Dante-enabled Product Catalog

The new DFT-EXP Series from Joseph Electronics provides for up to 12 channels of 3G/HD/SDI uncompressed video plus four additional slots that can be fitted with a selection of other convenient modules for audio, intercom, analog video, genlock, MADI, and even Dante. The new DFT-EXP Series from Joseph Electronics provides for up to 12 channels of 3G/HD/SDI uncompressed video plus four additional slots that can be fitted with a selection of other convenient modules for audio, intercom, analog video, genlock, MADI, and even Dante. All of this is transported on one or two singlemode fibers, depends on configuration) and can also be operated with optical redundancy. Housed in a compact 2RU fanless enclosure with redundant power supplies, up to six 2-channel modules can be fitted for signal counts from 4 to 12. The direction of each module is field reconfigurable to take care of last minute changes. Transmit modules are red and receive modules are blue. Red modules (TX) are slot specific and clearly labeled. The Blue modules (RX) are generic and can go in any slot.  Modules support SMPTE 424, SMPTE 292, SMPTE 259, SMPTE 297 and DVB-ASI. Units can be set to operate in non-reclocking mode to handle any signal from 50Mb/s all the way up to 3Gb/s. A Joseph’s exclusive is the Bypass Channel. This innovative feature allows you to add even more analog video, audio, or data. Units can be custom configured with fiber connections on the front or rear of the enclosure. Convenient LEDs indicate signal presence and link status. Trust Joseph Electronics to come up with innovative and flexible fiber transport solutions that allow broadcasters to make those last-minute changes quickly and easily.  The DFT-EXP Series is ideal for anyone who needs to move a variety of signals point-to-point. Whether for announce booths, host positions, stand-up postions, edit suites, or cross-campus feeds, The DFT-EXP takes advantage of the high bandwidth and dramatic weight savings offered by fiber to deliver pristine video signals and clean audio over far greater distances than copper. Custom features such as the placement of the fiber connectors; audio, Ethernet, data, and  intercom connectivity; optical redundancy; 1 or 2 fiber operation; and the By-Pass channel, ensure that customers get exactly wht they need and don’t pay for features that they don’t want or will never use. The ability to reconfigure your signals on-the-go has gone from being a convenience to being an absolute necessity. With nothing more than a basic screwdriver, users can quickly and easily make the last minute changes the inevitably come up on remote broadcasts.

The digital mixer DMIX-20 from IMG STAGELINE impresses with an intuitive operating concept of the well thought-out user surface. Due to the compact design, it is also ideally suited for applications on smaller to medium-sized stages and for your home studio. The device quickly impresses musicians, DJs and live sound mixers. Combined with an external router, the DMIX-20 also allows for a convenient operation via iPad using a network connection. The free app can be downloaded from the app store.  The digital mixer DMIX-20 from IMG STAGELINE impresses with an intuitive operating concept of the well thought-out user surface. Due to the compact design, it is also ideally suited for applications on smaller to medium-sized stages and for your home studio. The device quickly impresses musicians, DJs and live sound mixers. Combined with an external router, the DMIX-20 also allows for a convenient operation via iPad using a network connection. The free app can be downloaded from the app store.  The console features 16 mono input channels and 2 stereo input channels as well as +48 V phantom power. The 8 outputs can either be used as 4 aux paths and 4 subgroup paths or as 8 aux outputs. It features 2 integrated DSPs with 12 types of effects, noise gate, compressor/limiter, delay and a 100 mm motor-driven fader.  The colour LCD touch screen facilitates the operation with your settings which can easily be stored due to modern digital technology. Thus, you are able to store the channel's sound adjustments via parametric equalizer, graphical equalizer or use the options of the effect unit. Furthermore, the digital mixer enables you to store 24 overall settings. The full duplex port allows for simultaneous recording and reproduction from a PC or onto a PC. It is also equipped with two headphone outputs. 19-channel digital audio mixer,   with touch screen, 2 DSP effect units and USB interface.

• 16 mono input channels with assignable controls
• 8 channel inserts
• 2 stereo input channels
• 1 USB audio channel
• Full duplex USB port (allows for simultaneous recording and reproduction) for reproduction of digital audio data from a PC and for recording mixings onto a PC
• 2 integrated DSPs as well as equalizer, noise gate, compressor/limiter and delay
• 4 aux send outputs and 4 subgroup outputs or 8 aux outputs
• 6 DCA groups
• Control room function
• 100 mm motor-driven fader
• Sampling rate: 24 bits/48 kHz
• Settings can be stored
• Remote control possible via iPad (connected to external router)
• RJ45 network connection for remote control
• +48 V phantom power
• 2 adjustable headphone outputs
• 17.8 cm colour LCD touch screen (7") for display and settings
• Mounting facility for extension modules available at option
• 482 mm (19") rack installation, 8 RS, with supplied mounting brackets

VUE’s al-Class line array systems have achieved international recognition for raising the bar for sonic definition, through a combination of advanced transducer design and tightly integrated dsp. VUE’s al-Class line array systems have achieved international recognition for raising the bar for sonic definition, through a combination of advanced transducer design and tightly integrated dsp, The al-12 expands the scale of applications that can benefit from the al-Class’ pristine sonic performance.

• Three-way, High output true Line Array design
• Extremely Scalable System – combine al-12 acoustic elements with al-8s and al-4s to build complex hybrid arrays
• New State-of the Art transducer technology
• Integrated VUEDrive™ System Engines with SystemVUE control Dante Audio networking
• Fully compatible with VUEPoint beam steering technology
• Integrated flying hardware allows quick assembly of arrays
• Optional subwoofers, flying and transport accessories

The al-4 Subcompact Line Array System utilizes advanced technologies to extract unparalleled performance from an ultra-compact package. With cutting-edge technologies such as a beryllium compression driver, Kevlar/Neo low-frequency transducers, and precision amplification and DSP, the highly scalable al-4 system brings pristine sound quality and coveted line array performance to more applications than ever before. The al-4 Subcompact Line Array System utilizes advanced technologies to extract unparalleled performance from an ultra-compact package. With cutting-edge technologies such as a beryllium compression driver, Kevlar/Neo low-frequency transducers, and precision amplification and DSP, the highly scalable al-4 system brings pristine sound quality and coveted line array performance to more applications than ever before. The al-4 Line Array System is made up of the al-4 Acoustic Element, the rack mount V4 Systems Engine, which provides amplification, DSP and system control and the al-4SB flyable isobaric subwoofer, along with a wide range of accessories for suspension and transportation.  The system is sold in standard blocks, with eight al-4 Acoustic Elements and a single V4 Systems Engine (designed to power up to 8 al-4 elements) comprising a single block. Multiple al-4 blocks can be combined to address larger applications.

• al-4 elements make it easy to scale al-4 arrays for various applications (one rental inventory serves a variety of applications)
• Fly up to 24 al-4 elements from a single fly-bar
• 90° horizontal coverage angle per al-4 element
• VUE designed 1-inch exit neodymium compression driver with Truextent® beryllium diaphragm
• VUE designed dual 4-inch Kevlar cone neodymium woofers
• Compact and lightweight – each al-4 element is 18.9 x 5.5 x 10.3 inches (W x H x D) and 18 lbs.

The al-8 Compact Line Array System picks up where the al-4 Subcompact Line Array System leaves off. The al-8 ‘Acoustic Linearity’ element utilizes VUE’s most advanced technologies including CST™ to extract exceptional performance from its compact design. The al-8 Compact Line Array System picks up where the al-4 Subcompact Line Array System leaves off. The al-8 ‘Acoustic Linearity’ element utilizes VUE’s most advanced technologies including CST™ to extract exceptional performance from its compact design. Innovative technologies such as beryllium compression drivers, Kevlar/Neo transducers, precision amplification and DSP, onboard SystemVUE networking, and full compatibility with the VUEPoint beam steering technology allow the highly scalable al-8 to deliver unprecedented performance to a broad range of mid-sized sound reinforcement applications.

• Each al-8 Compact Line Array System “block” includes
• four al-8 Acoustic Elements and a single VUEDrive™ V6 Systems Engine
• Scalable by adding or removing other al-Class acoustic elements, or combining multiple al-8 blocks
• Onboard SystemVUE™ networking
• Fully compatible with VUEPoint beam steering technology
• Integrated flying hardware allows quick assembly of arrays of up to 16 elements per flybar
• Optional flying and transport accessories available

The DMIX-20DANTE is a module for integrating the digital mixer DMIX-20 into a Dante™ network. The DMIX-20DANTE is a module for integrating the digital mixer DMIX-20 into a Dante™ network.

• Audio transmission via Ethernet using Dante™ protocol
• Parallel transmission of 32 x 32 audio channels
• Easy signal control via network technology
• Mini PCI motherboard
• Connections: 2 x RJ45

The ANI22 is a Dante audio network interface for installation into WaveDynamics™ series amplifiers with two input and two output channels. The installation of this optional module allows these amplifiers to be integrated into any DANTE enabled AV network and flawlessly transfer digital audio with any compatible product on the market, ranging from music sources to microphone systems, mixers, and many more. The ANI22 is a Dante audio network interface for installation into WaveDynamics™ series amplifiers with two input and two output channels. The installation of this optional module allows these amplifiers to be integrated into any DANTE enabled AV network and flawlessly transfer digital audio with any compatible product on the market, ranging from music sources to microphone systems, mixers, and many more. A truly plug & play solution is created when linking multiple DANTE enabled devices with each other on a standard IP network. It offers self-configuring solutions by using DHCP configuration, while the connection can be made to any existinig 100 Mbit or 1 Gbit network infrastructure. Bidirectional signal transmissions with a sampling frequency of 48 kHz and a bit depth of 24 bit are simultaneously possible on all in and output channels, while guaranteeing an incredibly low latency with a maximum up to 5 ms. (depending of the network occupation). The network connection is implemented using an Ethercon compatible connector, allowing connection using standard RJ45 or ruggedized Ethercon connections in professional applications. The ANI22 comes delivered as an installation kit, including all required accessories for integration into the amplifier chassis. Installation shall be performed by qualified technicians.

The ANI44 is a Dante audio network interface for installation into WaveDynamics™ series amplifiers with four input and four output channels. The installation of this optional module allows these amplifiers to be integrated into any DANTE enabled AV network and flawlessly transfer digital audio with any compatible product on the market, ranging from music sources to microphone systems, mixers, and many more. The ANI44 is a Dante audio network interface for installation into WaveDynamics™ series amplifiers with four input and four output channels. The installation of this optional module allows these amplifiers to be integrated into any DANTE enabled AV network and flawlessly transfer digital audio with any compatible product on the market, ranging from music sources to microphone systems, mixers, and many more. A truly plug & play solution is created when linking multiple DANTE enabled devices with each other on a standard IP network. It offers self-configuring solutions by using DHCP configuration, while the connection can be made to any existinig 100 Mbit or 1 Gbit network infrastructure. Bidirectional signal transmissions with a sampling frequency of 48 kHz and a bit depth of 24 bit are simultaneously possible on all in and output channels, while guaranteeing an incredibly low latency with a maximum up to 5 ms. (depending of the network occupation). The network connection is implemented using an Ethercon compatible connector, allowing connection using standard RJ45 or ruggedized Ethercon connections in professional applications. The ANI44 comes delivered as an installation kit,including all required accessories for integration into the amplifier chassis. Installation shall be performed by qualified technicians.

Formerly known as BLU-DAN The Soundweb London BLU-DA offers Dante / AES67 audio and BLU link. This means that the BLU-DA can be used to ‘add’ Dante / AES67 to one or more BLU link devices. The BLU-DA is configurable through HiQnet™ Audio Architect. A rich palette of logic objects and a “drag and drop” method of configuration provide a simple and familiar design environment. The BLU-DA features Dante and AES67 audio. The BLU-DA is capable of simultaneously transmitting and receiving up to 64 channels of Dante or AES67 audio (64 x 64). Primary and Secondary ports are provided for fault tolerance. Control is through a separate Ethernet port. The BLU-DA also includes support for Dante 4.0 firmware for integration with Dante Domain Manager. The BLU-DA also features BLU link, a low latency, fault tolerant digital audio bus of 256 channels, which uses standard Category 5e cabling giving a distance of 100m between compatible devices. Fiber media converters can be used to increase the distance between devices to over 40km. A bi-directional locate function allows devices to be identified both from and within HiQnet Audio Architect. The BLU-DA and the other members of the Soundweb London family provide the building blocks of the perfectly tailored system solution. Features

• AES67 and Dante Audio with support for Dante Domain Manager
• BLU link – 256 Channel, Low Latency, Fault Tolerant Digital Audio Bus
• ‘Add’ Dante / AES67 to one or more BLU link devices
• Clear Front Panel LED Indication
• Bi-Directional Locate Functionality
• HiQnet Device
• Configuration, Control and Monitoring from HiQnet Audio Architect
• 1RU, Half-Rack Device
• AES67 Compatible

“KE” intelligent power amplifier built-in DSP has: high performance double 32 - bit audio dedicated floating-point processor, high performance A/D, D/A converter. Each channel has 5 parametric equalizer, all parameters are adjustable. Multiple frequency division mode, 6db, 12db, 24db or 48db NingKe. Each channel has its own latency, maximum 80 ms and 0.1 ms delay interval. Each channel is independent of compressors and limiter, all parameters can be adjusted; Function of phase inversion, digital noise gate. 30 users memory, intuitive and friendly computer interface, RJ45 interface, Channel Copy function, convenient debugging KE2400A is a two channel intelligent power amplifier. “KE” intelligent power amplifier built-in DSP has: high performance double 32 - bit audio dedicated floating-point processor, high performance A/D, D/A converter. Each channel has 5 parametric equalizer, all parameters are adjustable. Multiple frequency division mode, 6db, 12db, 24db or 48db NingKe. Each channel has its own latency, maximum 80 ms and 0.1 ms delay interval. Each channel is independent of compressors and limiter, all parameters can be adjusted; Function of phase inversion, digital noise gate. 30 users memory, intuitive and friendly computer interface, RJ45 interface, Channel Copy function, convenient debugging “KE” intelligent power supply of the power amplifier is quasi resonant full bridge topology, the high-frequency transformer magnetic core uses cutting-edge technology of nanomaterials. It has the advantage of high power efficiency, big power, and uses high power tube as the full bridge switch tube, the power of the internal resistance is low. It can provide high current moment to load.

