Features:

*High-performance quad-core ARM CorteX-A53 processor (2.0GHz), 32bit/48KHz AD/DA, professional DSP processing, providing  excellent high-quality sound.

*Features a 4-input, 4-output audio matrix, with adjustable input sensitivity based on the audio source. Each input supports 48V phantom power and can be individually configured to be on or off, offering flexibility and convenience.

*Input channels support preamplifiers, signal generators, expanders, compressors, equalizers (12-band parametric EQ, selectable 10/15/31-band graphic EQ adjustable, graphic EQ can be used for individual bandwidth adjustment), ducks, AGC automatic gain control, AM automatic mixing (threshold type, gain-sharing type), AFC adaptive feedback cancellation, AEC echo cancellation, ANC noise cancellation, and audio matrix; output channels support equalizers (16-band parametric EQ, selectable 10/15/31-band graphic EQ adjustable, graphic EQ can be used for individual bandwidth adjustment), delays, crossovers, high-pass and low-pass filters, and limiters; based on the feedback detection threshold update method, it has frequency shift + notch filter combined feedback suppression, and can use 24 programmable notch points, which can be freely assigned dynamic/static points and switched automatically/manually.

*It has a ducking function, which automatically ducks the microphone when background music is played. It offers a variety of parameter settings for convenient and flexible use on site.

*It features automatic gain control for the microphone, which controls the dynamic range of the microphone's pickup signal to ensure consistent sound quality at different distances.

*It features intelligent mixing capabilities, including gain-sharing mixing and automatic gate mixing. Input channels can be individually selected for participation in intelligent mixing, allowing for the selection of appropriate mixing modes based on different application scenarios. This effectively addresses pain points such as instability and feedback caused by multiple microphones operating simultaneously.

*Features equalizer functionality, offering both parametric and graphic equalizers. Each input channel offers a choice  of  12-band  parametric, 10-band, 15-band, or 31-band graphic equalizers, while each output channel offers a choice of 16-band parametric, 10-band, 15-band, or 31-band graphic equalizers. The parametric equalizer supports three types: high-mount, low-mount, and peak filter. The graphic equalizer supports single-point bandwidth adjustment.

*It has a crossover function, offering three filter types to choose from: Bessel, Linkwich-Rayleigh, and Butterworth. It also supports slope settings of 6/12/18/24/32/40/48dB/oct, and the filter is adjustable across the entire frequency band.

*It has an extender function to expand the dynamic range of the signal and is used to eliminate the background noise of the device.

*It has a compressor function to compress the dynamic range of the signal, which is used to compress the size of the output signal.

*It has a limiter function to limit the size of the output signal and prevent the signal from being too large and damaging the sound reinforcement equipment.

*It has a delay function, providing a maximum delay adjustment of 2000ms, which can be used to adjust the delay of each output signal so that each audio signal is synchronized when it reaches the listener's ears.

*It has an echo cancellation function for remote audio and video conferencing, eliminating echoes and increasing voice clarity; the echo cancellation can be selected according to the size of the video conferencing room (three levels are available: small room, medium room, and large room); the noise reduction level is adjustable, with five levels available: 6dB, 12dB, 18dB, 24dB, and 30dB.

*It has a noise cancellation function, which can effectively eliminate environmental noise such as air conditioner noise and fan noise, and improve speech clarity. The steady-state noise reduction level can be adjusted to different levels according to the needs of the site (five levels are available: 6dB, 12dB, 18dB, 24dB and 30dB).

*It has feedback suppression function, with two processing solutions: notch filter and frequency shifter, which effectively solves the acoustic feedback problem.

*Extremely low system processing latency, only 1.1ms.

*The panel features a 2-inch IPS true-color display that shows device network information, real-time levels, channel mute status, matrix mixing status, and other information.

*The panel has a USB interface, which supports multimedia storage and can store recordings or play them back.

*Supports scene presets, import, and export, supporting 8 scenes. Scene switching can be completed within 0.1 seconds.

*It has a factory reset function.

*It has an RS-232 interface, which can be used to connect to an external central control system to achieve centralized management and control.

*It has an RS-485 interface, which can be connected to the central control system and the camera tracking system to realize the automatic camera tracking function.

*Features an 8-channel programmable GPIO control interface (customizable inputs and outputs).

*Supports channel copy, paste, and joint control functions.

*Ethernet multipurpose data transmission and control port, which can support real-time management of single or multiple devices.

*The audio processor has cross-platform software that can run on Windows operating system, or domestic Galaxy Kylin desktop operating system (Zhaoxin version), or domestic Galaxy Kylin desktop operating system (Feiteng version), or macOS system, or Tongxin UOS or Ubuntu desktop operating system.

*The product has different control methods, including PC client, mobile phone, Android tablet, iOS mobile phone, and iOS tablet. It can connect to the device simultaneously by logging into the APP software and PC client, and realize the synchronization of data across multiple devices.

*Supports operation and control via Android mobile phone and  tablet APP software, including device login, scene switching, input and  output, matrix routing and channel settings.

*Supports automatic saving function after power failure.

*The rear panel features a 4-line audio Phoenix terminal balanced input interface (with 48V  phantom power), a  4-line audio Phoenix terminal balanced output interface, a DIP switch, an RJ45 interface, an RS232 interface, an RS485 interface, 8  programmable GPIO control interfaces, and a grounding post; the front panel features a 2.0-inch IPS true color display, an encoder knob, and a USB storage device interface.

*The equipment has a unified centralized control function, supporting centralized control of 65,535 devices through software.

*It has a matrix gain adjustment function, and the gain of each input channel participating in the mixing is adjustable, with a gain adjustment range of -72dB to 12dB.

*The device has an encoding knob and an IPS screen, which can be used to control and configure the device's mute, gain, and scene settings; the IPS screen can display the IP address, input and output channel real-time levels.

*It has a device location function. The client can locate similar devices in the local area network with one click, and the located device will display the location information.

*The audio processor software can be integrated into a conference audio management platform to achieve unified management of audio devices. The platform can scan the online status of digital conference hosts, audio processors, mixers, suppressors,  and  power amplifiers. It can also scan multiple online devices of the same product and display the device hardware name, hardware IP address,  online and offline status information. It has the function of one-click uploading configuration information to the cloud or saving it locally for backup and one-click restoration of configuration information.

*Supports central control functions, enabling gain adjustment, gain limiting, level bar query for each channel, and  gain adjustment step  value setting.

*Supports dual-machine hot backup; when the primary machine fails, the backup machine automatically takes over the service, with a switchover time of less than 2 seconds, ensuring that the system can operate normally without manual intervention.

*It adopts an AI wide-range learning algorithm for noise suppression. The AI voice noise reduction offers three adjustable levels (strong, medium, and weak) depending on the usage scenario, effectively eliminating non-steady-state noise such as clapping and knocking on  the table, ensuring speech clarity and intelligibility (noise reduction capability up to 40dB).

*It has a web page management function, which allows users to download control software and adjust the gain of each channel via the web page.

*The product enables cloud-based collaboration with the operation and maintenance platform, supports automatic  detection  of  OTA upgrade firmware packages after the device is connected to the network, and has complete remote firmware update capabilities.

Specification: