Dante-enabled Product Catalog

ARES8d HQ is a dual 8’’ enclosure line array with and 1.4’’HF driver and a 3’’ voice coil. Its B&C driver coupled to a waveguide provides a true cylindrical dispersion.

Thanks to its dedicated class D Powersoft amplifier and AUDIOFOCUS OEM DSP treatment (96kHz-40 bits floating point) with 24 presets, one cabinet can reach a peak level of 143 dB SPL (preset 64m).

System designer and sound engineer can control and monitor every speaker individually thanks to AUDIOFOCUS Network manager software and for more demanding needs the Dante version constitute the optimal needed tool to significantly boost sonic performances.

Thanks to dedicated custom AUDIOFOCUS presets, coupling and distance compensation are already calculated, making the deployment of the system faster and totally Plug and Play.

The APB-112 SB-D is an active, portable, AudioPressBox unit, 1 channel DANTE input, 12 LINE/Mic outputs.

• The portable DANTE-enabled unit features 1 channel DANTE input with high quality digital to analog converter.
• Powered via PoE from DANTE-network.
• The unit has 12 balanced, individually transformer isolated Line/MIC outputs.
• The unit is powered only via PoE from DANTE network.
• This is a PoE level 3 device

The M70WD is a steerable, flush-mount condenser microphone designed for distance miking in a variety of applications, including video conferencing, distance learning, hospital rooms, surveillance, and ambient room miking. The flush-mount configuration and a footprint of only three inches in diameter allow the M70WD to virtually disappear.

The M70WD capsule is housed within a patented cradle that allows the microphone to be aimed up to an angle of 45 degrees. Other features include a fully integrated preamp and an extremely high sensitivity rating of 38 millivolts. In addition, the M70WD has complete immunity from RF interference caused by cell phones and GSM devices. Installing the M70WD requires just one 2-inch hole in the ceiling, with no additional tools needed.

Features

• High output for distance miking
• Aimable up to a 45-degree angle
• Miniaturized integrated preamp circuitry
• Immunity from RF interference
• Two-color LED status indicator
• RJ45 connectors for plug-and-play integration with the Audix Dante | AES67 Integrated Microphone System
• No external power adaptor required
• Designed, assembled, and tested by Audixin the USA
• 3-year warranty
   

Dante Made Simple - By Audix

The M70WD is designed exclusively for the Audix Dante | AES67 Integrated Microphone System , which delivers high-quality audio and all microphone functions - including on-off contact closure and LED status indicators - through a single CAT5 - CAT7 cable with RJ45 connections. This simple configuration eliminates wiring errors, accelerates installation, and reduces cost. And like all Audix microphones in the system, the gain structure is optimized for its intended use, providing quality audio at the DSP. The result is true plug-and-play installation.

The M55WD is an innovative, hanging ceiling microphone system designed for applications where aesthetics, sound quality, and ease of installation are critical. All electronics are fully integrated with high sensitivity and low noise. The signal output is balanced to eliminate RF interference caused by cell phones and mobile devices. 

Installing the M55 requires just one 5/8-inch hole in the ceiling, with no additional tools needed. The M55's mounting hardware easily adjusts the hanging height up to four feet from the ceiling surface without removing ceiling tiles and allows for fingertip height adjustment and roHigh output allows distance miking.

Features 

• Optimized for voice recognition
• Low-noise preamp circuitry
• Immunity from RF interference
• Two-color LED status indicator
• RJ45 connectors for plug-and-play integration with the Audix Dante | AES67 Integrated Microphone System
• No external power adaptor required
• Designed, assembled, and tested by Audix in the USAtation control. An optional aiming clip (MCHANGER) easily makes any additional angular adjustments

Dante Made Simple - By Audix

The M70WD is designed exclusively for the Audix Dante | AES67 Integrated Microphone System , which delivers high-quality audio and all microphone functions - including on-off contact closure and LED status indicators - through a single CAT5 - CAT7 cable with RJ45 connections. This simple configuration eliminates wiring errors, accelerates installation, and reduces cost. And like all Audix microphones in the system, the gain structure is optimized for its intended use, providing quality audio at the DSP. The result is true plug-and-play installation. 

The DN4 microphone interface is a key part of the Audix Dante | AES67 Integrated Microphone System, the fast, trouble-free way to stream Audix audio performance through Ethernet networks. High-quality audio and all microphone functions - including on-off contact closure and LED status indicators - are available through a single CAT5 - CAT7 cable with RJ45 connections at both interface and microphone. This simple configuration eliminates wiring errors, accelerates installation, and reduces cost.

The DN4 features four inputs for Audix RJ45-equipped, single-element microphones and two 5V logic outputs per port, with source/sink capability for LEDs that are controlled by network-connected DSPs and one logic input per port for buttons. Low-pass and high-pass filter settings are available on each audio channel, as are eight selectable gain levels, with all settings configurable via software.