“KE” intelligent power amplifier built-in DSP has: high performance double 32 - bit audio dedicated floating-point processor, high performance A/D, D/A converter. Each channel has 5 parametric equalizer, all parameters are adjustable. Multiple frequency division mode, 6db, 12db, 24db or 48db NingKe. Each channel has its own latency, maximum 80 ms and 0.1 ms delay interval. Each channel is independent of compressors and limiter, all parameters can be adjusted; Function of phase inversion, digital noise gate. 30 users memory, intuitive and friendly computer interface, RJ45 interface, Channel Copy function, convenient debugging KE1600A is a two channel intelligent power amplifier. “KE” intelligent power amplifier built-in DSP has: high performance double 32 - bit audio dedicated floating-point processor, high performance A/D, D/A converter. Each channel has 5 parametric equalizer, all parameters are adjustable. Multiple frequency division mode, 6db, 12db, 24db or 48db NingKe. Each channel has its own latency, maximum 80 ms and 0.1 ms delay interval. Each channel is independent of compressors and limiter, all parameters can be adjusted; Function of phase inversion, digital noise gate. 30 users memory, intuitive and friendly computer interface, RJ45 interface, Channel Copy function, convenient debugging “KE” intelligent power supply of the power amplifier is quasi resonant full bridge topology, the high-frequency transformer magnetic core uses cutting-edge technology of nanomaterials. It has the advantage of high power efficiency, big power, and uses high power tube as the full bridge switch tube, the power of the internal resistance is low. It can provide high current moment to load.

KE1600B is a four channel intelligent power amplifier. “KE” intelligent power amplifier built-in DSP has: high performance double 32 - bit audio dedicated floating-point processor, high performance A/D, D/A converter. Each channel has 5 parametric equalizer, all parameters are adjustable. Multiple frequency division mode, 6db, 12db, 24db or 48db NingKe. Each channel has its own latency, maximum 80 ms and 0.1 ms delay interval. Each channel is independent of compressors and limiter, all parameters can be adjusted; Function of phase inversion, digital noise gate. 30 users memory, intuitive and friendly computer interface, RJ45 interface, Channel Copy function, convenient debugging “KE” intelligent power supply of the power amplifier is quasi resonant full bridge topology, the high-frequency transformer magnetic core uses cutting-edge technology of nanomaterials. It has the advantage of high power efficiency, big power, and uses high power tube as the full bridge switch tube, the power of the internal resistance is low. It can provide high current moment to load.

The DQ20 is a lightweight four-channel Class D switchmode power amplifier designed to give the best performance from any loudspeaker within the EM Acoustics product range, with advanced DSP and networking capabilities. The DQ20 is a lightweight four-channel Class D switchmode power amplifier designed to give the best performance from any loudspeaker within the EM Acoustics product range, with advanced DSP and networking capabilities. The DQ Series of advanced system amplifiers offer a unique combination of high power and audio performance, seamlessly combined with advanced DSP and network control. The DQ Series offers three models, physically identical four-in-four-out amplifiers but with varying total output power configurations of 6,000 watts (DQ6), 10,000 watts (DQ10) and 20,000 watts (DQ20) to suit specific loudspeaker models and audio applications. The DQ Series models offer extremely high power density in a convenient and flexible format for both mobile applications and installations alike. DQ Series advanced system amplifiers represent the most convenient solution for powering EM Acoustics loudspeakers. Each amplifier comes loaded with preset files for all EM Acoustics loudspeaker products, as well as a dedicated “linear” preset with no processor settings applied – but also blank memory slots for users to create their own setting files. Aside from the non-adjustable factory settings for each EM Acoustics product (such as high/low pass filters and limiters), users have access to a variety of processing functions including gain, delay and EQ (parametric and shelf) for system or venue optimisation. Power delivery is an area where the DQ Series amplifiers are unmatched. A state-of-the-art switchmode power supply with power factor correction ensures that a DQ Series amplifier can deliver all that is required, even under demanding load conditions and across a wide range of mains supply voltages. Each channel of a DQ Series amplifier can be configured to deliver its maximum power into either 2 or 4 Ohm nominal loads via the Podware software platform, therefore offering designers and integrators unparalleled flexibility and effectiveness. The use of state-of-the-art components and a finely optimised design results in generous power reserves, thus ensuring that pristine sound quality is maintained even under the most extreme conditions, anywhere in the world. DQ Series amplifiers have both analog inputs (4) and AES3 digital inputs (2) as standard, and can be supplied with an optional Dante™ networked audio input card if required (model numbers DQ6D, DQ10D and DQ20D). Maximum flexibility was the thought process with regards the input infrastructure – the four-in-four-out configuration allows the DQ Series models to be highly flexible whether deployed powering front of house systems, monitor mixes, or fill loudspeakers. The front panel provides an intuitive set of controls and indicators to enable simple and straightforward operation of all functions without the need for the remote software. A set of five LED indicators show signal level from -12dB to +12dB as well as clip, for each of the four DSP inputs. Six illuminated push buttons and two velocity-sensitive rotary encoders provide the user interface for all the input and output functions, as well as the utility features, in conjunction with the clear back-lit graphical display. A further set of LED indicators show output level and limiter status, and individual backlit channel mute buttons are also provided. A final set of four LED indicators give further status information for the amplifier including AES3 or Dante network audio input (if selected), device control network online status, and also the presence of a system EQ overlay if activated via the Podware remote control software platform.

