Dante-enabled Product Catalog

PRODIGY.MC is a modular audio converter that supports multiple formats, offering flexible I/O, networked audio, and extremely powerful hardware and software – in only 2RU of space.

PRODIGY.MC has been designed to address numerous applications in pro audio, broadcast, installation and studio applications, using a single hardware frame.

PRODIGY.MC supports the latest audio network technologies, including Dante, RAVENNA and SoundGrid. Its various remote options allow access to hardware settings, including a channel-based routing matrix. Since its design is fully modular, the hardware can be configured according to the real requirements of any application – results in budget saving as well as guaranteeing the flexibility to address changes in the global media environment through future upgrades.

Modular approach, maximum flexibility 

PRODIGY.MC provides a variety of I/O capabilities. The concept is based on its eight converter slots for analog line-level, microphone input and AES3 option modules. Equipped with eight channels per module and interface direction, PRODIGY.MC provides up to 64 inputs and outputs.

In addition, PRODIDGY.MC’s hardware supports MADI and a network audio option. The two MADI slots can be equipped with BNC, SC optical and SFP modules.

A network audio option board adds Dante, RAVENNA (AES67) or SoundGrid connectivity to this very powerful device, and increases its maximum channel capacity to 320 inputs and 324 outputs.

Control is key, multiple access locally and remotely 

The combination of unified remote control via globcon, a browser-based GUI and a 5-inch, user-friendly backlight IPS touch display on the front panel ensures ultimate usability.

As globcon software addresses the breadth of pro audio applications, PRODIDGY.MC offers consistent and coherent control of multiple pieces of equipment. It is available for Mac-, PC- and Linux-based systems.

In addition, PRODIGY.MC can be accessed via its integrated web server with remote access via HTML and JavaScript-based user interfaces.

The browser-based GUI allows remote access independent from the operating system. While the management network port is completely independent from the audio network connection, it can be patched into a built-in switch for in-band management of the device.

Third-party remote control protocols and plugins can be added for extended system integration.

Automated surveillance and red light control is also possible through GPIOs.

Networked audio and seamless interoperability 

PRODIGY.MC’s network audio module is designed to accommodate different network audio formats in order to increase flexibility and meet changing requirements in various applications.

DirectOut’s own RAVENNA stack is AES67 compliant, and supports SMPTE ST2110-30/31 and ST2022-7 for redundant streaming. Capable of processing up to 128 channels, it has proven itself to be a powerful, flexible and high-performance AoIP hardware solution in numerous sophisticated broadcast and studio environments.

With the 64 channel Dante option, the PRODIGY.MC hardware offers access to the most popular AoIP format on the market, and is able to act as front-end for Dante-enabled audio consoles, or as a standalone converter and routing system.

Waves Audio’s SoundGrid offers extremely low latency and 128 channel audio processing for live sound, studio and broadcast applications. Virtual soundcards for Mac OS® and Windows®, provided by Waves Audio, can directly interface with any DAW.

Safe operation, monitoring 

Two headphone outputs on the front panel of PRODIGY.MC provide monitoring of any I/O directly at the device.

The hardware provides maximum redundancy for a comprehensive professional operation, and is supported by two phase-redundant power supplies with separate IEC inlets and AC power switches on the rear panel. In addition, each single analog, AES3 and MADI module can be assigned to be redundant to a second. An internal switch on the network module also ensures redundancy for the network domain. While the Dante module supports Dante redundancy mode, the RAVENNA module offers stream redundancy in accordance with ST2022-7.

PRODIGY.MP is a multifunction audio processor that is designed to address numerous applications in pro audio, broadcast, installation and studio applications. PRODIGY.MP provides flexible I/O, dual networked audio, DSP (EQ, Delay, Limiter, Matrix/Mixing, Inserts…), Sample Rate Conversion, multiple sync generation/handling/distribution and powerful hard- and software.

Housed in a 2U rack frame, it supports networking selectable from Dante, RAVENNA and SoundGrid, while remote options allow access to hardware settings, DSP, SRC, network and a channel-based routing matrix. Fully modular, the hardware can be configured to specific requirements, giving budget savings as well as the flexibility to address changes in the global media environment through future upgrades.

Modular Approach, Maximum Flexibility

I/O includes four local slots for analog line-level/mic input and AES3 modules. With eight channels per module and interface direction, PRODIGY.MP provides up to 32 local inputs and outputs.

