Dante-enabled Product Catalog

D4.4K-D with their 48kHz sampling rate and 24-bit quantization precision, combined with IIR and FIR filter technologies, accurately generate linear phase curves, significantly improving pulse response quality. Perfect Integration of BrainCoreTM Technology Perfectly integrating BrainCoreTM technology, providing precise digital processing through superior Limita™ processing technology, ensuring the system operates safely and reliably. AES Digital Audio Input Supporting standard AES/EBU digital audio input, we offer customers a variety of widely recognized digital audio input solutions in the industry for high-quality audio transmission and precise control of audio equipment. Dante Network Audio Transmission and Control Integrating Dante network audio transmission and control technology, allowing for long-distance, high-precision audio signal transmission and control over Ethernet, ensuring stable and reliable sound quality.

D4.2K-D with their 48kHz sampling rate and 24-bit quantization precision, combined with IIR and FIR filter technologies, accurately generate linear phase curves, significantly improving pulse response quality. AES Digital Audio Input Supporting standard AES/EBU digital audio input, we offer customers a variety of widely recognized digital audio input solutions in the industry for high-quality audio transmission and precise control of audio equipment. Dante Network Audio Transmission and Control Integrating Dante network audio transmission and control technology, allowing for long-distance, high-precision audio signal transmission and control over Ethernet, ensuring stable and reliable sound quality. Supports dual signals and hot backup, with seamless switching Capable of handling both Dante digital and analog signals, it features automatic hot backup to ensure immediate and seamless switching in case of primary audio source failure, maintaining stable transmission. Users can easily manage backups, ensuring audio reliability.

D Series is a four-channel digital power amplifier that integrates Class D power amplification technology with high-precision DSP digital signal processing, enhancing audio performance and ease of operation. It comes with IIR filter, FIR filter, and optional Dante network functionality for precise audio tuning and optimization. Users can configure and control the amplifier rapidly via the display and USB interface on the front panel and the Ethernet. Thanks to its exceptional performance and user-friendly software interface, the D Series amplifier is the perfect choice for professional audio systems. Audinate’s Dante audio networking technology is a widely adopted networked audio solution in the world, extensively used in commercial installations, live performances, recording and production. The D Series digital power amplifiers feature full Dante capabilities, with a channel capacity of up to 512×512, ensuring immediate and seamless compatibility with the complete Dante system.

D Series is a four-channel digital power amplifier that integrates Class D power amplification technology with high-precision DSP digital signal processing, enhancing audio performance and ease of operation. It comes with IIR filter, FIR filter, and optional Dante network functionality for precise audio tuning and optimization. Users can configure and control the amplifier rapidly via the display and USB interface on the front panel and the Ethernet. Thanks to its exceptional performance and user-friendly software interface, the D Series amplifier is the perfect choice for professional audio systems.Audinate’s Dante audio networking technology is a widely adopted networked audio solution in the world, extensively used in commercial installations, live performances, recording and production. The D Series digital power amplifiers feature full Dante capabilities, with a channel capacity of up to 512×512, ensuring immediate and seamless compatibility with the complete Dante system.

Designed for modern broadcast, live production, and critical audio environments, the RM-X-64-DNT reference monitor range delivers uncompromising flexibility and reliability in a compact 1U form factor. As standard, the unit features seamless Dante AoIP and MADI integration, intuitive control, fast source selection, and full redundancy, serving as the monitoring backbone of any professional workflow. Product Features

• 128 channels, 64-channel Dante AoIP and 64-channel MADI.
• 2 expansion slots for additional sources.
• Expansion source options:
  • 8 Analogue input channels.
  • 8 Stereo Digital inputs (16-channels).
  • 12G-SDI inputs (up to 64-channels).
• Built-in ported speakers with integrated grille.
• Built-in microphone for talkback.
• Digital & analogue XLR line out.
• Full power and data redundancy.
• AC and 12V DC power inputs.
• Ambient light sensor.
• Fully Dante compliant.
• AES67 compatible.
• Compact 1U design.
• Depth less than 20.5cm (8 inches).