“KE” intelligent power amplifier built-in DSP has: high performance double 32 - bit audio dedicated floating-point processor, high performance A/D, D/A converter. Each channel has 5 parametric equalizer, all parameters are adjustable. Multiple frequency division mode, 6db, 12db, 24db or 48db NingKe. Each channel has its own latency, maximum 80 ms and 0.1 ms delay interval. Each channel is independent of compressors and limiter, all parameters can be adjusted; Function of phase inversion, digital noise gate; 30 users memory, intuitive and friendly computer interface, RJ45 interface; Channel Copy function, convenient debugging KE1000B is a four channel intelligent power amplifier. “KE” intelligent power amplifier built-in DSP has: high performance double 32 - bit audio dedicated floating-point processor, high performance A/D, D/A converter. Each channel has 5 parametric equalizer, all parameters are adjustable. Multiple frequency division mode, 6db, 12db, 24db or 48db NingKe. Each channel has its own latency, maximum 80 ms and 0.1 ms delay interval. Each channel is independent of compressors and limiter, all parameters can be adjusted. Function of phase inversion, digital noise gate; 30 users memory, intuitive and friendly computer interface, RJ45 interface; Channel Copy function, convenient debugging “KE” intelligent power supply of the power amplifier is quasi resonant full bridge topology, the high-frequency transformer magnetic core using cutting-edge technology of nanomaterials. It has the advantage of high power efficiency, big power, and use high power tube as the full bridge switch tube, the power of the internal resistance is low

ECLER DN44BOB is a 4 inputs and 4 outputs DANTE Break Out Box. It allows to convert 4 analogue audio signals into 4 DANTE channels and send them into an Ethernet LAN, as well as to take 4 DANTE channels from the Ethernet network and convert them to analogue audio signals. ECLER DN44BOB is a 4 inputs and 4 outputs DANTE (Audio Over IP) Break Out Box. It allows to convert 4 analogue audio signals into 4 DANTE channels and send them into an Ethernet LAN, as well as to take 4 DANTE channels from the Ethernet network and convert them to analogue audio signals. It includes 4 GPI control ports (0-10VDC) and 4 GPO control ports (NO/NC relays). Can be used together with EclerNet DANTE ready products or as a stand-alone DANTE interface, together with any audio device including analogue audio inputs and/or outputs.

The RPio222 is an audio interface that enables you to flexibly configure and expand I/Os for the RIVAGE PM10 system as required by your application or the scale of your system. The RPio222 is an audio interface that enables you to flexibly configure and expand I/Os for the RIVAGE PM10 system as required by your application or the scale of your system.

• Two RY card slots that enable you to expand analog inputs and outputs, and/or digital inputs and outputs.
• Two HY card slots that are capable of transmitting/receiving up to 256 ins/outs of digital audio signals/control signals.
• HY card slot 1 features 256 ins/outs, and HY card slot 2 features 128 ins/outs.
• Two mini-YGDAI slots to support various audio formats.
• Dual redundant power supply built-in

Shure ULX-D Digital Wireless offers uncompromising 24-bit audio clarity and extremely efficient RF performance with single, dual, and quad channel receivers for any size professional sound reinforcement application. Shure ULX-D Digital Wireless offers uncompromising 24-bit audio clarity and extremely efficient RF performance with single, dual, and quad channel receivers for any size professional sound reinforcement application. Scalable, intelligent hardware delivers the best performing digital wireless available, with a wide selection of trusted Shure microphones to choose from. Unmatched spectrum efficiency enables a dramatic increase in the number of simultaneous active transmitters on one TV channel, with rock-solid signal over the entire range. Optional rechargeable accessories eliminate the need for disposable batteries while offering extended run time and metering accuracy in hours and minutes. For secure transmission, all ULX-D components include AES-256 encryption. Generations ahead of any other available system in its class, ULX-D brings a new level of performance to professional sound reinforcement. Features Uncompromising Digital Wireless Audio