The hardware additionally supports two MADI and two network audio options. MADI slots can be equipped with BNC, SC optical and SFP modules, while network audio boards add Dante, RAVENNA/AES67 or SoundGrid, and increase channel capacity to 416 inputs/420 outputs.

DSP, SR & Sync management

With FPGA-based processing, user-definable processing paths and pick-off points, analog and digital inserts, outputs and summing matrices, PRODIGY.MP boasts very low latency DSP for EQ (FIR and IIR filters), Time Adjust/Delays and Limiters. The internal DSP reference can be set to Word Clock or Video input, any network/digital input, or internally generated clock. Sample Rate Converters on AES inputs and free-running MADI and network I/Os are available, allowing simultaneous Media/Format/Sample Rate and Sync conversion, DSP and full routing at up to 192 kHz.

Control Is Key, Multiple Access Locally & Remotely

globcon remote control, a browser-based GUI, and a backlight front panel IPS touch display ensure ultimate usability. Where globcon software offers integrated control of multiple pieces of equipment in Mac/PC/Linux-based systems, an integrated web server supports remote access via HTML and JavaScript-based user interfaces.

The browser-based GUI allows immediate remote access without the need to install software. The management network port is independent of the audio network, and can employ a built-in switch for in-band management of the device.

Third-party remote control protocols can extend system integration further. Automated surveillance and red light control is possible through GPIOs, while standard MIDI input and outputs give communication and active/passive control via MIDI. USB ports on the front and rear panels for connection and control of legacy devices.

Networked Audio & Seamless Interoperability

PRODIGY.MP’s network audio modules accommodate audio formats including DirectOut’s AES67-compliant RAVENNA stack that supports SMPTE ST2110-30/31 and ST2022-7 for redundant streaming. With the 64-channel Dante option, the hardware can be a front-end for audio consoles, or a standalone converter, DSP, SRC and routing system.

Waves Audio’s SoundGrid offers extremely low latency, and 128 channels of audio processing for live sound, studio and broadcast. Waves servers can be connected via SoundGrid for plug-in inserts. Virtual soundcards for macOS and Windows can directly interface with any DAW for multitrack recording and virtual soundchecking.

Secure Operation, Monitoring

Two front panel headphone outputs provide monitoring of any I/O at the device. Two phase-redundant power supplies with separate IEC inlets and AC power switches support hardware redundancy. An internal switch on the network module also ensures redundancy for the network domain. While the Dante module supports Dante redundancy, the RAVENNA module offers stream redundancy in accordance with ST2022-7. EARS (Enhanced Automatic Redundancy Switching) is available for several combinations of inputs.

HARVEY is a flexible audio and media control matrix and a key component for PA systems and conference systems. It is equipped with analog audio inputs and outputs as well as a large number of different control interfaces and the ideal matrix for conference rooms, home cinemas, museums, educational facilities, multipurpose rooms and theatres. In addition to audio signal processing, lighting and media control, the new generation allows for a flexible response to different market requirements with regard to the number of channels. The audio input and output channels can be scaled in steps of 4, allowing HARVEY to cover the spectrum from small projects with an 8×8 audio matrix to special applications with e. g. 4×24 channels. An appealing wall-mounted remote control for the most important functions completes the product portfolio.nts are composed in a project by drag & drop and configured immediately in the context menu. Complex projects are configured with only a few mouse-clicks.

DD4 devices are gateways used to convert DANTE (Digital Audio Network Through Ethernet - developed by AUDINATE) protocol. These devices allow the extraction of analog audio signals from the DANTETM network as well as the introduction of analog channels within the same network.

DD4 is a device designed for use in installation, for which the analog inputs and outputs are made through Euroblock 3.5mm connectors.

It has balanced inputs and outputs to provide greater immunity to noise that interferes with audio signal lines.

The DMP 128 FlexPlus series of compact Dante Digital Matrix Processors features twelve FlexInput channels with full DSP, including AEC, and Dante connectivity, in only a half rack space. With an extensive mix matrix, a USB audio interface, and analog compatibility, the DMP 128 FlexPlus provides the ability to easily connect and route multiple types of source signals to accommodate diverse analog and digital system requirements. Featuring an ACP bus for audio control panels, an expansion port for bidirectional digital audio connection with a DTP CrossPoint or another DMP, eight audio file players, and configurable macros, the DMP 128 FlexPlus can be used anywhere from a collaboration space, to a large conference room.

