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.
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.
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.
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.
The new Catchbox Hub is the latest innovation within the Catchbox Plus wireless microphone system and was designed to be the central point in education spaces where audio complexity is growing — but budgets and human resources are not. With support for up to four Catchbox wireless mics and up to ten external audio sources over analog, USB and Dante, the Hub mixes, processes and routes audio for in-room amplification, conferencing, recording and assistive listening. The Catchbox Hub eliminates the need for an external DSP, says Catchbox, allowing AV designers to simplify their setup and reduce costs.
Key features include the following:
Separate DSP paths for a smooth in-room and video conferencing audio experience, with smart features like feedback suppression, auto-leveling, echo reduction and more.
Flexible input options, supporting up to 4 Catchbox wireless microphones, 4 analog sources and 4×4 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).
Equally suited to FOH duties as a highly capable system manager and console switcher, or as the hub of a complex installation system, the DM88 can control and process it all, simply and capably. The DM88 incorporates not only the best quality audio conversion and DSP, but also new levels of flexibility for routing and processing. Featuring 16 Dante, 8 analogue and 8 AES inputs, these are all available to be matrixed to one of 16 processing channels, optimised for either general input processing or for more specialised output speaker processing.
Supporting an array of interfaces, this multi-channel processor series provides essential video production features in a compact 1RU enclosure. Besides offering core 2K/4K frame sync functionality, the FA-1616 series serves as a color corrector, video processing amp, audio remapper, and much more. For an optimal configuration, use optional software to add your preferred functions to both processor blocks. You can also customize your system economically with a range of expansion cards that add audio (including AES/EBU, Dante, and MADI) and GPI interfaces. Add an MoIP card to convert between IP and baseband formats or between IP formats (ST 2022-6* ↔ ST 2110), enabling use as an MoIP/SDI gateway.
Built on the tried and tested Yamaha DME platform, the NEXO DME10 goes a step further than any other immersive processor by unlocking its raw processing power to users and allowing near limitless customisation.
A 256 x 256 channel open architecture processor , DME10 ships with 32 object inputs / 16 speaker outputs included as standard, expandable in blocks of 32 x 16, 64 x 32 and 128 x 64 via a flexible licensing structure. Need the full channel count for a show today but two completely independent 64 channel immersive processors tomorrow? You can do that .
With superior musical expression and sound field design, DME10 features 96kHz processing and is equipped with a wide range of DSP components that can be freely selected, connected, and configured, allowing for flexible system construction.
And such is the processing power of the DME10, separate speaker system and immersive processors are no longer required. It’s all inside DME10.
Performance ModesMicflex offers seven different modes for multiple mix scenarios:
Ensemble: Increases overall gain before feedback by up to 10dB for choirs, orchestras, and other group performances.
Duet: Increases gain by up to 10dB for two separate choir, orchestra, or other group sources performing at the same time, doubling the power of Ensemble mode.
Audience: Enhance your Broadcast mixes and In-Ear Monitor mixes with increased isolation of your audience mics from PA bleed and room noise.
Stage: Increases gain by up to 10dB for as many as eight independently processed microphones mixed into a house PA.
Wedge: Increases gain by up to 10dB for as many as eight independently processed microphones mixed into eight separate floor wedges.
Lecture: Increases gain by up to 10dB for spoken word in four separate instances – Lecture rooms, podiums, classrooms, lavalier microphones or headset microphones.
Sports: Increase isolation of referee speech or a sideline microphones from the PA sound in arenas and stadiums, enhancing your live and broadcast mixes.
Choirflex is a proprietary, adaptive digital audio processor designed to increase the clarity and stability of choir mixes and greatly reduce background noise, including feedback. Choirflex intelligently learns the venue’s acoustical environment and provides processed choir audio that results in much improved gain before feedback and increased clarity.