• 24-bit/48 kHz digital audio that delivers incredibly clear and accurate reproduction of the source material
• 20 Hz–20 kHz frequency range with flat response
• Greater than 120 dB dynamic range through the analog outputs
• 130 dB dynamic range (typical) using Dante digital networked audio
• 60 dB of adjustable system gain easily accessible from the receiver front panel
• No transmitter gain adjustments needed: optimized for any input source

The GS-FW012 ip is the Dante network audio version of the GS-FW012 4W unit. The GS-FW012 ip is the Dante network audio version of the GS-FW012 4W unit.   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.  

The VITA is often used as a remote link to the main commentary unit. It is favoured by pitch side interviewers, and is popular in combinations with wireless systems. Inputs The main input is switchable between mic, mic+48v phantom power, and line. The input has an adjustable gain control to meet a wide range of dynamic, condenser, and ribbon microphones. The input passes through a broadcast quality dsp controlled compressor/limiter circuit prior to the A/D converters to maximize headroom and designed to manage the peaks of excited commentary. The A/D converters are full broadcast spec low noise, 48K 24bit designed for on air use. Outputs/Talkback There are two output circuits. One is used as the main programme audio, the other as a talkback circuit. There are separate button controls for each of these using Config+. They can each can be independently configured for latching or momentary, with the programme button also having a 'cough' feature. All talkback I/O connections are via the RJ45/CAT5 Dante network audio connection. Monitoring There are two monitoring inputs each with independent level control. Each of these inputs can be switched from the left ear, right ear, or both ears to the stereo headphone amplifier. The headphone amplifier is our high performance intelligent circuit which automatically adjusts its output level for high or low impedance headphones and can also drive mono earpieces and headphones. There are 2 off 4 LED level meters both indicating -12 to +6 dB. One LED meter is located on the front panel to enable easy visualization of the audio levels if the VITA is being used as a table top device and the other meter is located on the top panel facing the operator when used by its beltclip. Power Via PoE on the network connection, or via an external DC power supply. The VITA is often used as a remote link to the main commentary unit. It is favoured by pitch side interviewers, and is popular in combinations with wireless systems.

The SMQ series are quad channel power amplifiers providing a new standard for affordable audio amplification in both fixed and mobile audio installations. The SMQ series are quad channel power amplifiers providing a new standard for affordable audio amplification in both fixed and mobile audio installations. Their advanced features and availability in different output powers for 350 Watt, 500 Watt and 750 Watt provide an enormous flexibility for numerous applications. The WaveDynamics™ DSP processor in combination with the 2.5” LCD display gives an unmatched user experience with intuitive functions overview for easy configuration. Acoustics can be optimized using Butterworth, Bessel and Linkwitz-Riley filters with selectable rolloff which can be configured as Low / High & Band Pass and the 7-Band equalizer which has adjustable frequencies and Q-factor. Other provided functions are delay and dynamic bass boost. These settings can be custom configured using the front panel of the amplifier, whereby access can be given on two different levels (User & Administrator) using password or USB-key protection. Loudspeaker protection is provided by an output power limiter whereby the maximum output power for every channel (in Watt) can be configured. Configuration is made simple with loudspeaker presets and full system configurations which can be selected from a library and uploaded with an USB flash drive. This ensures the best acoustical performance with a bullet-proof loudspeaker protection. A great input flexibility and source compatibility is provided by the input selection matrix in combination with the XLR and terminal block signal connections. The outputs are made using Speaker and Terminal block connections. System integration is made easy using the RS-232 control port. An optional Dante™ interface is available for the WaveDynamics™ amplifiers. The installation of this network interface allows receiving and sending of low latency, high quality audio over a standard Ethernet network. 