Features

• Two models with 60 total inputs and 32 total outputs — The DMP 128 FlexPlus offers signal routing and management for a wide range of sound system applications:
  • Inputs: 48 Digital via Dante or Expansion Bus, 4 Analog balanced/unbalanced mic/line level, 8 Aux for USB or VoIP
  • Outputs: 16 Digital via Dante or Expansion Bus, 8 Analog balanced/unbalanced line level, 8 Aux for USB, VoIP or File Players
• 48x24 Dante audio networking with Dante Domain Manager and AES67 support — Dante audio networking provides scalability for creating larger audio matrixes over a local area network using standard protocols. A built‑in two‑port Gigabit switch can be configured to support primary and redundant Dante audio networks.
• Twelve channels of AEC - acoustic echo cancellation — The DMP 128 FlexPlus includes twelve independent channels of high performance AEC, as well as selectable noise cancellation. Extron AEC features advanced algorithms that deliver fast echo canceler convergence for optimal intelligibility in situations that challenge AEC performance.
• All models include a USB Audio interface, providing up to four channels of digital audio sends and returns — Support for USB Audio simplifies implementation of systems with either soft-codec conferencing or computer-based audio and enables use of the same interface for both small and large scale systems.

The DMP 128 FlexPlus series of compact Dante Digital Matrix Processors features twelve FlexInput channels with full DSP, including AEC, and Dante connectivity, in only a half rack space. With an extensive mix matrix, a USB audio interface, and analog compatibility, the DMP 128 FlexPlus provides the ability to easily connect and route multiple types of source signals to accommodate diverse analog and digital system requirements. For collaboration applications, the V model includes up to eight VoIP lines. Featuring an ACP bus for audio control panels, an expansion port for bidirectional digital audio connection with a DTP CrossPoint or another DMP, eight audio file players, and configurable macros, the DMP 128 FlexPlus can be used anywhere from a collaboration space, to a large conference room.

Features

• Two models with 60 total inputs and 32 total outputs — The DMP 128 FlexPlus offers signal routing and management for a wide range of sound system applications:
  • Inputs: 48 Digital via Dante or Expansion Bus, 4 Analog balanced/unbalanced mic/line level, 8 Aux for USB or VoIP
  • Outputs: 16 Digital via Dante or Expansion Bus, 8 Analog balanced/unbalanced line level, 8 Aux for USB, VoIP or File Players
• 48x24 Dante audio networking with Dante Domain Manager and AES67 support — Dante audio networking provides scalability for creating larger audio matrixes over a local area network using standard protocols. A built‑in two‑port Gigabit switch can be configured to support primary and redundant Dante audio networks.
• Twelve channels of AEC - acoustic echo cancellation — The DMP 128 FlexPlus includes twelve independent channels of high performance AEC, as well as selectable noise cancellation. Extron AEC features advanced algorithms that deliver fast echo canceler convergence for optimal intelligibility in situations that challenge AEC performance.
• All models include a USB Audio interface, providing up to four channels of digital audio sends and returns — Support for USB Audio simplifies implementation of systems with either soft-codec conferencing or computer-based audio and enables use of the same interface for both small and large scale systems.
• V model includes up to eight VoIP lines that support generic Session Initiation Protocol - SIP 2.0 connectivity per RFC 3261— The DMP FlexPlus C V AT is compatible with select VoIP systems and can be used for a single VoIP line or as a VoIP farm for up to eight rooms:
  • VoIP configuration via onboard web page separates AV and IT workflows
  • Flexible local conference calling with Extron exclusive independent audio processing and control per call - level, mute, hold, and hang-up per caller
  • Onboard SIP logging eliminates the need for network capture software
  • Supports standard and wideband codecs, with selectable priority
  • Quality of Service – QoS support for Layer 2 and Layer 3
  • Transport Layer Security – TLS and Secure Real-Time Transport Protocol – SRTP ensures industry standard call security
  • NAT Traversal supported via STUN, TURN, and ICE simplifies integration across challenging network topologies
  • VLAN support via 802.11Q Tags

The Extron DMP 64 Plus series of compact Digital Matrix Processors features six DSP input channels with AEC, in only a half rack space. With a flexible mix matrix, a USB audio interface, and 6x4 analog connectivity, the DMP 64 Plus provides the ability to easily connect and route signals to accommodate system requirements. The AT models can be powered by PoE+ and provide 32x16 Dante connectivity for popular Dante microphone arrays, and other Dante devices, while the V models include up to eight VoIP lines for collaboration applications. Featuring an ACP bus for audio control panels, an expansion port for bidirectional digital audio connection with a DTP CrossPoint® or another DMP, eight audio file players, and configurable macros, the DMP 64 Plus can be used in a wide variety of smaller systems.

