Dante-enabled Product Catalog

VIVO-X8D8 is an EclerPLEX Host matrix with 8 analogue and 8 Dante®/AES67 audio input ports, 8 analogue and 8 Dante®/AES67 audio output ports, scalable thanks to two X-link ports. Each X-link port allows the connection of one BIO Series expansion unit. VIVO-X8D8, performing as a Host unit in an EclerPLEX deployment, can contain multiple projects running simultaneously and serve user control panels to end users.

---

Description VIVO-XØD32 is an EclerPLEX Host matrix featuring 32 Dante®/AES67 audio input ports and 32 Dante®/AES67 audio output ports. It is scalable via its X-link port, which allows the connection of a BIO Series expansion unit. As a Host unit within an EclerPLEX deployment, it can run multiple projects simultaneously and provide user control panels for end users.

The AQZ32 integrates AquaControl™ with Dante® Embedded technology, significantly enhancing its I/O capabilities making it a versatile addition to any audio setup. Powered by Ashly’s next-generation AquaControl™ DSP engine, the AQZ32 offers an intuitive, web-browser based interface that requires no specialized training, getting you up and running quicker and smoother. It includes 16 analog mic/line inputs each with software controlled 48V Phantom Power and mic gain settings up to 66dB. It also has 16 analog balanced line outputs. Dante inputs 1-16 can be independently selected instead of each corresponding analog input number. So, you can choose if you need processed analog or Dante sources. Dante inputs 17-32 are available on each mixer to expand the mix with these Dante sources. Dante outputs 1-32 allow you to route processed inputs and outputs to amps and other devices. A pair of AQZ32s can be setup to provide a 32x32 audio system.

HALO-4B150 is a 4-channel Class D amplifier with networked DSP, delivering 4 × 150 W in low-impedance mode at 4 Ω or 8 Ω. Its four output channels can be bridged to drive high-impedance 70 V or 100 V loudspeaker lines. It features 4 balanced analogue inputs, 4 Dante®/AES67 digital inputs and outputs, 2 balanced auxiliary outputs, and a flexible 4 × 6 digital matrix. Its comprehensive DSP includes input and output parametric equalization, graphic output equalization, crossovers, delay, and configurable peak and RMS limiters. The built-in web application provides intuitive configuration, control and monitoring of the amplifier. Two Ethernet ports with integrated switch functionality allow network daisy-chaining and provide access to control, Dante®/AES67 audio transport and remote management. HALO-4B150 also supports customizable user control panels and integration with EclerCLOUD and the PULSO digital control panel series.

HALO-2B300 is a 2-channel Class D amplifier with networked DSP, delivering 2 × 300 W in low-impedance mode at 4 Ω or 8 Ω. Its two output channels can be bridged to drive high-impedance 70 V or 100 V loudspeaker lines. It features 4 balanced analogue inputs, 4 Dante®/AES67 digital inputs and outputs, 2 balanced auxiliary outputs, and a flexible 4 × 4 digital matrix. Its comprehensive DSP includes input and output parametric equalization, graphic output equalization, crossovers, delay, and configurable peak and RMS limiters. The built-in web application provides intuitive configuration, control and monitoring of the amplifier. Two Ethernet ports with integrated switch functionality allow network daisy-chaining and provide access to control, Dante®/AES67 audio transport and remote management. HALO-2B300 also supports customizable user control panels and integration with EclerCLOUD and the PULSO digital control panel series.

HALO-4B300 is a 4-channel Class D amplifier with networked DSP, delivering 4 × 300 W in low-impedance mode at 4 Ω or 8 Ω. Its four output channels can be bridged to drive high-impedance 70 V or 100 V loudspeaker lines. It features 4 balanced analogue inputs, 4 Dante®/AES67 digital inputs and outputs, 2 balanced auxiliary outputs, and a flexible 4 × 6 digital matrix. Its comprehensive DSP includes input and output parametric equalization, graphic output equalization, crossovers, delay, and configurable peak and RMS limiters. The built-in web application provides intuitive configuration, control and monitoring of the amplifier. Two Ethernet ports with integrated switch functionality allow network daisy-chaining and provide access to control, Dante®/AES67 audio transport and remote management. HALO-4B300 also supports customizable user control panels and integration with EclerCLOUD and the PULSO digital control panel series.

T88D is a powerful 8-in 8-out audio matrix that supports 8 analog audio inputs and 8 analog audio outputs. T88D also supports 4×4 Dante network audio input and output. Multiple options of input and output provide you with the flexibility required to integrate into the system. It can meet the control and audio processing for various places. Through the Windows system computer software BrainCore Net, you can remotely control all devices in the local area network and efficiently control all parameters of the processor. Through the joint debugging function, multiple devices of the same model can be debugged simultaneously. Some functions can also be controlled through the wall panel. Through the scene preset function, users can directly use intelligent central control to achieve scene recall, volume adjustment and other functions.