RedNet PCIeR combines extremely low latency with Ethernet redundancy, offering the best possible system performance coupled with exceptional reliability. RedNet PCIeR combines extremely low latency with Ethernet redundancy, offering the best possible system performance coupled with exceptional reliability. The card requires a standard PCIe slot in a Windows or Mac computer and delivers up to 128 channels I/O with under 3ms latency*. Dual RJ45 Gigabit Ethernet connectors switch automatically according to network availability and link the audio computer to the rest of the network. Key Features and Performance

• Dual Ethernet ports for full redundancy
• 128 channels in and 128 channels out at 96kHz and below
• 64 channels in and 64 channels out at 176.4kHz and 192kHz
• Compatible with any Dante network
• Up to 192 kHz sample rate
• Under 3ms analogue-to-analogue round-trip latency through DAW with Focusrite RedNet A-D/D-A*
• ASIO and Core Audio drivers
• Thunderbolt PCIe chassis compatible

*Dependent on ASIO/Core Audio buffer size

Where to Buy: Focusrite RedNet PCIeR card

RedNet A8R expands Focusrite’s growing range of 1U Dante interfaces with network and power supply redundancy, providing eight channels of analogue input and output.  RedNet A8R expands Focusrite’s growing range of 1U Dante interfaces with network and power supply redundancy, providing eight channels of analogue input and output. RedNet A8R offers high quality 192kHz/24-bit A-D and D-A digital conversion with up to 119dB dynamic range, plus two channels of AES/EBU digital I/O, to and from the Dante network. RedNet A8R provides 24-bit conversion at 44.1, 48, 88.2, 96, 176.4 and 192kHz sample rates (remotely-selectable) and includes pull-up/down capability. Electronically-balanced analogue audio I/O is provided on DB25 connectors wired to AES59 specifications. Internal signals are balanced throughout. Inputs and outputs are remotely configurable for 0dBFS= +18dBu or +24dBu signal levels. In addition, two channels of AES/EBU I/O are provided on XLR connectors, with sample rate conversion on the input. The clock source for the unit may be internal, or external via Word Clock or DARS (via the XLR input). RedNet A8R also outputs Word Clock. RedNet A8R features power and network redundancy, with two separate universal (100–240V) PSUs, which include IEC connectors with retaining clips, plus dual network interfaces on locking etherCON connectors. These ports can be configured remotely to Redundant or Switched (daisy-chain) mode. Key Features

• Eight channels of high-quality line-level analogue I/O
• 24-bit precision Focusrite conversion up to 192kHz
• 119dB dynamic range, flat from 20Hz – 20kHz, +/-0.15dB
• Internally balanced throughout
• Dante-based remote-controlled audio-over-IP connectivity
• Dual power supplies and network connections with full power and network redundancy, including latching connectors for maximum reliability
• Word Clock I/O and DARS for increased flexibility, plus an additional two channels of AES/EBU I/O
• Analogue I/O to AES59 standard on DB25 connectors
• Comprehensive front-panel indicators
• Software-based routing and control – no hardware patching required

The Express ip was designed to provide a quick and easy solution when it is necessary to provide interfacing for two commentators, with simple facilities, into a Dante network audio network. The Express ip was designed to provide a quick and easy solution when it is necessary to provide interfacing for two commentators, with simple facilities, into a Dante network audio network. This high quality commentary unit is ideal for those looking for cost effective solutions without paying for unwanted features; the Express Box is worthy of investigation. Inputs Two front panel mic inputs with selectable 48v phantom power. Outputs/Talkback Each mic input has its own individual output. The second mic output can be switched on the rear panel to be a mix of both mic inputs.  The two talkback circuits have individual outputs and are common for both commentators. Monitoring There are four inputs for external sources, and one sidetone pot of their own voice. These are available independently to each commentator on individual pots, so each can adjust the inputs for their own preference of mix level. There are two 6.35mm headphone sockets – 1 for each commentator. A 7 segment LED PPM meter displays level.  Network The network connection is AES67/Dante compatible and is available on the rear panel on an RJ45/CAT5 connection.  It offers 4 input channels and 4 output channels. Power There is an internal switch mode AC power supply, or the Express ip can be powered by PoE if powered by a switch that provides power over Ethernet.

Brio is the most powerful and compact digital broadcast audio console in its class. It has a comprehensive broadcast feature set to support a wider breadth of broadcasters, and the same market-leading levels of quality and customer support which Calrec is renowned for. Brio is the most powerful and compact digital broadcast audio console in its class. It has a comprehensive broadcast feature set to support a wider breadth of broadcasters, and the same market-leading levels of quality and customer support which Calrec is renowned for. The smallest in Calrec’s Bluefin2 family, Brio’s control surface is unlike any other console. At only 892mm wide with a 36 dual-layer fader surface, Brio is entirely self-contained with analogue and digital I/O, and GPIO, built into the surface. Additional expansion I/O slots are also fitted to allow for further I/O to be connected. A Hydra2 module can be fitted to connect to and share audio over Calrec’s Hydra2 network. This means Hydra2’s sophisticated management facilities can be utilised for network-wide control, including interfacing with multiple Video and Audio over IP networks such as SMPTE2022, Dante, AES67, Ravenna and Soundgrid. This protects the system against any future formats that may emerge and allows the console to sit on multiple networks simultaneously if desired. With lots of delay resource, dynamics, integrated talkback, and multiple monitor outputs, Brio is broadcast-ready with no compromise or workarounds.