Designed for broadcast, live production, and professional AV environments, the AVN-AESIO8-T-DNT provides a cost-effective and dependable solution for transporting low-latency audio between AES3 and Dante network domains. Product Features

• Dante AoIP.
• 8 x balanced digital stereo AES3 inputs on terminal block connectors, supporting input rates of 32kHz – 192kHz.
• 8 x balanced digital stereo AES3 outputs on terminal block connectors, output matches Dante system sample rate.
• 2 x RJ45 connectors (1Gb/s Ethernet Port) allowing the unit to operate in redundant or switched modes.
• PoE and link LED status indicators for each ethernet port.
• Front panel clock, PoE and link LED status indicators.
• Front panel AES3 lock LED status indicators for each input.
• Powered by PoE on either (or both) ethernet ports, offering power supply redundancy.
• Alternative USB-C DC power input, when PoE is unavailable.
• Configuration using Dante Controller.
• Dante Domain Manager compliant.
• AES67 operation.
• 1U 19″ rack-mount aluminium chassis.

Designed for broadcast, live production, and professional AV environments, the AVN-AESIO8-DNT provides a cost-effective and dependable solution for transporting low-latency audio between AES3 and Dante network domains. Product Features

• Dante AoIP.
• 8 x balanced digital stereo AES3 inputs on Neutrik XLR connectors, supporting input rates of 32kHz – 192kHz.
• 8 x balanced digital stereo AES3 outputs on Neutrik XLR connectors, output matches Dante system sample rate.
• 2 x RJ45 connectors (1Gb/s Ethernet Port) allowing the unit to operate in redundant or switched modes.
• PoE and link LED status indicators for each ethernet port.
• Front panel clock, PoE and link LED status indicators.
• Front panel AES3 lock LED status indicators for each input.
• Powered by PoE on either (or both) ethernet ports, offering power supply redundancy.
• Alternative USB-C DC power input, when PoE is unavailable.
• Configuration using Dante Controller.
• Dante Domain Manager compliant.
• AES67 operation.
• 1U 19″ rack-mount aluminium chassis.

Designed for broadcast, live production, and professional AV environments, the AVN-AO16-T-DNT provides a cost-effective and dependable solution for transporting low-latency audio between analogue and Dante network domains. Product Features

• Dante AoIP.
• 16 x balanced analogue outputs on terminal blocks.
• Adjustable global 0dBFS line up selection (+12dBu/+18dBu/+24dBu).
• Powered by PoE on either (or both) Ethernet ports, offering power supply redundancy.
• Alternative USB-C DC power input, when PoE is unavailable.
• 2 x RJ45 connector, (1Gb/s Ethernet Port), allowing the unit to operate in redundant or switched modes.
• PoE and Link LED status indicators for each Ethernet port.
• Rear-panel mount for single/top locking USB-C cables.
• Rear-panel cable tie mount to provide strain relief.
• Clock LED status indicator.
• Configuration using Dante Controller.
• AES67 operation & Dante Domain Manager compliant.
• 1U 19″ rack-mount aluminium chassis.

Designed for broadcast, live production, and professional AV environments, the AVN-AO16-DNT provides a cost-effective and dependable solution for transporting low-latency audio between analogue and Dante network domains. Product Features

• Dante AoIP.
• 16 x balanced analogue outputs on Neutrik XLR.
• Adjustable global 0dBFS line up selection (+12dBu/+18dBu/+24dBu).
• Powered by PoE on either (or both) Ethernet ports, offering power supply redundancy.
• Alternative USB-C DC power input, when PoE is unavailable.
• 2 x RJ45 connector, (1Gb/s Ethernet Port), allowing the unit to operate in redundant or switched modes.
• PoE and Link LED status indicators for each Ethernet port.
• Rear-panel mount for single/top locking USB-C cables.
• Rear-panel cable tie mount to provide strain relief.
• Clock LED status indicator.
• Configuration using Dante Controller.
• AES67 operation & Dante Domain Manager compliant.
• 1U 19″ rack-mount aluminium chassis.