Features

• Four models - The DMP 64 Plus offers signal routing and management for a wide range of sound system applications:
  • Inputs: 32 Digital via Dante or Expansion Bus, 6 Analog balanced/unbalanced mic/line level, 8 Aux for USB or VoIP
  • Outputs: 16 Digital via Dante or Expansion Bus, 4 Analog balanced/unbalanced line level, 8 Aux for USB, VoIP or File Players
• 32x16 Dante audio networking with Dante Domain Manager and AES67 support — Dante audio networking provides scalability for creating larger audio matrixes over a local area network using standard protocols. A built in two port Gigabit switch can be configured to support primary and redundant Dante audio networks.
• Six channels of AEC - acoustic echo cancellation — The DMP 64 Plus includes six independent channels of high performance AEC, as well as selectable noise cancellation. Extron AEC features advanced algorithms that deliver fast echo canceler convergence for optimal intelligibility in situations that challenge AEC performance.
• Ability to be powered over PoE+ on AT models
• All models include a USB Audio interface, providing up to four channels of digital audio sends and returns — Support for USB Audio simplifies implementation of systems with either soft-codec conferencing or computer-based audio and enables use of the same interface for both small and large scale systems.
• V model includes up to eight VoIP lines that support generic Session Initiation Protocol - SIP 2.0 connectivity per RFC 3261— The DMP 64 Plus C V AT is compatible with select VoIP systems and can be used for a single VoIP line or as a VoIP farm for up to eight rooms:
  • VoIP configuration via onboard web page separates AV and IT workflows
  • Flexible local conference calling with Extron exclusive independent audio processing and control per call - level, mute, hold, and hang-up per caller
  • Onboard SIP logging eliminates the need for network capture software
  • Supports standard and wideband codecs, with selectable priority
  • Quality of Service – QoS support for Layer 2 and Layer 3
  • Transport Layer Security – TLS and Secure Real-Time Transport Protocol – SRTP ensures industry standard call security
  • NAT Traversal supported via STUN, TURN, and ICE simplifies integration across challenging network topologies
  • VLAN support via 802.11Q Tags

The Extron DMP 64 Plus series of compact Digital Matrix Processors features six DSP input channels with AEC, in only a half rack space. With a flexible mix matrix, a USB audio interface, and 6x4 analog connectivity, the DMP 64 Plus provides the ability to easily connect and route signals to accommodate system requirements. The AT models can be powered by PoE+ and provide 32x16 Dante connectivity for popular Dante microphone arrays, and other Dante devices, while the V models include up to eight VoIP lines for collaboration applications. Featuring an ACP bus for audio control panels, an expansion port for bidirectional digital audio connection with a DTP CrossPoint® or another DMP, eight audio file players, and configurable macros, the DMP 64 Plus can be used in a wide variety of smaller systems.

Features

• Four models - The DMP 64 Plus offers signal routing and management for a wide range of sound system applications:
  • Inputs: 32 Digital via Dante or Expansion Bus, 6 Analog balanced/unbalanced mic/line level, 8 Aux for USB or VoIP
  • Outputs: 16 Digital via Dante or Expansion Bus, 4 Analog balanced/unbalanced line level, 8 Aux for USB, VoIP or File Players
• 32x16 Dante audio networking with Dante Domain Manager and AES67 support — Dante audio networking provides scalability for creating larger audio matrixes over a local area network using standard protocols. A built in two port Gigabit switch can be configured to support primary and redundant Dante audio networks.
• Six channels of AEC - acoustic echo cancellation — The DMP 64 Plus includes six independent channels of high performance AEC, as well as selectable noise cancellation. Extron AEC features advanced algorithms that deliver fast echo canceler convergence for optimal intelligibility in situations that challenge AEC performance.
• Ability to be powered over PoE+ on AT models
• All models include a USB Audio interface, providing up to four channels of digital audio sends and returns — Support for USB Audio simplifies implementation of systems with either soft-codec conferencing or computer-based audio and enables use of the same interface for both small and large scale systems.