Key Features
Greater Acoustical Gain: With up to 10dB more gain-before-feedback, easily get your choir on top of the mix
Adapts to Your Environment: Listens and intelligently adapts to your mix environment
Increased Clarity & Stability: Vocal intelligibility of every section in the choir, from bass to soprano
Maintains Natural Tonality: Clean, natural tone of your choir’s vocal blend
Designed and built in the United States, the SA-8 DSP is a compact 2RU amplifier that delivers eight channels of high-performance amplification at 200W RMS per channel into eight ohms and 400W per channel into four ohms, all channel-driven. This DSP model adds a DSP board, enabling all eight channels to provide the necessary digital active crossover, delay, gain, and digital EQ profiles, in addition to a Dante Digital Audio Input, and FIR Filters, for the ultimate in sound and loudspeaker setup and EQ. The high efficiency of the SA-series design also means that the amplifiers can be placed near each other, as in an equipment rack, without undue overheating.
An open architecture-type digital signal processor, which offers state-of-the-art performance and connectivity, 96 kHz sound quality and support for up to 256 input/output channels.*
* DEK-DME7-DX64, the license to expand I/O channels and matrix mixer, and ProVisionaire Cloud service will be provided in the future.
The ALF-DSP44-UD is an 4 in / 4 out Audio Digital Signal Processor with Advanced audio processing capabilities and features 4 analogue inputs and 4 analogue outputs, 4 Dante inputs and 4 Dante outputs, 2 Channels of AEC processing, USB Audio, RS-232, RS-485, and IP control.
The ALF-DSP1616-UD is an 16 in / 16 out Audio Digital Signal Processor with DANTE, Advanced audio processing capabilities and features 16 analogue inputs and 16 analogue outputs, 16 DANTE inputs and 16 DANTE Outputs, USB Audio, 8 General Purpose Input / Output (GPIO) connections, RS-232, RS-485, and IP control.
The ALF-DSP88-UD is an 8 in / 8 out Audio Digital Signal Processor with Advanced audio processing capabilities and features 8 analogue inputs and 8 analogue outputs, 8 Dante inputs and 8 Dante outputs, Single Channel AEC processing, USB Audio, 8 General Purpose Input / Output (GPIO) connections, RS-232, RS-485, and IP control.
Hybrid support for both web and real life. Compact and highly functional WR-DX200 series
Suitable for small to medium-sized meeting rooms, as well as large conference rooms such as those in convention centers
The RAMSA WR-DX200/WR-DX200DAN solves audio problems such as echo, noise, and feedback during hybrid conferences. It can also control video equipment such as projectors and flat panels, connecting remote spaces to create the ideal meeting/lecture format.
Multi-channel processor series with essential video production features in a compact 1 RU enclosure. Besides its core role as a frame synchronizer for up to 32 channels, supporting IP/SDI (12G/6G/3G/HD-SDI) and 4K, the FA-1616 series also serves as a color corrector, video processing amp, and audio remapper.
Multi-channel processor series with essential video production features in a compact 1 RU enclosure. Besides its core role as a frame synchronizer for up to 32 channels, supporting IP/SDI (12G/6G/3G/HD-SDI) and 4K, the FA-1616 series also serves as a color corrector, video processing amp, and audio remapper.
Multi-channel processor series with essential video production features in a compact 1 RU enclosure. Besides its core role as a frame synchronizer for up to 32 channels, supporting IP/SDI (12G/6G/3G/HD-SDI) and 4K, the FA-1616 series also serves as a color corrector, video processing amp, and audio remapper.
The Q-SYS Core 6000 CXR is an industry-first solution that combines the processing capabilities of the Q-SYS realtime operating system with the compact, ruggedized, MIL STD, NEBS and Marinecompliant Dell PowerEdge XR11 server
The Q-SYS Core 610 represents the next-generation of Q-SYS processing, pairing the Q-SYS OS with enterprise-grade Dell COTS server hardware to deliver a flexible and scalable audio, video and control solution for a vast range of larger scale applications. It is a fully networked AV&C processor, letting you distribute network I/O where it’s most convenient.
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.