Designed for broadcast, live production, and professional AV environments, the AVN-AI16-T-DNT provides a cost-effective and dependable solution for transporting low-latency audio between analogue and Dante network domains.

Product Features

• Dante AoIP.
• 16 x analogue inputs on terminal blocks.
• Accepts balanced and un-balanced analogue line signal.
• Adjustable global 0dBFS line up selection (+12dBu/+18dBu/+24dBu).
• Powered by PoE on either (or both) Ethernet ports, offering power supply redundancy.
• Alternative USB-C DC power input, when PoE is unavailable.
• 2 x RJ45 connector, (1Gb/s Ethernet Port), allowing the unit to operate in redundant or switched modes.
• PoE and Link LED status indicators for each Ethernet port.
• Rear-panel mount for single/top locking USB-C cables.
• Rear-panel cable tie mount to provide strain relief.
• Clock LED status indicator.
• Configuration using Dante Controller.
• AES67 operation & Dante Domain Manager compliant.
• 1U 19″ rack-mount aluminium chassis.

Designed for broadcast, live production, and professional AV environments, the AVN-AI16-DNT provides a cost-effective and dependable solution for transporting low-latency audio between analogue and Dante network domains.

Product Features

• Dante AoIP.
• 16 x analogue inputs on Neutrik XLR.
• Accepts balanced and un-balanced analogue line signal.
• Adjustable global 0dBFS line up selection (+12dBu/+18dBu/+24dBu).
• Powered by PoE on either (or both) Ethernet ports, offering power supply redundancy.
• Alternative USB-C DC power input, when PoE is unavailable.
• 2 x RJ45 connector, (1Gb/s Ethernet Port), allowing the unit to operate in redundant or switched modes.
• PoE and Link LED status indicators for each Ethernet port.
• Rear-panel mount for single/top locking USB-C cables.
• Rear-panel cable tie mount to provide strain relief.
• Clock LED status indicator.
• Configuration using Dante Controller.
• AES67 operation & Dante Domain Manager compliant.
• 1U 19″ rack-mount aluminium chassis.

Designed for broadcast, live production, and professional AV environments, the AVN-AIO8-T-DNT provides a cost-effective and dependable solution for transporting low-latency audio between analogue and Dante network domains.

Product Features

• Dante AoIP.
• 8 x analogue inputs on terminal blocks.
• 8 x analogue outputs on terminal blocks.
• Accepts balanced and un-balanced analogue line signal.
• Adjustable global 0dBFS line up selection (+12dBu/+18dBu/+24dBu).
• Powered by PoE on either (or both) Ethernet ports, offering power supply redundancy.
• Alternative USB-C DC power input, when PoE is unavailable.
• 2 x RJ45 connector, (1Gb/s Ethernet Port), allowing the unit to operate in redundant or switched modes.
• PoE and Link LED status indicators for each Ethernet port.
• Rear-panel mount for single/top locking USB-C cables.
• Rear-panel cable tie mount to provide strain relief.
• Clock LED status indicator.
• Configuration using Dante Controller.
• AES67 operation & Dante Domain Manager compliant.
• 1U 19″ rack-mount aluminium chassis.

Designed for broadcast, live production, and professional AV environments, the AVN-AIO8-DNT provides a cost-effective and dependable solution for transporting low-latency audio between analogue and Dante network domains.

