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.
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.
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.
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.
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-TR Networked Audio Transceiver provides 8 audio inputs and 8 audio outputs, together with high-precision A/D and D/A conversion modules. It enables high-fidelity conversion between analogue and digital signals while maintaining very low noise throughout the conversion process, meeting the demands of professional applications such as broadcasting, conferencing, and live performance.
The DCS-TR also adopts a modular design and supports device cascading and system expansion, allowing users to build applications ranging from small local systems to large-scale networked audio systems with considerable flexibility.
Leveraging the mature Dante platform, the device delivers efficient audio routing and data transport while maintaining low latency and high stability even in multi-channel transmission environments. Its strong interoperability also supports seamless integration with audio equipment from other brands, creating an open, flexible, and highly compatible professional audio ecosystem.
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.
Audio Routing Matrix Switches, Digital Matrix Systems
The UR-D1 represents Relacart's latest fully self-developed digital wireless microphone system, combining powerful performance with user-friendly operation. With a 64MHz operating bandwidth and low intermodulation distortion, the system supports up to 20 simultaneously operating transmitters within an 8MHz bandwidth and accommodates over 60 compatible channels in a single frequency band, ensuring stable wireless transmission even in congested RF environments.
Featuring 24-bit/48kHz digital audio resolution, the system surpasses all analog alternatives in reproducing nuanced sound details. With a total system latency of <3ms and a flat frequency response across 20Hz–20kHz, it accurately captures the full spectrum of vocal characteristics—from deep resonances to high-frequency details—delivering transparent, high-fidelity audio. A dynamic range exceeding 120dB ensures exceptional signal-to-noise performance.
The system incorporates intelligent predictive diversity reception technology, which proactively monitors signal quality and utilizes algorithmic optimization to maintain uninterrupted transmission stability. This makes it ideal for complex multi-channel and large-scale audio applications.
For secure communication needs, AES-256 encryption is available to protect privacy in confidential settings such as classified meetings. This encryption also eliminates RF interference by restricting reception to authorized encrypted signals.
Designed for seamless integration, the WireMicX Wireless management system enables effortless remote maintenance, monitoring, and control for streamlined system operation.
The UR-D2 is Relacart's latest fully self-developed digital architecture wireless microphone system. Combining powerful functionality with operational flexibility, it features a 64MHz operating bandwidth and low intermodulation distortion, enabling 20 simultaneously active transmitters within an 8MHz bandwidth and supporting over 60 compatible channels in a single frequency band. This ensures stable wireless transmission even in crowded RF environments.
With 24-bit/48kHz digital audio quality, the system accurately reproduces sonic details beyond the capabilities of analog systems. A total system latency of <3ms and a flat frequency response across 20Hz–20kHz ensure precise vocal reproduction—from deep resonances to high-frequency nuances—delivering transparent, high-fidelity sound. A dynamic range exceeding 120dB provides exceptional signal-to-noise performance.
The system incorporates intelligent predictive diversity reception technology, which preemptively assesses signal quality and uses algorithmic optimization to maintain stable transmission. It fully meets the complex demands of multi-channel, large-scale audio applications. For secure transmission needs, AES-256 encryption is available—ideal for confidential meetings—ensuring privacy protection. This encryption also eliminates RF interference by allowing only encrypted signals to be received.
Designed for easy installation and integration, the system can be seamlessly expanded via existing Dante networks, scaling from small conference rooms or classrooms to large campus and enterprise installations. Advanced features enable detailed configuration management while simplifying complex channel and audio setups. Additionally, the WireMicX Wireless management system supports remote system maintenance, monitoring, and control for streamlined operation.
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.
