acoustic. All DSP functions can be controlled in real time using FOHHN AUDIO SOFT. The DI-2.2000 has an output power of 2000 W / 4 ohms per channel, four input- and two output channels. It has a compact 2U / 19“ enclosure and weighs just 9.4 kg. The amplifier works extremely quietly and its fans are not used while idling.  Features

• developed for mobile applications and fixed installations
• network compatible, remote control, remote monitoring
• extensive DSP input and output processing
• input options: AES/EBU (terminal block or XLR), Dante™, Optocore or Analog
• output options: terminal block or Speakon connectors
• microprocessor-controlled universal power supply, 100-240 V AC, with Power Correction Factor (PFC) and sensory functions for mains voltage, power consumption and temperature
• Auto Power Save for fan-free idling, temperature-controlled fans
• two configurable switching contacts, one fault contact
• performance presets for all relevant Fohhn loudspeakers
• up to 10 seconds peak power

Designed originally for EVS operators and monitoring stations in outside broadcast trucks, with 20 Dante and 2 analogue inputs it is also a great general purpose monitor for any racks room or studio. Bella 22 is a summing matrix confidence monitor designed for outside broadcast trucks, studios, theatre and professional audio applications. It is perfect for easy and cost effective monitoring of multiple network audio sources, ideal for busy production environments and perfect for monitoring multiple network intercom & programme feeds. The 2 front panel loudspeakers are driven from a DSP to compensate for their size, the result is surprisingly good, with clear crisp vocals and highly intelligible reproduction of wider band audio sources. They are driven from a class D amplifier and have more than sufficient output level for most environments. Robust proven construction techniques, simple reliable interface and excellent specification will help make your technician’s life hassle free. Whilst the low cost and long asset life will keep the accountant satisfied.

The Glensound B1 is an ultra-compact single channel belt pack intercom with crystal clear audio designed for broadcast, theatre and professional audio applications. Size matters, so we spent a long time making sure the B1 is as small and lightweight as possible without compromising durability. The Glensound BEATRICE B1 is an ultra-compact, robust, portable single channel beltpack intercom designed for broadcast, theatre and professional audio applications. This small beltpack was designed to be very easy to use for the operator and simple to setup for the technician. It includes all the basic functionality required for small intercom systems and none of the overly complex installation requirements normally associated with large systems. Being part of our Beatrice range it includes standard features such as being powered by PoE for simple setup, a Call Facility for getting other users attention, LED presence indicators to help identify who is talking, Crystal Clear audio, IFB looping facility and no central matrix. It also has a GPI for external triggering of the talk switch making it ideal for camera operators.

The Glensound B2 is an ultra-compact two channel belt pack intercom with crystal clear audio designed for broadcast, theatre and professional audio applications. Size matters, so we spent a long time making sure the B2 as small and lightweight as possible without compromising durability. This small beltpack was designed to be very easy to use for the operator and simple to setup for the technician. It includes all the basic functionality required for small intercom systems and none of the overly complex installation requirements normally associated with large systems. Being part of our Beatrice range it includes standard features such as being powered by PoE for simple setup, a Call Facility for getting other users attention, LED presence indicators to help identify who is talking, Crystal Clear audio, IFB looping facility and no central matrix. It also has a GPI for external triggering of the talk switch making it ideal for camera operators.

Glensound invented the Broadcaster’s mobile phone and our GS-MPI005 and GS- MPI005HD mobile phone racks have been hugely popular with broadcasters and can be found in daily use in outside broadcast trucks and studios around the World. On the surface the new GS-MPI005HD MKII looks different to its predecessors as it has touch screen colour displays for each phone module fitted. A traditional telephone handset interface is also now provided to allow an operator to communicate off air before handing the call over to broadcast. The handset interface can also dial and answer calls. No new modern piece of audio broadcast equipment is complete without an AoIP solution so the new GS-MPI005HD MKII has a Dante® interface. Of course the crystal clear sound and audio inputs & outputs that are interference free of cell phone noise that made the original units so popular also feature in this new design.