• Dante AoIP.
• 8 x analogue inputs on Neutrik XLR.
• 8 x analogue outputs on Neutrik XLR.
• Accepts balanced and un-balanced analogue line signal.
• Adjustable global 0dBFS line up selection (+12dBu/+18dBu/+24dBu).
• Powered by PoE on either (or both) Ethernet ports, offering power supply redundancy.
• Alternative USB-C DC power input, when PoE is unavailable.
• 2 x RJ45 connector, (1Gb/s Ethernet Port), allowing the unit to operate in redundant or switched modes.
• PoE and Link LED status indicators for each Ethernet port.
• Rear-panel mount for single/top locking USB-C cables.
• Rear-panel cable tie mount to provide strain relief.
• Clock LED status indicator.
• Configuration using Dante Controller.
• AES67 operation & Dante Domain Manager compliant.
• 1U 19″ rack-mount aluminium chassis.

Catchbox Hub is equipped with advanced DSP features. Serving as the central point in education spaces where complexity is growing but budgets and human resources are not, the Catchbox Hub brings everything together in one compact, powerful processing device. Key features of the Hub:

• Separate DSP paths for a smooth in-room and video conferencing audio experience, with smart features like feedback suppression, autoleveling, echo reduction, and more.
• Flexible input options, supporting up to 4 Catchbox wireless microphones, 4 analog sources, and 4x4 Dante™ channels for mixing and processing external sources like third-party mic systems.
• Custom-configurable output mixes for assistive listening, recording, and conferencing.
• USB integration for seamless connection with software platforms like Zoom, MS Teams, or lecture capture tools.
• Centralized control and monitoring software for campus-wide AV scalability (coming soon).

The Catchbox Plus is a wireless microphone solution designed to provide professional-grade audio for dynamic, unsupervised environments such as lecture halls, hybrid meetings, and collaborative spaces. At the core of this system is the Hub, which integrates powerful DSP technology to elevate the in-person and far-end participant audio experience. The system includes the Cube, Clip, and Stick microphones, each designed to cater to different speaking styles and environments. To ensure every voice sounds perfect, the Hub is equipped with advanced DSP features. The Automixer detects the active speaker and focuses the microphone on them, reducing background noise and keeping conversations sharp and engaging. Feedback Suppression reduces feedback during voice-life by continuously changing the pitch of the amplified sound. Features like Dante® audio over IP integration and remote control and monitoring ensure smooth maintenance for space managers.

Multiple audio outputs

Adopt 6.3mm unbalanced output, XLR balanced output, Dante network audio output and USB digital audio output, it is convenient to connect various audio processing equipment.

DSP audio processing algorithm

Built-in high and low pass filters, parametric equalizer, feedback suppression, volume detection, audio matrix, volume control and other functions.

Alarm trigger input port

When the alarm port receives a high-level signal, the alarm function is activated, automatically muting all output audio until the alarm state is cleared.

Public address system input

For constant voltage power amplifier input, use the switch to select the input voltage level: 0V, 25V, 70V, 100V. The public address input also features a priority function. When an audio signal is input to the PA interface, other audio sources will be automatically muted. Once the public address audio signal ends, the mute will be canceled.

Features

Dante network audio protocol

Supports the Dante network audio protocol, which has the characteristics of low latency, high precision, low cost, and uncompressed audio signals.

FBK Feedback Suppression Function

It has the domestically pioneered FBK function, which can automatically and effectively prevent howling, enabling the singer to achieve a perfect performance.

Magic EQ audio software

The connection software has a built-in digital equalization processor, with 9 tone modes to choose from, 7 of which are fixed modes, covering various international brand tone effects, and 2 custom modes. The tone can be adjusted freely through dedicated operating software to achieve individual needs.

Optional dual-bay chargers

Optional contact-type dual bay changer, simply place the microphone into it for charging. Utilizing rechargeable NiMH batteries, it offers a longer lifespan compared to alkaline batteries, ensuring enhanced safety, durability, and environmental friendliness. This convenient charging method is suitable for various scenarios requiring frequent charging.

Features

Dante audio transmission protocol

Equipped with a Dante interface, it can communicate and transmit audio with any device equipped with Dante.