P48D with its 96kHz sampling rate and 24-bit precision, combined with IIR and FIR filtering technologies, accurately generates linear phase curves, significantly improving the quality of the impulse response. This combination of technologies meets the requirements of an ideal crossover, enabling sound engineers to precisely reproduce the perfect sound quality of speakers in live settings, providing users with a balanced, natural, transparent, and authentic auditory experience. AES Digital Audio Input The P series supports standard AES/EBU digital audio input, providing customers with a variety of widely recognized digital audio input solutions in the industry. This enables 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.

D4.10K-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.

D4.4K-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.

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.

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.

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.

The DCS-ITG is a high-performance access gateway for modern VoIP intercom and telephone systems, designed to convert VoIP audio efficiently into Dante audio. The device supports the internationally standard SIP protocol and uses G.722 or G.711 codecs to connect voice information from intercom or telephone systems to the public address system quickly and accurately, enabling smooth information delivery. The DCS-ITG supports up to 8 concurrent VoIP audio channels, maintaining stable transmission and audio quality even in high-traffic environments to meet information delivery requirements in both routine office use and emergency situations.

SoundCoreHero Ultimate is a flagship-grade audio engine combining a player, DSP processor, and automation controller for the most demanding Dante and AES67 Audio-over-IP (AoIP) infrastructures. It is purpose-built for mission-critical, large-scale installations that demand maximum processing power, scalability, and operational reliability. Supporting between 128 and 512 bi-directional channels, SoundCoreHero Ultimate offers exceptional signal density and independent zone management. Its dual redundant power supplies, hardware-based clock synchronization, and robust system architecture ensure uncompromising uptime. With its integrated web-based management interface and support for open control standards, SoundCoreHero Ultimate delivers complete oversight and confidence for professional environments where audio performance is non-negotiable.

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.

Advanced Audio Processing Technologies: The audio processor integrates a range of advanced audio algorithms, including multi-channel independent Acoustic Echo Cancellation (AEC), Adaptive Feedback Cancellation, Crossover, FIR filters, and more. These features are tailored to meet the needs of different application scenarios, ensuring high-quality audio processing and stable transmission. Customizable Analog Channels: Our device is designed with 8 customizable analog channels, allowing users to define each of them as either input or output based on actual needs. We provide open access for users to freely configure any of these 8 channels, making the processor’s channel management more flexible and convenient. High-Quality Audio Transmission: The processor supports up to 12 analog input and 8 analog output channels, along with 64 digital input and 64 digital output channels. It is integrated with the Dante audio protocol, enabling high-quality multi-channel audio transmission and processing, delivering a clear and stable audio experience.

The Flex HTn brings networked audio capability to our popular pocket‑sized home theater processor. By integrating Dante Audio‑over‑IP (AoIP) with HDMI ARC/eARC support, it delivers exceptional versatility for modern multichannel systems. It accepts multichannel audio through HDMI eARC or USB Audio, with additional stereo inputs available via SPDIF and TOSLINK optical. Inside, you’ll find the complete suite of miniDSP’s flexible routing and processing tools: bass management, parametric EQ, configurable crossovers, advanced biquad programming, and precise delay and gain controls. The Flex HTn also supports optional upgrades such as full‑bandwidth Dirac Live®, the industry‑leading room correction platform.

Its four analog RCA outputs offer outstanding low‑noise, low‑distortion performance. With Dante/AES67 compatibility, the Flex HTn can stream multi-channel digital audio over IP from any input source—USB, HDMI, TOSLINK, or SPDIF—directly to networked speakers or amplifiers. An OLED front display and encoder/volume knob provide intuitive local control.

The Flex EIGHT-N joins the miniDSP Flex family as a Dante‑capable audio processor that natively streams Dante/AES67 audio. It delivers eight Dante/AES67 AoIP channels plus four analog outputs for seamless, low‑latency multi-channel distribution within any Dante ecosystem.

Featuring multiple digital inputs, eight digital outputs and four analog outputs, the Flex EIGHT-N can drive active speakers up or be set up as a multi-zone processor via our flexible routing. Every output has miniDSP’s full processing suite: parametric EQ, IIR and FIR filters, configurable crossovers, advanced biquad programming and precise delay.

Digital inputs include USB, S/PDIF (RCA and optical) and Bluetooth (LDAC/aptX). An OLED front panel and encoder/volume knob provide local control, and firmware supports upgrades such as Dirac Live room correction.

In short, the Flex 8N is a compact, Dante‑ready powerhouse for modern active speaker and subwoofer systems, offering direct network streaming and sophisticated loudspeaker processing.

It has 4 analog inputs and 4 analog outputs, as well as 4 Dante inputs and 4 Dante outputs, recommended for medium-sized meeting rooms or classrooms.

It has 8 analog inputs and 8 analog outputs, as well as 8 Dante inputs and 8 Dante outputs, making it suitable for independent sound reinforcement systems as well as medium to large scale remote conferencing and local amplification scenarios.