Audio Features

• Compatible with all Audix RJ45-equipped, single-element, analog microphones 
• High-fidelity 24-bit, 48kHz audio
• Balanced audio inputs with RFI filtering
• 48V phantom power output
• Each channel has eight selectable pre-amp gain levels, as well as low- and high-pass filters
• All channels controlled independently
• Flash memory prevents loss of settings in the event of power loss

Dante® Features

• Supports AES67 audio streams
• Dante Domain Manager (DDM) ready
• Dynamic or static IP address
• Evaluated to the requirement of UL2043 and is suitable for use in above-ceiling applications

Power Features

• Operates on PoE from any IEEE 802.3af compliant equipment

The DN43 microphone interface is a key part of the Audix Dante | AES67 Integrated Microphone System, the fast, trouble-free way to stream Audix audio performance through Ethernet networks. High-quality audio and all microphone functions - including on-off contact closure and LED status indicators - are available through a single CAT5 - CAT7 cable with RJ45 connections at both interface and microphone. This simple configuration eliminates wiring errors, accelerates installation, and reduces cost.

The DN43 features one microphone input specifically designed for the Audix M3 tri-element microphone and one input for Audix RJ45-equipped, single-element microphones. It has two 5V logic outputs per port, with source/sink capability for LEDs that are controlled by network-connected DSPs. Low-pass and high-pass filter settings are available on each audio channel, as are eight selectable gain levels, with all settings configurable via software.

Audio Features

• High-fidelity 24-bit, 48kHz audio
• Balanced audio inputs with RFI filtering
• 48V phantom power output
• Each channel has selectable pre-amp gain, as well as low- and high-pass fi lters
• All channels controlled independently
• Flash memory prevents loss of settings in the event of power loss
• Compatible with the M3 tri-element microphone (DN43 only), the flush-mount,
• steerable M70WD ceiling microphone, and the M55WD ceiling microphone

Dante Features

• Supports AES67 audio streams
• Dante Domain Manager (DDM) ready
• Dynamic or static IP address
• Evaluated to the requirement of UL2043
• and is suitable for use in above-ceiling
• applications

Power Features

• Operates on PoE from any IEEE 802.3af compliant equipment

BR-4*4IO Dante audio converter box is equipped with 4 line/mic inputs, phantom power and balanced audio output. The gain, phantom power and mute on each input can be debugged by software. Dual backup network port design, external power supply. With many high-quality features, and no design, no need to learn. Provides a complete audio processing solution for the optimization of specific locations and systems. Suitable for Dante audio conversion to upgrade the traditional audio system.

Blustream's DA11AU has been designed to convert unbalanced or balanced 2ch analogue audio to Dante® digital signal. The DA11AU allows any non-Dante® audio source such as mixing consoles, amplifiers, computers, and Blustream Matrix or distribution products to be connected as a source within a Dante® network. The DA11AU is a plug & play device that is powered using PoE (Power over Ethernet) from any PoE network switch and supports AES67 RTP audio transport.

Bosch's OMN-ARNIS ARNI-S is an OMNEO interface that provides better scalability through the use of either a DHCP server or DNS server, which extends the use of PTP over multiple subnets. The ARNI-S supports redundancy, and up to 450 OMNEO nodes in a single subnet network, or up to 450 OMNEO nodes in each subnet of a routed network when used in conjunction with an ARNI-E device.

Bosch's OMN-ARNIE ARNI-E is an OMNEO Interface that provides better scalability through the use of either a DHCP server or DNS server, which extends the use of PTP over multiple subnets. It also supports redundancy, and up to 450 OMNEO nodes across a routed network with multiple subnets, or up to 10.000 nodes across 40 subnets when used in conjunction with an ARNI-S device in each subnet.

The Tracking Intelligent Microphone (TIM) breaks the mould for ceiling microphones. Designed for use in AV conference and huddle room applications.

• Tracks meeting participants within a room
• APT technology for best possible sound output
• Eliminates unwanted steady state noise e.g. air conditioning , projectors
• Built in equalizer
• Supports DANTE and AES 67
• Intutive GUI through standard Web browser
• Powered over Ethernet (PoE+)
• Separate control and data paths for added security
• API commands via UDP
• Easy setup and install

DiGiCo's DMI-Dante 64@96 card is a Dante Interface Card for its 4REA4, S-Series and select SD-Range sound consoles (SD12 and SD7 Quantum). Providing 64 input and 64 output channels at both 48KHz and 96kHz, the new DMI (DiGiCo Multichannel Interface) card is equipped with Primary and Secondary (backup) Gigabit Ethernet ports for connection and uniquely offers a Sample Rate Conversion feature. Natively allowing DiGiCo products to communicate with other devices on a Dante network, the DMI-Dante 64@96 effectively doubles the previous DMI Dante card’s I/O channel count at the higher 96kHz resolution.

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.