Dante-enabled Product Catalog

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

The BluePOD Solo Bluetooth® audio conferencing system is designed to simplify making conference calls in your meeting rooms with your mobile phone or laptop. The BluePOD Solo Bluetooth® audio conferencing system is designed to simplify making conference calls in your meeting rooms with your mobile phone or laptop. With BluePOD Solo, making a conference call is as simple as tapping the Bluetooth connect button on your phone; no special software is needed. The Bluetooth Wireless Access Point (B-WAP) ensures a strong and smooth Bluetooth connection. Adjusting the volume, muting and connecting is all done on your phone, just like you do it now. The BluePOD Solo makes it easy to add professional-grade Bluetooth Audio into your existing audio system. The simplified design and functionality of the BluePOD solo makes it easy to integrate into your current audio system setup.

The Versarray™ Pro 112 Powered Ribbon Tweeter Line Source Array speaker system consists of a 12″ Neo Black Widow® woofer combined with a Neodymium based Peavey RD™ 2.6 ribbon tweeter in a cabinet with a simple, quick, yet flexible rigging system.  Designed to provide modular coverage of small to medium venues, and intended for use with the companion Versarray™ Pro Sub models, the Versarray™ Pro 112 offers excellent versatility with a very high performance capability. The two-way system consists of the following driver components: a 12″ Black Widow ® Neo series woofer with Neodymium magnet structure. Capable of over 500W of continuous power handling (AES Std 2-1984), the woofer can handle a lot of sheer power. The high frequencies are handled by two Peavey RD™ 2.6 ribbon tweeters utilizing a composite sandwich ribbon, a Neodymium magnet system, and a low distortion waveguide.

Power for bi-amping is supplied by some very efficient power amplifier systems, controlled by a sophisticated and refined DSP operations system with Dante connection capability.  Total system power is 3400W total peak power, with 2000W peak power for the woofer, and 1400W total peak power for the two ribbon tweeters. This sheer power is controlled precisely and processed by a high performance DSP system, which provides all the crossover and EQ functions, as well as providing all limiting, compression and driver protection duties with unfailing attention to every detail of the music.

A unique set of control buttons on the rear panel allow analog signal input and use without the need for a network controller, by setting the system for typical use situations via push-button.  This feature allows the use of the system without the requirement for a full-blown digital networking controller or system.

The adjustable rigging system provides for a classic straight line-array configuration, or a number of different angling options, providing easy aiming of the system. Angles between the array modules is adjustable from 0 degrees (straight), to 7.5 degrees in 2.5 degree increments. Total angle possible between two cabinets is 15 degrees.

Quick-lock pins are supplied with the rigging hardware to couple the Versarray™ Pro 112 modules together and lock the angles between them into place, as well as for the rigging halo and fly bar configurations of a line array. The flexibility of the Versarray system allows the use of anywhere from 1 to 10 Versarray™ Pro 112 modules in conjunction with anything from one Versarray 118 Pro Sub to as many Versarray 218 or 215 Pro Subs as you can! An optional special groundstack bracket set mounts to the Versarray 218 Pro Sub, and allows up to three of the Versarray™ Pro 112’s to be mounted on top of the Versarray 218 Pro Sub, and angled upward, for use on stage in a stadium seating situation. 

The Versarray™ Pro 215 Sub Powered Subwoofer speaker system consists of a pair of 15″ Low Rider® woofers combined with a super-solid cabinet with a simple, quick, yet flexible rigging system.  Designed to provide modular coverage of small to medium venues, and intended for use with the companion Versarray™ Pro 112 models, the Versarray™ Pro 215 Sub offers excellent versatility with a very high performance capability. The subwoofer system consists of the following woofer components: a pair of 15″ Low Rider® series woofers with a very long throw 4″ voice coil structure. Capable of over 800W of continuous power handling (AES Std 2-1984), these woofers can project a lot of low frequency energy. 

 The Versarray™ Pro 215 Sub incorporates Peavey’s UniVent air pumping venting system.  The UniVent venting system literally pumps air through the enclosure, exchanging the stale hot air inside the cabinet for the cooler outside air. This helps keep the woofer operating temperatures from getting so high, and increases reliability and reduces power compression under heavy continuous drive conditions. The air pumping action is achieved without excessive turbulence or any significant net asymmetry of total vent air flow.

Power for both woofers is supplied by some very efficient power amplifier systems, controlled by a sophisticated and refined DSP operations system with Dante connection capability.  Total system power is 3000W total peak power, with 1500W total sine wave power for the woofers. This sheer power is controlled precisely and processed by a high performance DSP system, which provides all the crossover and EQ functions, as well as providing all limiting, compression and driver protection duties with unfailing attention to every detail of the music.

The adjustable rigging system provides for hanging multiple Versarray™ Pro 112 cabinets below a single Sub, or multiple Subs.  Quick-lock pins are supplied with the rigging hardware to couple the Versarray™ Pro 215 Subs together, and to the rigging Halo and Frames.  The flexibility of the Versarray™ FlyQWIK™ system allows the use of anywhere from 1 to 5 Versarray™ Pro 215 Subs in conjunction with an appropriate number of Versarray™ Pro 112 cabinets.

An optional special Sub Support Frame rigging adaptor mounts to the Versarray™ 215 Pro Sub, and allows Versarray™ Pro 112’s to be flown below the Sub, with the Sub being flown from a Versarray™ Halo.   See Crest Audio® Model Versarray™ Mk3 Halo Spec Sheet for details of the allowable mix of Versarray™ Subs to 2-ways.

LivePremier™ is a full range of modular and scalable 4K/8K multi-screen presentation systems and videowall processors, delivering uncompromising presentation experiences to high-end staging and premium system integration. LivePremier™ is a full range of modular and scalable 4K/8K multi-screen presentation systems and videowall processors, delivering uncompromising presentation experiences to high-end staging and premium system integration. In its largest chassis configuration, Aquilon offers up to 24 inputs and 20 outputs and up to 24x 4K layers. Aquilon combines industrial grade reliability, real-time 10/12-bit 4:4:4 video processing power and pure 4K60p on each input and output with ultra-low latency. The system is easy to configure and operate via its cutting-edge HTML5-based user interface. With the native Dante audio routing capability, up to 64 channels of audio can be embedded or de-embedded over a Dante network. The LivePremier™ Aquilon series is comprised of seven high-performance pre-configured products:

• RS alpha
• RS1
• RS2
• RS3
• RS4
• C
• C+

The two-way e-351 Nano Speaker redefines small loudspeaker capabilities by producing high output exceptionally high-quality sound form a very compact aluminum die-cast weather and corrosion resistant system. The two-way e-351 Nano Speaker redefines small loudspeaker capabilities by producing high output exceptionally high-quality sound form a very compact aluminum die-cast weather and corrosion resistant system. With reference quality flat response from 130 Hz to well above what the most discerning humans can hear, the e-352 is capable of output levels of systems typically twice the size. The e-351 utilizes applying state-of-the-art materials and technology to achieve audiophile quality at professional sound pressure levels. State-of-the-art 3.5-inch co-axial cone transducer achieves high output by combining a very powerful Neodymium magnetic circuit for efficiency with high-temperature voice coils and physical optimization of the magnet assembly for maximum heat dissipation to the aluminum enclosure. Coaxially mounted through the pole piece of the driver is a unique beryllium 25 mm compression tweeter with Neodymium magnetic assembly for low distortion extend high-frequency response. The passive crossover is precisely mated to the specialized DSP in the VUEDrive V242 Systems Engine to maximize response accuracy while providing maximum power and transducer protection. The die-cast aluminum enclosure provides considerable rigidity for resonance-free operation as compared with other compact systems with plastic enclosures. The aluminum baffle features an integrated high-frequency waveguide and very low diffraction to ensure the full high-frequency clarity of the beryllium dome hf driver is not compromised. The Aluminum enclosure’s ability to withstand corrosion and moisture damage also makes the e-352 well suited for use outdoors and in harsh weather applications. Features

• Redefines the SPL-to-size ratio: Architect approved design with esoteric audiophile quality and professional audio output
• Very low distortion and flat response provides ultra high-end audiophile quality with professional sound pressure output
• State-of-the-art Coax 3.5-inch high output transducer
• 25 mm Truextent® beryllium compression tweeter with exceptionally strong Neodymium magnetic circuit for  high-frequency bandwidth, and clarity
• Extremely small die-cast Aluminum Body — the e-351’s elegant enclosure is made from solid die-cast aluminum for maximum rigidity and corrosion resistance (available in black and white with optional full frontal metal grill)
• Passive system specially designed for use with VUEDrive V242 Systems Engine to provide a new level of sonic accuracy and output
• Flexible mounting with included bracket enables quick and intuitive speaker aiming, with M6 safety anchor.

The IPD-SM82-EZ-II is a Dante-enabled, digital endpoint solution in a surfacemount loudspeaker.  The IPD-SM82-EZ-II is a Dante-enabled, digital endpoint solution in a surfacemount loudspeaker. This speaker is Dante addressable and includes a 40W PoE amplifier for simplified installation and set up. Driver components consists of one 8.0 inch polypropylene woofer and one 1.0 inch soft dome tweeter in a ported enclosure. The included premountable surface-mount bracket allows for quick and easy installation and the powder-coated aluminum grille keep the speaker looking good in most environments. The system is UL 60065 approved. Included accessories: Safety cable, wall mount bracket, conduit connector and optional port plug.

The IPD-SM52-EZ-II is a Dante-enabled, digital end-point solution in a surface-mount loudspeaker.  The IPD-SM52-EZ-II is a Dante-enabled, digital end-point solution in a surface-mount loudspeaker. This speaker is Dante addressable and includes a 40W PoE amplifier for simplified installation and set up. Driver components consists of one 5.25 inch polypropylene woofer and one 1.0 inch soft dome tweeter in a ported enclosure. The included pre-mountable surface-mount bracket allows for quick and easy installation and the powder-coated aluminum grille keep the speaker looking good in most environments. The system is UL 60065 approved. Included accessories: Safety cable, wall mount bracket, conduit connector and optional port plug.