Triangular tube gooseneck microphone

The microphone adopts a small triangular shape gooseneck design, which has a simple and elegant appearance, and can easily and flexibly adjust the pickup capsule to the required position.

Zinc alloy conference base

The conference base is made of strong and durable zinc alloy, which is artistic and sturdy. Equipped with mute button and working indicator light, it is easy to grasp the status of the equipment.

Light - touch switch troubleshooting

It is equipped with a unique light - touch switch troubleshooting function. After pressing the button, the LED will flash and the location of the faulty microphone can be quickly found, which greatly reduces the difficulty of subsequent maintenance work.

Supports TDM digital control software

HR-32S Single-Channel True Diversity Wireless Microphone System The HR-32S is a single-channel wireless microphone system integrating a spectrum analyzer and wireless audio transmission. The spectrum analyzer serves to visualize the system's frequencies, ensuring optimal controllability and stability in any performance setting, and delivering pristine, immersive sound quality. Features: Digital Hybrid Dante Wireless System Revolutionary design that combines digital audio with an analog FM radio link to provide both outstanding audio quality and exemplary, noise-free RF performance. WireMicX Wireless Management System WireMicX Wireless System Management Software, can be connected to PC network to monitor and search for radio wave interference points in the actual environment of the scene and frequency dead spots on the stage. Signal Frequency Visualization The radio frequency details of the current environment can be monitored in real time on the display screen of the receiver, and the interference-free frequency can be viewed and selected through the screen. DSP Audio Digital Extension High-speed DSP chip, 24-bit high-performance ADC, DAC and self-developed digital audio extension algorithm are used to reduce the noise caused by wireless transmission. PLL Frequency Synthesis Oscillation Circuit The stable PLL phase-locked oscillation circuit, combined with the"noise detection" mute control function, can effectively block the noisy radio frequency interference in the working environment. Digital Audio Output HR series provides two digital audio outputs – for DANTE/AES67

Specifically designed for large-scale stage performances, speeches, and various major sports and cultural events. Features a full-color LCD dual-screen (master/sub) interface design, enabling quick operation and user-defined shortcut keys. Pilot squelch technology eliminates RF interference when the transmitter is turned off. Offers interference-free frequency settings and switching, allowing multiple units to operate simultaneously without mutual interference. Utilizes automatic diversity reception to eliminate dropouts, extend receiving range, and enhance stability; operates in the UHF band. PLL phase-locked loop frequency synthesis technology. Infrared auto-sync technology for automatic and rapid frequency matching with transmitters. Auto Scan function for quick searching of interference-free channels. Dual-channel true diversity wireless microphone system with software-adjustable timbre mode functionality. Built-in Dante module, supporting the Dante networked audio protocol. Both handheld and bodypack transmitters feature LCD color displays.

Specifically designed for large-scale stage performances, speeches, and various major sports and cultural events. Features a full-color LCD dual-screen (master/sub) interface design, enabling quick operation and user-defined shortcut keys. Pilot squelch technology eliminates RF interference when the transmitter is turned off. Offers interference-free frequency settings and switching, allowing multiple units to operate simultaneously without mutual interference. Utilizes automatic diversity reception to eliminate dropouts, extend receiving range, and enhance stability; operates in the UHF band. PLL phase-locked loop frequency synthesis technology. Infrared auto-sync technology for automatic and rapid frequency matching with transmitters. Auto Scan function for quick searching of interference-free channels. Dual-channel true diversity wireless microphone system with software-adjustable timbre mode functionality. Built-in Dante module, supporting the Dante networked audio protocol. Both handheld and bodypack transmitters feature LCD color displays.

LEA Professional's Connect Series Half-Rack two-channel smart amplifiers are designed to deliver maximum flexibility in a compact form factor. The CS122D comes Dante enabled, pushes 120W per channel, and features built-in DSP, WebUI, SharkWare, LEA Cloud, third-party API integration, and access to more than 3,000 speaker tunings. The CS122D also supports Hi-Z (70V/100V) and Lo-Z operation on a per-channel basis, eliminating the need for separate amplifiers while simplifying system design and reducing installation costs. Plus, all required mounting hardware is included for fast, hassle-free installation.

