The DCon series features three advanced products, all designed to fit standard European
in-wall junction boxes with a depth of 50mm and compatible with widely used EU cover
frames, making them perfect for seamless integration into commercial, educational,
and entertainment AV environments.
This two-channel XLR output panel converts Dante audio to analog. Its compact and sleek
design allows it to fit snugly into junction boxes, ensuring a clean and professional
installation. DConXo offers seamless connectivity between digital Dante networks and
traditional analog audio systems
The DCon series features three advanced products, all designed to fit standard European
in-wall junction boxes with a depth of 50mm and compatible with widely used EU cover
frames, making them perfect for seamless integration into commercial, educational,
and entertainment AV environments.
This 2×2 Bluetooth converter brings wireless audio flexibility to any Dante installation.
DConBT offers a host of control features, including call pickup, reject, music player control,
and customizable pairing button behavior. Additionally, users can define the Bluetooth radio
name to personalize their wireless audio experience. Like its counterparts, DConBT is
API-enabled for easy integration with third-party applications.
The DCon series features three advanced products, all designed to fit standard European
in-wall junction boxes with a depth of 50mm and compatible with widely used EU cover
frames, making them perfect for seamless integration into commercial, educational,
and entertainment AV environments.
This two-channel XLR input panel converts analog audio to Dante, the industry-standard
audio-over-IP protocol. Each input is equipped with an adjustable preamp that offers up to
42dB of amplification plus -100 to +27dB digital gain, along with switchable Phantom Power
for handling microphones and other audio sources with precision. DConXi also provides an
API for integration with third-party control systems, making it an ideal choice for customized
installations.
The Brick4.4 is a compact and versatile 4×4 Dante audio interface designed for seamless integration of analog audio into professional Dante networks. Its slim surface-mount design allows flexible installation in racks, under tables, or inside AV enclosures, making it ideal for commercial, educational, and entertainment environments.
The unit features four balanced MIC/LINE inputs and four balanced LINE outputs, enabling bidirectional conversion between analog audio and Dante. Each input channel includes an adjustable gain control up to 45 dB and individually switchable 48V phantom power, allowing precise gain control for btoh microphones and line-level sources.
Supporting 48 kHz and 96 kHz sampling rates, the Brick4.4 ensures high-quality audio performance. Clear front-panel LED indicators provide status monitoring for power, system,
synchronization, and signal activity. The device can be powered via standard PoE or 5V USB-C, offering flexible installation options for professional AV systems.
DCS-DIC is the core device of a public address and voice alarm (PA/VA) system, the DCS-DIC combines advanced Dante audio networking with high-performance DSP audio processing. Built on mature Ethernet technology, it delivers the benefits of centralized management and distributed control, while offering strong adaptability and scalability. Whether deployed as a small-scale broadcasting system with a single DCS-DIC or as a large-scale system formed by expanding multiple DCS-DIC units or using them in stacked redundancy, it ensures stable audio quality together with high security and flexibility. The system can operate independently on a dedicated broadcasting network or be integrated into an existing local area network, significantly simplifying deployment and reducing operating costs.
ACE is a multifunctional audio interface, converter, matrix and digital converter in a single package. Main professional applications are for live environments, broadcast and installation whenever flexibility and high audio quality are required.
The Extron AXI 22 AT D Plus is a single-gang decorator-style Dante® audio interface for integrating two mic/line sources onto a Dante-enabled audio system. It features two XLR audio inputs and switchable 48 volt phantom power for connecting and powering condenser microphones, and remote control via the network. The AXI 22 AT D Plus also features two line outputs for routing any two Dante channels from the network to the analog input of a destination device, or connecting into an optional secondary wallplate with two XLR outputs. The AXI 22 AT D Plus interfaces with any Dante-equipped audio processor, such as an Extron DMP 128 Plus AT, over a standard local area network, and can be powered through PoE. This allows a single network cable connection for bidirectional audio, control, and power from a central equipment rack.
The DU44 can be used with multiple AlSpeech high-end microphone products. it is seamlessly integrated withfor the AIMIC-MCS06 al-in-one ceiling microphone with built-in pickup and Voice Lift, supporting Dante Auto-Routing to significantly simplify system deployment and commissioning. Combined with USB or PoE powersupply and low-latency transmission, the DU44 provides a reliable, simple, and stable audio interconnection solution for professional conferencing systems.
The Model 395 GPO Interface supports two general-purpose output (GPO) signals for a range of applications. The unit is housed in a lightweight, aluminum, tabletop enclosure.