The Green-GO Dante interface links up to 8 audio streams from and into the Green-GO network.

Stay digital. The Dante interface will directly convert digital audio streams from Dante or AES67 to Green-GO and vice-versa, without the conversion losses that occur during analog to digital (and contrary) conversion.

This allows various audio signals to be injected into and extracted from the Green-GO network All Green-GO ports can be configured individually through the setup menu of the Green-GO Dante Interface. The Dante ports are manageable with the Dante Controller

Each of the 4 dual ports can operate in a full user based mode or a general purpose line in/out mode The Dante interface has a large TFT color screen for easy and quick setup and menu navigation. The display will give a quick overview of transmitted and received audio signals and levels.

Features

• Easily combinable with any other Green-GO Beltpack or Multi-Channel Station (no separate matrix or interface required)
• Connects up to 8 Dante/AES67 audio streams to and from the Green-GO network
• Get Audio into and out of the Green-GO system, like Program audio or announcements.
• Connect an audio source from anywhere in the world with a Green-GO bridge device
• Intercom routing data, system setup & preferences are stored in local memory
• Create your own configuration with the free Green-GO Control software
• Dimensions: (L/W/H): 485 x 160 x 45 mm Weight: 2,38 Kg
• PoE (802.3af) or optional external 12V DC Power Supply. Power consumption < 5 Watt

Harmonic Design® beamforming loudspeakers offer great potential for improving sound distribution in complex architectures and highly reverberant rooms. The goal is to focus the sound only on the listener’s surfaces, while shading areas that create unwanted reflections and disturbing room resonances. This leads to a much better speech intelligibility and a first-class listening experience for speech and music performances down to the last rows. Furthermore, the Beamforming technology, which offers a wider range of functions than just Beam Steering, enables unobtrusive positioning of the speakers even in less ideal locations for flexible adaptation to the most diverse requirements.

Beamforming Technology

• 2In/32Out High-End hd BeamDSP
• redundant input section with fallback algorithm
• analog & digital inputs with Dante®
• AdvancedDirectivityOptimization Algorithmus
• 384 Taps FIR filter per input/output
• 16x IIR filter (double precision) per input/output
• modular equipping in segments of 8
• 75W Class-D amplifiers with digital input for best SNR
• ImpedanceControl with fault detection
• multistage Limiter-Architecture
• hd LevelZ Remote control software via USB
• standard colors: black RAL 9005 & white RAL 9010

Models:

• hd ML8Beam
• hd ML16Beam
• hd ML24Beam
• hd ML32Beam
• hd PL8Beam
• hd PL16Beam
• hd PL24Beam
• hd PL36Beam

The PSP-D is a desktop subscriber panel. Besides the RGB illuminated rotary switches, the PSP-D is also equipped with dual GbE PoE powered network connectors, an internal speaker, a stereo headset interface and a 48V phantom powered microphone input. With their HiQ audio interfaces the PSP-D can also be used for voice-over or as a commentary unit

The PSP-T is the smallest subscriber panel of the PicaPi series. With its low profile of 32mm it can easily built into a table plate. It gets powered over Ethernet (PoE / 13W) and is equipped with internal microphone and speaker.

The PSP-R is a rack-mountable (19"/2RU) variant and the biggest user interface in the PicaPi series. Equipped with two wide-angle TFT touch displays it is possible to do intercom calls and using apps at the same time. The internal wideband speaker is powerful enough to be heard also in a noisy environment. The PSP-R offers two GbE network connectors, one of which can serve as a PoE power source.

The headset module (HSM) enables an attachment of a stereo headset into a Dante network. Either dynamic or electret microphones can be used. The headphone power level can be individually adjusted for left and right channel. With a power boost switch an additional gain of 9dB can be set for noisy environments. The module is PoE powered (13W).

With the GPIO module (GPM) GPIO signals can be attached to a Dante network. In potential free configuration 4 in and out can be used, in refrerence to a common potential up to 8 GPI / GPO's are available. The module is PoE powered (13W).