Providius BMG (Broadcast Media Guard) is a deploy-anywhere media observability platform purpose-built for modern IP-based media environments. Running on standard servers, virtual machines, or cloud infrastructure, BMG continuously analyzes live media streams and timing services to identify packet loss, jitter, synchronization issues, standards compliance violations, and content degradation before they impact operations. Supporting technologies such as SMPTE ST 2110, IPMX, AES67, Dante, NDI, SRT, MPEG, H.264, HEVC, PTPv1, PTPv2, NTP, and JPEG XS, BMG gives engineers real-time insight into the health, quality, and performance of media workflows through a centralized web interface and automated alerting system.
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 Extron NF 4C LP is a PoE powered, Dante-enabled 4” ceiling speaker that includes an integrated four channel audio amplifier and DSP. The NF 4C LP receives signal and power over a single network cable. It automatically recognizes connected PoE, PoE+, or PoE++ to deliver the corresponding amount of output power. This amplifier can drive up to three optional passive speakers. The NF 4C LP includes integrated audio DSP that offers the ability to independently configure processing per zone. Dante Domain Manager and AES67 support ensures wide compatibility with enterprise configurations and other network audio devices. Available in packages of two, three, or four speakers, the NF 4C LP meets the requirements of complex, decentralized systems with a self-contained PoE powered ceiling speaker solution.
The Extron NetPA U 4004 FX is a four channel, high performance Dante-enabled power amplifier with software configurable outputs that can simultaneously drive 8 ohm, 4 ohm, 70V, or 100V loads in multi-zone systems. For maximum flexibility, the total power of two adjacent 400 watt channels is shared, delivering up to 800 watts per single channel, according to system requirements. With its integrated DSP, the NetPA U 4004 FX includes essential signal processing for complex, decentralized systems while AES67 support ensures compatibility with other network audio devices. This Ecofficient Level 2 design features InstaWake, the ability to go from standby to full power in under 50 milliseconds. The fan cooled 1U enclosure does not require empty rack spaces for ventilation and includes front and rear rack mount hardware for stability.
The Extron NetPA U 2004 FX is a four channel, high performance Dante-enabled power amplifier with software configurable outputs that can simultaneously drive 8 ohm, 4 ohm, 70V, or 100V loads in multi-zone systems. For maximum flexibility, the total power of two adjacent 200 watt channels is shared, delivering up to 400 watts per single channel, according to system requirements. With its integrated DSP, the NetPA U 2004 FX includes essential signal processing for complex, decentralized systems while AES67 support ensures compatibility with other network audio devices. This Ecofficient Level 2 design features InstaWake, the ability to go from standby to full power in under 50 milliseconds. The fan cooled 1U enclosure does not require empty rack spaces for ventilation and includes front and rear rack mount hardware for stability.
The Extron NetPA U 2008 FX is an eight channel, high performance Dante-enabled power amplifier with software configurable outputs that can simultaneously drive 8 ohm, 4 ohm, 70V, or 100V loads in multi-zone systems. For maximum flexibility, the total power of two adjacent 200 watt channels is shared, delivering up to 400 watts per single channel, according to system requirements. With its integrated DSP, the NetPA U 2008 FX includes essential signal processing for complex, decentralized systems while AES67 support ensures compatibility with other network audio devices. This Ecofficient Level 2 design features InstaWake, the ability to go from standby to full power in under 50 milliseconds. The fan cooled 1U enclosure does not require empty rack spaces for ventilation and includes front and rear rack mount hardware for stability.
The Extron XTP II CP HD 8K Output boards provide local connectivity for uncompressed 8K HDMI and separate digital or analog stereo audio. These XTP® I/O boards support HDMI 2.1 performance at video resolutions up to 8K/60 at data rates up to 40.1 Gbps. They comply with HDCP 2.3. Independent routing of embedded digital audio signals enables audio breakaway switching. For enhanced system design versatility, output boards provide analog stereo audio outputs. The XTP II CP HD 8K I/O matrix boards deliver a future-proof AV infrastructure that supports convenient growth and upgrade without needing to replace equipment. They are ideal for XTP Systems® that require flexible switching of uncompressed HDMI video and audio signals.
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.
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.