Athens 2 is twin stereo phono preamplifier with excellent RIAA equalisation suitable for both moving magnet and moving coil turntable cartridges. It has redundant network circuits with 2 x copper & 2 x SFP interfaces. Athens 2 is twin RIAA phono preamplifier with Dante® and AES67 network audio outputs. It is a mains powered device with two high quality stereo turntable preamplifiers inputs. Each preamplifier has its own set of preset controls to allow accurate matching of the Athens 2 input circuitry to the turntable's cartridge. Input gain, input impedance and input capacitance can all be set for each of the two stereo inputs.The Athens 2 is suitable for use with both moving coil and moving magnet cartridges. To provide the best possible equalisation of the RIAA curve the preamps feature high quality high tolerance analogue circuitry followed by a high quality analogue to digital converter and DSP. The combination of the analogue circuitry and DSP provides a very good quality RIAA equalisation resulting in a very flat output.

ERIS ONE is a Dante audio interface which provides the user with a high performance, low latency and uncompressed digital audio networking.  ERIS ONE is a Dante audio interface which provides the user with a high performance, low latency and uncompressed digital audio networking. The equipment offered standard multipurpose I/O commonly used in the professional audio environment. Eris Ons is versatility, scalable architecture and high-performances it can be fully used in the most demanding recording studios, live music, audiovisual installations and broadcast environments. It is designed to be used quite independently, all the ERIS ONE functionality is accessible through an easy and intuitive interface or operated remotely. As a system, it offers the user scalability to adapt its necessities through IOX extensions, allowing to increase I/O capacity in a modular way. The internal and external elements are placed to achieve the peak heat dissipation without the need for forced ventilation. This characteristic gives the longest possible life and most reliable machine. In addition to this, it makes it less vulnerable to dust contaminants and humidity conditions.. The compound audio processor that according to its application lets the user choose and upload the wanted audio effects. All this leads to an efficient, strong and adaptive system. The conversion system possesses programmable oscillators which enable us to provide our users the highest accuracy and digitally conversion quality in each sample without having to rely on temperature settings. The ERIS ONE front panel, is fully machined as a single piece of aluminum. Silicone backlit buttons, the metallic controls and the chassis in a U position confers a robust and long life-span equipment.

TF-RACK is a compact rack-mount version of the TF Series digital mixing consoles, offering the same levels of performance and groundbreaking operability. TF-RACK is a compact rack-mount version of the TF Series digital mixing consoles, offering the same levels of performance and groundbreaking operability.

• All-in-One rack-style digital mixer
• Inputs: 16 mic/line (XLR/TRS combo) + 1 stereo line (RCA pin)
• Output: 16 (8 XLR + 8 TRS phone)
• Channels: 40 (32 mono + 2 stereo + 2 return)
• Aux Buses: 20 (8 mono + 6 stereo)
• Stereo Buses: 1
• Sub Buses: 1
• Dimensions(WxHxD): 480 mm (18-7/8 in) x 132 mm (5-1/4 in) x 409 mm (16-1/8 in)
• Net weight: 9.2 kg (20.3lb)

AVN-AIO4 is a multi-channel Dante® Audio interface, which provides 4 balanced analogue inputs and outputs to and from the Dante network.  AVN-AIO4 is a multi-channel Dante® Audio interface, which provides 4 balanced analogue inputs and outputs to and from the Dante network. Simple to configure and operate, these cost effective rack-mount solutions offer an easy solution for AV professionals and system integrators. Packaged in a 1U form factor, they use Neutrik XLR connectors, are Dante® AES67 compliant and use PoE for power.

AVN-AIO8 is a multi-channel Dante® Audio interface, which provides 8 balanced analogue inputs and outputs to and from the Dante network.  AVN-AIO8 is a multi-channel Dante® Audio interface, which provides 8 balanced analogue inputs and outputs to and from the Dante network. Simple to configure and operate, these cost effective rack-mount solutions offer an easy solution for AV professionals and system integrators. Packaged in a 1U form factor, they use Neutrik XLR connectors, are Dante® AES67 compliant and use PoE for power.

The NJR-AB08DAN is an audio bridge interface that converts audio signal protocols between IP-NINJAR and Dante networks. The NJR-AB08DAN converts audio packet on the IP-NINJAR network into the suitable Dante packets and vice versa for routing signals efficiently to the correct destination. It supports up to 8×8 audio I/O channels.

• Transcode Audio signal between IP- NINJAR (SDVoE) and Dante protocol
• Receives up to four (4) audio streams from IP-NINJAR transmitters and output as Dante protocol up to eight (8) channels
• Receives up to 8 channel Dante audio from Dante devices and output as IP-NINJAR (SDVoE) protocol up to four (4) audio streams
• Dante audio embedded on HDMI signal at IP-NINJAR receiver
• Dante audio de-embedded to analog audio signal at IP-NINJAR receiver