The Model 394 GPI Interface supports two general-purpose input (GPI) signals for a range of applications. The unit is housed in a lightweight, aluminum, tabletop enclosure.
he Model 544D is an audio interface that supports Dante networking and provides two analog line-level inputs and two analog line-level outputs. Also includes two installer-accessible general-purpose inputs (GPI) and general-purpose outputs (GPO). The Model 544D is designed to serve as a dual-channel analog input/output interface for various applications. Configured using the STcontroller software application. Housed in a lightweight tabletop enclosure; optional mounting kits available. Power-over-Ethernet (PoE) powered. Can also be powered by 12 volts DC.
The Model 543D IFB Interface provides broadcast-standard powered and line-level (non-powered) analog IFB outputs from audio signals from a Dante network. IFB, also known as "interruptible foldback" or talent cueing, is a method commonly used by broadcast on-air talent and related personnel to receive one-way (listen-only) audio signals associated with live-event broadcasts. Especially important in sports and entertainment events, IFB plays a crucial role in virtually all broadcast applications that require people to stay "in the know." Dante has found wide acceptance as an audio "backbone" due to its ease of use, high performance, strong interoperability, and wide adoption by a large number of equipment manufacturers. Key Features Dante audio-over-Ethernet technology Two powered 2-channel IFB outputs Two line-level IFB outputs Excellent audio quality Level metering and status LEDs Uses STcontroller for configuration PoE and 12 volts DC powering Table-top, portable, or optional rack-mount use
A compact A/V switcher in the EasyIP ecosystem, the EasyIP Dock connects in EasyIP cameras, video, and audio to a computer while simultaneously extending that computer’s HDMI output over the network for flexible presentation and conferencing spaces.
Features:
Manages up to four EasyIP cameras or EasyIP video sources with switching and camera control
1080p/60 video over USB 3.0 with USB 2.0 backwards compatibility and AEC
HDMI In converts to EasyIP output for networked extension with simultaneous HDMI output loop
8x8 Dante audio matrix (15x13 full cross-point audio matrix)
EasyIP Multicast allows content sharing with any number of EasyIP endpoints or destinations
The BSS Soundweb OMNI 32e Open Architecture I/O Expander offers the ability to onramp and offramp Analog/AES3 to/from Soundweb OMNI 512p and 256p Digital Signal Processors. The 32e converts these signals to and from Dante/AES67 for processing by Soundweb OMNI DSPs. The all-new design ensures the seamless handling of today’s most complex installations to be future-ready to meet the evolving needs of tomorrow.
The 32e features 8x8x8 Analog or 16x16x16 AES3 audio channels; 32x32 Dante/AES67 networked audio I/O channels; 24 GPIO ports (fully flexible); enterprise-grade network security; studio quality 60dB microphone pre-amp on all inputs; high resolution DAC and ADC, primary and secondary Dante connections, redundant HControl ports; full-color, front panel LCD with metering; stereo AES3 on every audio connector; and more.
Each audio I/O port on Soundweb OMNI I/O Expanders is format configurable as either analog or stereo AES3 and eight of the audio I/O ports are also configurable to be either inputs or outputs. All GPIO ports are configurable as either input or output ports, and their format can be either analog or digital.
With support for centralized or distributed DSP topologies, system integrators can add as many Soundweb OMNI processors to a single project file as needed, while still enjoying the simplicity of a single canvas for device configuration.
HARMAN's HControl protocol, a human-readable, standards-based, developer-friendly API is also supported. It offers an intuitive, flexible, and powerful alternative to other protocols to support developers working at any scale.
Soundweb OMNI systems are configured, controlled, and monitored with the AVX software suite through two user-friendly applications. AVX Architect is a desktop app for system diagramming, device configuration, network management, and custom UI design. An unlimited number of processors and expanders can be added to a single project file. AVX Control is a lightweight app offering easy access to custom UIs for controlling and monitoring Soundweb OMNI devices, via Windows, iOS, macOS, Android, and AMX Varia touch panels. These custom UIs also work with Crown DCi amplifiers and previous generation Soundweb London devices.
Built on decades of innovation, BSS Soundweb OMNI is a highly scalable and flexible DSP solution. Combining powerful hardware, intuitive software, and developer-friendly features, it effortlessly scales to meet the needs of any installed application. Experience BSS Soundweb OMNI, the new standard in open architecture DSP.
The BSS Soundweb OMNI 16e Open Architecture I/O Expander offers the ability to onramp and offramp Analog/AES3 to/from Soundweb OMNI 512p and 256p Digital Signal Processors. The 16e converts these signals to and from Dante/AES67 for processing by Soundweb OMNI DSPs. The all-new design ensures the seamless handling of today’s most complex installations to be future-ready to meet the evolving needs of tomorrow.