The Connect Series Glass wall controller expands the LEA product lineup with a sleek, control solution for modern commercial and residential audio installations. The controller features integrated Bluetooth® connectivity and built-in music streaming capabilities, while delivering simple, flexible control for integrators and end users across a wide range of applications. Engineered to seamlessly integrate with Dante-enabled Connect Series amplifiers and the full suite of LEA software, including WebUI, SharkWare, and LEA Cloud, the PoE-powered controller allows users to easily manage volume, source selection, and zone control without the need for additional devices or complex programming. Featuring Bluetooth pairing and direct streaming, this controller lets users quickly connect and play audio from personal devices or streaming platforms, ensuring added convenience and reliable performance.

The Connect Series Glass wall controller expands the LEA product lineup with a sleek, control solution for modern commercial and residential audio installations. The controller features integrated Bluetooth® connectivity and built-in music streaming capabilities, while delivering simple, flexible control for integrators and end users across a wide range of applications. Engineered to seamlessly integrate with Dante-enabled Connect Series amplifiers and the full suite of LEA software, including WebUI, SharkWare, and LEA Cloud, the PoE-powered controller allows users to easily manage volume, source selection, and zone control without the need for additional devices or complex programming. Featuring Bluetooth pairing and direct streaming, this controller lets users quickly connect and play audio from personal devices or streaming platforms, ensuring added convenience and reliable performance.

The DCS series call stations utilize the latest network audio technology and support three-layer network transmission. They can serve as standard call stations for routine operational announcements or be used for emergency evacuation broadcasts following sudden emergencies. Depending on usage habits and the specific location, users can choose between the gooseneck microphone (DCS-CSG) or the handheld microphone (DCS-CSH). All buttons on the call station can be freely configured and programmed to meet the user’s specific needs. In addition to basic broadcasting functions, this call station supports secondary development of industry-specific broadcasting applications through interactive interfaces with dynamic operational databases across various sectors. For example, it can integrate with systems such as the Airport Operations Database (AODB) and the Integrated Business Exchange Platform (IMB) to enable dynamic automated broadcasting, personalized voice announcements for airlines, and the display of passenger flow status and other information.

The DCS series call stations utilize the latest network audio technology and support three-layer network transmission. They can serve as standard call stations for routine operational announcements or be used for emergency evacuation broadcasts following sudden emergencies. Depending on usage habits and the specific location, users can choose between the gooseneck microphone (DCS-CSG) or the handheld microphone (DCS-CSH). All buttons on the call station can be freely configured and programmed to meet the user’s specific needs. In addition to basic broadcasting functions, this call station supports secondary development of industry-specific broadcasting applications through interactive interfaces with dynamic operational databases across various sectors. For example, it can integrate with systems such as the Airport Operations Database (AODB) and the Integrated Business Exchange Platform (IMB) to enable dynamic automated broadcasting, personalized voice announcements for airlines, and the display of passenger flow status and other information.  

The DCS-AMP12004 is a high-performance power amplifier that combines Class-D amplification technology with an SMPS switched-mode power supply and is designed for large-scale audio system applications. Its compact size, low power consumption, and high efficiency deliver reliable performance for demanding installations. Intelligent power sharing further enhances system flexibility by dynamically allocating available power across channels for maximum utilisation. One of the key strengths of the DCS-AMP12004 is its intelligent power sharing capability. A total output of 1200 W can be flexibly distributed across four channels, with up to 600 W available per channel. This allows the system to adjust channel output dynamically according to actual load requirements, avoiding the inefficiencies associated with conventional fixed-power designs. It is particularly well suited to complex applications such as public address systems, multi-zone music systems, airports, metro systems, nuclear power plants, and sports venues.