Open Architecture Distributed Network Audio Processing

Amu 950 is a distributed network audio system processing service center, serving as a core AI audio algorithm processing platform for enterprises. It integrates powerful hardware processing capabilities with a diverse array of advanced audio algorithms. The system employs the first domestically developed audiovisual transmission protocol, TENDZONE VaiLink, specifically for the professional video field, providing ultra-low latency, uncompressed network audio input, output, and processing capabilities with a capacity of 128x128 channels. It supports 16x16 RTSP network audio streams, with audio matrix routing reaching 144x144, and is expandable and compatible with protocols like DANTE and AES67, supporting a maximum of 128x128 channels.

The Amu 950 features a configurable slot design with a total of 9 slots, allowing flexible configuration of up to 8 analog input/output cards, with optional DANTE expansion cards. The system's functions are software-defined, allowing for custom audio processing capabilities and system architecture, and it supports centralized management of network microphones, network speakers, audio nodes, and other devices.

It also incorporates a 1+1 hot backup design, supporting dual AC power supplies, ensuring the system's reliability in applications of any scale. With its exceptional performance and reliability, it presents an outstanding solution.

The Amu 950's extensive network audio channel access capacity and powerful processing capabilities make it suitable for a variety of large applications, including major command centers, conference centers, administrative centers, corporate headquarters, theme parks, and transportation hubs.

1. Network Audio Capabilities

128x128 VaiLink Channels: Supports uncompressed, ultra-low-latency audio over the VaiLink protocol.

144x144 Audio Matrix Routing: Full-scale audio routing with precision and flexibility.

16x16 RTSP Stream Support: Integration of real-time audio streaming over IP.

DANTE and AES67 Compatibility: Expandable for industry-standard protocols up to 128x128 channels.

2. Flexible Hardware Architecture

Modular Slot Design: 9-slot chassis design, configurable with:

Up to 8 analog input/output cards

Optional DANTE expansion cards

Software-Defined Functions: Customizable audio workflows and processing functions via design platform.

3. Advanced Audio Processing

16 AEC Modules: With up to 512ms tail length echo cancellation.

40 AFC Modules: Flexible allocation for adaptive feedback control.

Hundreds of Audio Algorithms:

5A Algorithms

AI Noise Reduction

AI Auto Gain Control

Automatic Mixing (Threshold & Gain-Sharing)

Dynamic Compression and more.

4. Intelligent Microphone Management

MAOM System (Microphone Array Optimization Management):

Manage up to 128 microphones

Automatic interference analysis to prevent comb filtering

Intelligent gain allocation to prevent feedback

All microphones can remain open without screeching

5. Preset and Workflow Customization

Modular Preset Management: Add/exclude processing modules as needed.

Refined Presets: Apply settings at individual matrix crosspoints.

Offline Design Tools: Simulate workflows and system behavior prior to deployment.

6. User Interface & Control

Visual Management Interface:

Real-time dynamic audio level feedback

Crosshair cursor for precise signal routing

Multi-Platform UI: Compatible with Windows, Kylin, Android, iOS

Custom Interface Design: Personalized UI for different user roles

Multi-User Support: Hierarchical control authority, simultaneous operation

7. System Integration and Reliability

Open API: For seamless third-party integration

Industrial-Grade Architecture:

Linux OS-based system for security and stability

Immune to common viruses

1+1 Hot Backup Design:

Dual AC power supply for uninterrupted operation

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