The 16e features 4x4x4 Analog or 8x8x8 AES3 audio channels; 16x16 Dante/AES67 networked audio output channels; 12 GPIO ports (fully flexible); enterprise-grade network security; studio quality 60dB microphone pre-amp on all inputs; high resolution DAC and ADC, primary and secondary Dante connections, redundant HControl ports; full-color, front panel LCD with metering; stereo AES3 on every audio connector; and more.
Each audio I/O port on Soundweb OMNI I/O Expanders is format configurable as either analog or stereo AES3 and four of the audio I/O ports are also configurable to be either inputs or outputs. All GPIO ports are configurable as either input or output ports, and their format can be either analog or digital.
With support for centralized or distributed DSP topologies, system integrators can add as many Soundweb OMNI processors to a single project file as needed, while still enjoying the simplicity of a single canvas for device configuration.
HARMAN's HControl protocol, a human-readable, standards-based, developer-friendly API is also supported. It offers an intuitive, flexible, and powerful alternative to other protocols to support developers working at any scale.
Soundweb OMNI systems are configured, controlled, and monitored with the AVX software suite through two user-friendly applications. AVX Architect is a desktop app for system diagramming, device configuration, network management, and custom UI design. An unlimited number of processors and expanders can be added to a single project file. AVX Control is a lightweight app offering easy access to custom UIs for controlling and monitoring Soundweb OMNI devices, via Windows, iOS, macOS, Android, and AMX Varia touch panels. These custom UIs also work with Crown DCi amplifiers and previous generation Soundweb London devices.
Built on decades of innovation, BSS Soundweb OMNI is a highly scalable and flexible DSP solution. Combining powerful hardware, intuitive software, and developer-friendly features, it effortlessly scales to meet the needs of any installed application. Experience BSS Soundweb OMNI, the new standard in open architecture DSP.
Designed for broadcast, installed AV, and live production, Auro Versa offers scalable, customizable I/O with flexibility for evolving AoIP networks. Built on an 8-slot modular chassis, it allows users to configure up to 64 I/O channels in 8 channel increments. Available modules include Analog Mic/Line, AES/EBU and GPIO. Custom OEM modules are also available.
Auro Versa is engineered for long-term adaptability with updateable AoIP protocols. This feature will enable users to upgrade to emerging audio networking standards without replacing their core I/O hardware, ensuring a future-proof investment in their audio infrastructure.
The CXD-32CF 32×32 I/O Digital Audio Network Interface is a 32×32 48kHz I/O device that supports Dante® and AES67. It provides routing and expansion capabilities into and out of a CrewNet infrastructure via EtherCon Copper and Dual LC Fiber connections. This half-rack device can be powered with either net power over a CrewNet copper connection or via the supplied 48VDC power supply (PPS-48V-02).orking in the field.
KEY FEATURES:
32 x 32 I/O Dante-Based Digital Audio I/O
Includes an AES67 Operation Mode
Single EtherCon/Dual LC Fiber CrewNet Port
Primary and Secondary Redundant EtherCon/Dual LC Fiber Dante Ports
Supports 48kHz Sampling Frequency
Front and Rear Panel Status Indicators
Micro USB Connection for Firmware Updates
Dual Powering Options
Network Power (Power over CrewNet)
External Power (48VDC) (Included with Product)
Compact 1U Half-Rack Size Chassis Design with Forward or Rear-Facing Orientation (rack kits sold separately)
The IP-8i is an AES67 audio interface allowing broadcasters to convert their audio in AES/EBU (8 stereo channels) or Analogue (4 stereo channels) format to AES67 (RAVENNA, Livewire, Dante).
A total number of 8x RS232 inputs are available for transport of UECP/RDS/PAD data.
The IP-8i offers you a straightforward audio to AES67 interface offering a variety of features and opportunities to ease your daily work.
Flexible in application: The encoder is designed for easy converting audio to AES67. In case of audio silence, the IP-8i can automatically switch to an (optional) Icecast/shoutcast stream or play files from the internal memory.
Cobalt® introduces the ARIA OG-AUD2-DANTE card in a openGear® form factor. The COBALT ARIA OG-AUD2-DANTE is a 2x channel unit with 12G SDI and DANTE/ AES/MADI which can simultaneously embed and de-embed audio between SDI, DANTE, AES and MADI with flexible routing and mixing. This card also includes a built-in frame sync. This openGear® card features two gigabit Ethernet ports, and a 32x32 Dante channel matrix mixer. The Cobalt ARIA OG-AUD2-DANTE — High-density, compact openGear® card-based solution, with multiple devices is able to be combined into a single frame for multi-channel operation, as well as offering the standard features of redundant hot-swappable power supplies and hot-swappable cards. Free DashBoard™ software for remote control and configuration.