The PRU basically serves as a audio router with up to 4 interfaces with 64 channels each in a 19"/1RU case. Every single input can be routed to one or many outputs using the EmBER+ interface. For each of the 4 interfaces the protocol can be chosen to be either Dante (Audinate's Brooklyn II) or SMPTE-2110 (Ross Video's Liberty). The built-in sample rate converters allow therefore up to 4 independently synchonized clock domains to be connected to the PRU. Using two of each module type, a 128-channel bridge between Dante and SMPTE-2110 can be formed. Optionally PRU can be extended by a central intercom matrix. The 256x256 matrix is fully equipped with summing and level adjusting features.

Flex Focus loudspeakers employ shaded line array technology with ribbon HF drivers to extend sound projection in longer meeting rooms. Their Frequency response (120-20,000 Hz ±3 dB) has been optimized for speech intelligibility.

Flex Focus Vertical Arrays are available in two configurations. A standard version with loudspeakers oriented for monitors which are mounted at a normal viewing height and an LAC (Lowered Acoustic Center) version for monitors which are mounted above normal viewing height.

Flex Focus loudspeakers are also available in three powered configurations, each with 100- watt class D amplifiers:

• P – Basic power amp
• PX – Power amp with pre-amp & DSP
• PXD – Network version which adds Dante and ethernet control

Dante-enabled models include:

• Flex Focus FF-V2.4 PXD
• Flex Focus FF-V2.6 PXD
• Flex Focus FF-V2.6 PXD LAC
• Flex Focus FF-V2.10 PXD
• Flex Focus FF-V2.10 PXD LAC
• Flex Focus FF-H1.4-50 PXD
• Flex Focus FF-H1.6-50 PXD
• Flex Focus FF-H1.4-25 PXD

Flex Synergy Compact ultra slim conferencing platform which is custom sized to match the exact width of video displays from 46” to 70”. It is available in a standard version, with onboard powered DSP and a compact camera pocket (H 2.75″ x W 3.75″ x D 3.25″), and an enhanced version which adds Dante connectivity and Ethernet control. Its frequency response (120-20,000Hz ±3 dB) has been optimized for speech intelligibility.

Adding an Innovox Flex- Companion Series powered sub-woofer will extend low-frequency performance below 50Hz to deliver high impact full range music program.

Key Features

• Made to match exact width of Video Display
• Integrated camera pocket and microphone
• DSP and on-board class D power
• Network control and optional Dante Connectivity
• Works with Flex Companion Series powered subwoofer

Applications

• Support for Software-based Video Conference Applications
• Superior Audio for Collaboration and Huddle Spaces
• High Impact Playback when used with Companion-series Subwoofer

Flex Precision display mount loudspeakers employ ribbon HF and extended range LF drivers in an ultra slim enclosure. Their frequency response (120-20,000 Hz ±3 dB) has been optimized for speech intelligibility.

Flex Precision loudspeakers are also available in three powered configurations, each with 100-watt class D amplifiers:

• P – Adds two 100-Watt channels of Class D amplification with balanced analog inputs.
• PX – provides two 100-Watt channels of Class D amplification, plus DSP.  Inputs include balanced line level, unbalanced stereo, optical digital, speaker level, and USB. A processed sub-bass signal is provided for use with an Innovox Companion 2×6 powered subwoofer.  Control options include: IR, RS-232 & USB.
• PXD – Provides all of the above, plus Dante and Ethernet control.

Dante-enabled models:

• FP-V2PXD
• FP-H1PXD
• FP-H2PXD

The SOFIA 568 provides analog, AES3-digital and AoIP-streaming audio outputs. It collects histograms of signal parameters and dis-plays HD Radio™ album artwork, station logos and similar visuals via the Web interface.

Multiple stations can be sequentially monitored using the programmable StationRotation™ feature. The SOFIA 568 includes a built-in BandScanner™, a real-time clock and full SNMP functionality.

The elite SiteStreamer+™ (‘Plus’) series offers advanced features for the most critical monitoring situations.

SiteStreamers™ also deliver important RF and audio signal measurements and dispatch email or text-message alarms for out-of-limits conditions and other reception errors.

The Just Add Power 3G+ Dante® Sound System Transmitter adds compatibility with Dante® and AES67 audio networks to the UltraHD over IP system.