Step into a new era of spatial sound with the TiMax SoundHub, a true “show-in-a-box” solution that combines immersive audio spatialisation, localisation, system processing, playback, and show control in one compact, high-performance unit. TiMax

• Unrivalled audio immersion: use object-based 3D mixing to place sound naturally across your venue, so every seat hears a full, authentic spatial mix.
• Powered by real-time FPGA cores:  support for 32×32 or 64×64 I/O (Dante or MADI) plus optional analog/AES modules.
• Cueing, automation & show control built in:  timeline sequencing, MIDI/OSC/UDP integration, and advanced show control means you can run your show with or without an external PC. TiMax
• Flexible spatial mapping & scaling: map any venue in 3D, define “image points,” scale shows to different room sizes (even live), and dynamically morph sound movement across speakers. TiMax
• Reliable & integrated:  industrial rack design comes fitted with double PSU and Fan, always-on readiness, and minimal external dependencies.

Whether for theatre, concert halls, immersive installations, or experiential events, the TiMax SoundHub gives sound designers and engineers full control over location, movement, and intelligence of audio — making sound follow the story.

Yealink AP08 DSP, a cutting-edge digital signal processor, elevates your meeting experience with exceptional automation capabilities. It supports up to 16x16 Dante audio channels, including 8 optimized channels with dynamic AEC, intelligent noise cancellation, and reverberation suppression, as well as 4x4 analog audio channels and two simultaneous USB 3.0 Type-B device interfaces. The AP08 DSP ensures smooth audio transmission and crystal-clear communication. Additionally, it provides PoE+ power for up to three audio devices, simplifying your deployment even further. With Yealink Room Designer software, the AP08 DSP seamlessly integrates automated design and acoustic environment-adaptive technology, setting a new standard for audio quality and convenience in your meetings. Choose Yealink AP08 DSP for an unparalleled audio experience.

The BSS Soundweb OMNI 512p Open Architecture Digital Signal Processor is the most powerful enterprise-grade processor ever developed by BSS. With processing capability that surpasses previous BSS products, 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 512p features 512x512 at up to 96kHz Dante/AES67 networked audio channels; enterprise-grade network security support; an onboard scripting engine developed by AMX; 8 GPIO ports (fully flexible); 160 AEC channels; 64 VoIP lines; high precision word clock and BNC clock (in/out); dual redundant power supplies, Dante and HControl; 128 channel internal media player with 1TB SSD; full-color, interactive front panel LCD; and more. Every GPIO port is format (analog or digital) and directionally (input or output) configurable. Any channel can also be configured to function as a digital binary, to provide 0 or 5 volts, or configured to receive or send any voltage between 0 and 12 volts. The Soundweb OMNI family also includes two I/O Expanders, the 32e and 16e, which provide unparalleled I/O flexibility as needed. The integrated 128-channel internal media player facilitates the playback of pre-recorded audio files directly from an internal 1TB SSD. This is useful for announcements, paging chimes, advertisements, background soundscapes, and more. Supporting up to 160 AEC channels and 64 VoIP lines, the 512p provides seamless integration capability with building phone systems to ensure efficient and reliable communication. The onboard scripting engine allows previous generation BSS Soundweb Contrio EC wall controller devices to control new Soundweb OMNI devices, while also allowing new AVX software to integrate previous generation BSS Soundweb London DSPs and Crown DCi amplifiers into a unified custom UI, directly alongside Soundweb OMNI. 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 256p Open Architecture Digital Signal Processor is the most powerful enterprise-grade processor ever developed by BSS. With processing capability that surpasses previous BSS products, 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 256p features 256x256 at up to 96kHz Dante/AES67 networked audio channels; enterprise-grade network security support; an onboard scripting engine developed by AMX; 24/48kHz and 24/96kHz supported bit depth/sample rates; 8 GPIO ports (fully flexible); 80 AEC channels; 32 VoIP lines; high precision word clock and BNC clock (in/out); dual redundant power supplies, Dante and HControl; 128 channel internal media player with 500GB SSD; full-color, interactive front panel LCD; and more. Every GPIO port is format (analog or digital) and directionally (input or output) configurable. Any channel can also be configured to function as a digital binary, to provide 0 or 5 volts, or configured to receive or send any voltage between 0 and 12 volts. The Soundweb OMNI family also includes two I/O Expanders, the 32e and 16e, which provide unparalleled I/O flexibility as needed. The integrated 128 channel internal media player facilitates the playback of pre-recorded audio files directly from an internal 500GB SSD. This is useful for announcements, paging chimes, advertisements, background soundscapes, and more. Supporting up to 80 AEC channels and 32 VoIP lines, the 256p provides seamless integration capability with building phone systems to ensure efficient and reliable communication. The onboard scripting engine allows previous generation BSS Soundweb Contrio EC wall controller devices to control new Soundweb OMNI devices, while also allowing new AVX software to integrate previous generation BSS Soundweb London DSPs and Crown DCi amplifiers into a unified custom UI, directly alongside Soundweb OMNI. 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.