The Spectera Base Station is a 1RU hardware device that provides up to 64 channels (32 in/32 out) and utilizes up to two wideband RF carriers to accommodate all of the unit’s wireless transmission.
It features redundant Dante and optional redundant Madi, assuring seamless integration into your audio network. Additional antennas can be added for redundancy, extended zone coverage, or additional spectrum capacity.
The Spectera Base Station supports sample rates of 48 kHz and 96 kHz and features individual sample rate converters for all audio interfaces and wordclock I/O. Base Station must be activated with license per region.
19’’ 1 RU Base Station with redundant power supplies
Provides up to 64 channels (32 in / 32 out) and allows for up to two independent wideband RF carriers in UHF or 1G4 (6 or 8 MHz) – country specific per region license
4 ethernet antenna ports for redundancy, extended zone coverage, and additional spectrum usage
Redundant Dante® audio interface
Two optional slots for redundant MADI (OM or BNC)
Sample rates of 48 kHz and 96 kHz
Individual sample rate converters for all audio interfaces
Wordclock I/O BNC
Synchronized clock for all audio over RF allows 3D recordings and playback
Adjustable RF power from 10 – 100 mW (country specific)
RJ45 network interface for full remote control with secured interface
11 Audio Link modes with selectable audio codecs (OPUS, SeDAC) or PCM
Secure transmission of confidential content using AES 256 encryption
Future expansion via Cascade port
Selective headphone output with volume control on front panel
OLED display, jog wheel navigation, and rugged metal housing
Base Station activation via LinkDesk: Spectera license defines respective frequency spectrum per region (license sold separately)
THE PERFECT LINK The MT 48 brings Neumann quality to the digital workflow. It is the first audio interface in its class that is able to capture the full dynamic range of Neumann microphones and is optimally matched to the KH-Line of studio monitors and Neumann headphones. The result is an ideal signal chain without any bottlenecks: uncompromising sound quality from the microphone diaphragm to your ear.
SWISS PRECISION INSIDE The MT 48 was developed in close cooperation with the Swiss manufacturer Merging Technologies, whose high-precision AD converters offer a staggering dynamic range of 136 dB. This gives the MT 48 four times the resolution of comparable audio interfaces. The MT 48 comes fully equipped for the reference-class personal studio: two extremely low-noise mic/line preamps with up to 78 dB gain, two line/instrument inputs, four monitor/line outputs and two superb headphone outputs, each with state-of-the-art AD/DA converters.
The MT 48’s connectivity may be expanded via ADAT / S/PDIF and AES67; a MIDI interface is also integrated. Four separate monitor mixes can be controlled directly on the unit via touchscreen. On-board processing includes a fully parametric 4-band EQ and a 3-way dynamics processor (gate/compressor/limiter) per channel as well as a high-quality reverb. So musicians get inspiring sound for top performances. And there is still plenty of DSP power for future developments.
The DP648 retains the popular 4-input, 8-output format and features an optional Dante card, allowing for a combination of AES3, analog, and Dante/AES67 to be used. Comprehensive system processing is provided by dedicated input and output processing similar to the DP448 but enhanced with the multiband dynamic DEQ algorithms from the XTA DP548 and SIDD processors.
The system offers full front panel control and Ethernet remote control via Audiocore Amped Edition, with integration in globcon XTA MC² Edition coming later this year. The combination of dynamics processing, IIR, all-pass, and phase filters, paired with very low latency (0.5ms), makes it the perfect choice for system engineers managing live shows, theatres, and clubs.
The DP648 is supported by an extensive preset library and introduces the X Channel Preset concept built into XTA hardware for the first time.
The DP648 completes the system for XTA. The MX36 provides fast source switching, the DP648 offers comprehensive system management, and its DPA and DNA amplifiers bring the power to drive systems large and small. Control is enhanced by the grouping system which works across multiple DP648 processors and DPA amplifiers. The 32 groups behave like a virtual 32-output processor.
Ultra-Flexible Networked Audio Mainframe with Superior Sound Quality Hapi Mk III is an exceptional audio centerpiece that effortlessly adapts to the most demanding scenarios. With its customizable I/O options, it offers unparalleled networked audio solutions, making it the perfect choice for broadcast, live events, and studios. This Swiss-made device is crafted to the highest standards and delivers audio quality that surpasses industry standards.
Third party applications that use Dante Ready™ allow customers to purchase and activate Dante audio channels after installation using the Dante Activator tool in Dante Controller 4.5 or higher. Dante Activator is tied to an active Audinate user account and accepts standard credit card payments.