Input up to 8 channels into the Dante® network. Extract up to 8 channels from the Dante® network, embed onto the HD over IP signal, and distribute across the Just Add Power network. Mix-and-match! The 3G+ Dante® Sound System Transmitter is fully compatible with all 2GΩ/3G and 3G devices. Not just for Dante®. AES67 compatibility for interoperability with other leading audio-over-IP technologies.

The KH5 is the biggest speaker of the K-array Concert Series and represents a revolutionary compact line array element for large scale events.

The KH5 has 2 x 15" neodymium woofers with a 3" voice coil and 4 x 1 . 4"  compression drivers.

The K-array KS5 is a self-powered subwoofer system. It is best suited for high-powered, extended bass response applications in arenas, theatres, stadiums, concert halls, houses of worship and outdoor events.

The 21" loudspeakers feature Neodymium magnets with 6" voice coils driven by powerful Class D amplifiers

The rear panel provides inputs for a balanced line signal, a balanced microphone signal with phantom power, and digital signals in AES/EBU protocol, also on an XLR for ease of cabling.

KH2 speakers are part of K-array Concert Series, which includes a wide range of professional audio products serving as sound system for small, medium and big scale events. The KH2 is a revolutionary compact line array element ideal for club or a theater.

The smallest mid-hi line array element offered in the Concert series is comprised of 2 x 8" neodymium magnet woofers each with 2 . 5" voice coil and 1 . 4" compression drivers with a 2 . 5" voice coil each.

However small in size, the KH2 delivers a really large sound, ensuring exceptionally high performance with a peak SPL of 136 dB.

The KH2SYS6 is an incredibly compact and lightweight system perfect for intimate indoor settings with the possibility to expand for larger venues.

Lightweight, compact solutions to meet small venue needs

Exceptionally high performance with a peak SPL of 136 dB

The ADX-2400D is the AES version of the ADX system head-end unit. It provides the network interface for 24 channels of AES digital audio in each direction (inputs and outputs). This unit may be used as the 'truck end' of an announce booth system using ADX-120s and ADX-140s, may be used with other ADX-2400s to create a networked transmission system, or may be used stand-alone as a high-quality, compact 24-channel delay unit.

In the Announce Booth mode each ADX-2400D supports up to four remote devices, either the ADX-120s or ADX-140s in any combination.  Multiple ADX-2400s (analog and/or AES) may be operated on the same network to expand system capabilities to meet any required system configuration.

The Duet digital wireless monitor system represents a significant step forward for touring performers and other professionals requiring high definition audio in their ears. With 24 bit digital audio, low latency, Dante inputs and a compact, lightweight, diversity bodypack unit, the Duet system should please even the most demanding users, both on and behind the stage.

The new Duet digital wireless monitor system consists of the M2T dual-stereo half-rack transmitter and M2R diversity belt pack receiver. The Duet covers the UHF frequencies of 470-608 MHz (138 MHz) in a single range, uses digital modulation for transmission, and can accept analog or Dante™ digital inputs. The new system can be considered a "wireless IEM" or a deluxe IFB product, designed and developed with professional touring, installation, theater, filmmaking and broadcast customers in mind.

The M2T half-rack transmitter houses two independent stereo transmitters allowing for up to four stereo or dual-mono transmissions in a single rack space. The audio inputs can be individually configured to be analog or Dante compatible. The analog input connectors are full size XLR/TRS combo types for balanced line level analog signals while the input preamp circuits use a special balanced amplifier with very high common mode rejection to minimize hum and noise. A Dante Ultimo™ interface via dual RJ45 connectors accepts Dante networked audio inputs and can cascade the digital stream to additional units via CAT6 cables. An additional RJ45 jack provides an Ethernet connection for programming and control via Wireless Designer™ software and a USB jack on the front panel allows for firmware updates. The transmitter features a full color, backlit LCD for high visibility in any environment, while the chassis is all-metal and can be racked singly or in pairs using included rack mount hardware.

The M2R bodypack receiver employs advanced antenna diversity switching during digital packet headers for seamless audio. The 24 bit digital audio stream guarantees high resolution sound quality with wide dynamic range, low noise floor, and rock-solid stereo image. The headphone jack is fed from a high-quality stereo amplifier with 250 mW available to drive headphones or earphones to sufficient levels for stage performance or other noisy environments. A high-resolution, color LCD allows for a wide range of menu options, including precise limiter settings, using detailed graphics. The M2R runs on two AA batteries for an operating time of over 4 hours with alkaline cells.