Description：

This high-performance audio processor features 12 analog balanced inputs and 12 analog balanced outputs. It integrates Dynamic Range Control (DRC), Automatic Gain Control (AGC), Affected Feedback (AFC), Adaptive Noise Reduction (ANS), AI Intelligent Noise Reduction, Adaptive Echo Cancellation (AEC), and audio filters (GEQ, PEQ, crossover). Primarily designed for professional sound reinforcement applications, it meets the needs of conference rooms, courtrooms, auditoriums, multi-purpose halls, performance venues, classrooms, and other similar settings.

Features：

* It features a built-in high-performance quad-core processor with a main frequency of 2.0GHz, 32-bit/48KHz AD/DA, and professional DSP processing, providing excellent high-quality sound.

* It features a 12-input, 12-output audio matrix, and the input sensitivity can be adjusted according to different audio sources. Each input supports 48V phantom power and can be individually configured to be turned on or off, offering flexibility and convenience.

* The input channels support preamplifiers, signal generators, expanders, compressors, equalizers (12-band parametric EQ, with optional 10/15/31-band graphic EQ adjustable, the 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; the output channels support equalizers (16 - band parametric EQ, with optional 10/15/31-band graphic EQ adjustable, the 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 features a ducking function to automatically duck the microphone when background music is played, and offers a variety of parameter settings for flexible on-site use.

* It features automatic gain control for microphones, 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 both gain-sharing mixing and automatic gate mixing. Individual input channels can be selected for participation in intelligent mixing, allowing users to choose the appropriate mixing mode based on different application scenarios. This effectively addresses pain points such as instability and feedback in the sound reinforcement system caused by multiple microphones operating simultaneously.

* It 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 features a frequency divider 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 noise floor of the device.

* It has a compressor function, compressing the dynamic range of the signal and 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 features 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 remains synchronized when it reaches the listener's ears.

* It features echo cancellation and noise reduction capabilities for remote audio and video interaction, eliminating echoes and improving voice clarity and fluency. Echo cancellation offers three selectable levels based on room size (small, medium, and large rooms). Adjustable noise reduction levels are available: 6dB, 12dB, 18dB, 24dB, and 30dB. Built-in AI echo cancellation can simultaneously suppress linear and nonlinear echoes in complex two-way communication, strong reverberation, and noisy environments.

* It features feedback suppression and offers two processing solutions: notch filter and frequency shifter, effectively solving the acoustic feedback problem.

* Extremely low system processing latency, only 1.1 ms.

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

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

* It supports scene presets, imports, and exports, and supports 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 external central control systems 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.

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

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

* It features an Ethernet multipurpose data transmission and control port, enabling real-time management of single or multiple devices.

* It features cross-platform software and can run on Windows operating systems, domestic Galaxy Kylin desktop operating systems (Zhaoxin version), domestic Galaxy Kylin desktop operating systems (Feiteng version), macOS systems, Tongxin UOS systems, or Ubuntu desktop operating systems.

* It offers different control methods, including PC client, mobile phone, Android tablet, iOS mobile phone, and iOS tablet. Users can log in to the APP software and PC client at the same time to connect to the device and achieve data synchronization across multiple devices.

* It supports operation and control via Android phones and tablets via an app, including device login, scene switching, input/output, matrix routing, and channel settings.

* Supports automatic saving function in case of power failure.

* The rear panel features a 12-line audio Phoenix terminal balanced input interface (with 48V phantom power), a 12-line audio Phoenix terminal balanced output interface, a DIP switch, an RJ45 interface, an RS232 interface, an RS485 interface, eight 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 features matrix gain adjustment, allowing for adjustable gain for each input channel involved in mixing, with a gain adjustment range of -72dB to 12dB.

* The device features an encoder knob and an IPS screen for controlling and configuring device mute, gain, and scene settings; the IPS screen displays the IP address and real-time levels of the input and output channels.

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

* The audio processor software can be integrated into a comprehensive 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 restoring configuration information.

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

* It supports dual-machine hot backup; when the main 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 employs AI deep learning algorithms for noise suppression, offering three adjustable AI voice noise reduction levels to suit different usage scenarios. This effectively eliminates non-steady-state noise such as clapping and tapping on a table, as well as steady-state noise such as air conditioner and fan sounds, ensuring speech clarity and intelligibility (noise reduction capability ≥40dB).

* It features web page management capabilities, allowing 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.

Specifications：