The latest addition to Soundcraft's Vi range of digital live consoles, the new Vi2000 brings unparalleled mixing power to the live touring, install and corporate AV markets in a single-box, cost-effective package. The latest addition to Soundcraft's Vi range of digital live consoles, the new Vi2000 brings unparalleled mixing power to the live touring, install and corporate AV markets in a single-box, cost-effective package. The new console combines the unique Vistonics-based control surface concepts of the highly successful Vi3000, 5000 and 7000 consoles, with Soundcraft® SpiderCore - a powerful integrated DSP and IO engine based on STUDER technology and compact enough to reside inside the control surface itself. Running the latest V6.1 console software, the Vi2000 benefits from the acclaimed BSS DPR901ii Dynamic EQ via an insertable processing pool which also includes Lexicon Multi-effects and the ability to insert any external device. In addition to a full complement of on-board mic and line IO, configurable up to 48 mic line inputs and 16 line outputs using combinations of 16ch XLR modules in 4 rear-mounted slots, the console includes two 64ch expansion slots allowing up to two MADI-based Stageboxes to be connected, or alternatively the slots provide access to the extensive range of D21m IO option cards, addressing all industry standard audio formats. The total I/O count of the console is 246 in 246 out. In common with other Vi000-series consoles, a built-in 64x64ch Dante interface provides direct recording/playback connection via Ethernet to any PC or Mac-based recording software, or allows the Vi2000 to integrate into an existing Dante network. The Dante interface is complimented by an additional optical MADI interface designed for either record feeds or connection of the Soundcraft Realtime Rack Plug-in engine, adding the power of Universal Audio UAD-powered plug-ins to the Vi2000's pristine audio quality. The package is completed by an ultra-compact frame design only 1.15m in width, with 16 input faders and 8 output faders, allowing the console to fit into both space and budget restricted applications. Patented features like Faderglow, VM2 microphone monitoring and Vistonics provide a uniquely user-friendly mixing experience and superb sound quality is assured by a 40-bit floating-point DSP environment, running STUDER, BSS, Lexicon and dbx algorithms.

CDD-LIVE 8 is an ultra-compact, self-powered, two-way system suitable for small-medium scale sound reinforcement, AV presentations and HoW to distributed systems, frontfill/infill and use as a micro stage monitor. It incorporates a two-channel Class D amplifier, DSP and Dante™ digital audio networking — simplifying set-up, enhancing control and eliminating amplifier racks. CDD-LIVE 8 is an ultra-compact, self-powered, two-way system featuring an 8” (200mm) LF/1” (25mm) exit HF Coaxial Differential Dispersion driver which can be easily rotated for horizontal or vertical orientation. Its small size gives no indication of its high output capability and exceptional fidelity suitable for small-medium scale sound reinforcement, AV presentations and HoW to distributed systems, frontfill/infill and use as a micro stage monitor.  With wide coverage close-up, the Coaxial Differential Dispersion technology employed in the CDD- LIVE 8 delivers more consistent audience coverage than systems with fixed X° x Y° coverage patterns, while also achieving ‘point source’ summation of the LF and HF sections — eliminating off-axis variations in frequency response associated with non-coaxial designs.  A self-powered system, the CDD-LIVE 8 incorporates a two-channel Class D amplifier, DSP and Dante™ digital audio networking — simplifying set-up, enhancing control and eliminating amplifier racks.

CDD-LIVE 12 is a high-performance, self-powered system designed for applications that require high output levels and exceptional fidelity from a very compact enclosure. It incorporates a two-channel Class D amplifier, DSP and Dante™ digital audio networking — simplifying set-up, enhancing control and eliminating amplifier racks. CDD-LIVE 12 is a high-performance, self-powered system designed for applications that require high output levels and exceptional fidelity from a very compact enclosure. With a peak output capability of 128dB at 1 metre, it is ideal for a multitude of stand-alone and distributed sound reinforcement requirements — from touring, theatre and portable live sound, to concert hall and HoW installations, AV events and stage monitor use. Featuring a high-specification 12” (300mm) LF/1” (25mm) exit HF Coaxial Differential Dispersion driver, it delivers more consistent audience coverage than a conventional system with a fixed X° x Y° coverage pattern and has wide 110° horizontal coverage close-up. It also achieves ‘point source’ summation of the LF and HF sections — eliminating off-axis variations in frequency response associated with non-coaxial designs.  CDD-LIVE 12 incorporates a two-channel Class D amplifier, DSP and Dante™ digital audio networking — simplifying set-up, enhancing control and eliminating amplifier racks.