Company Introduction

Domestic Leading Audio Algorithm R&D Base     Diversified ODM/OEM Manufacturer  

Level 3 Partner Confidentiality Program       Customized Native UI Platform

Shenzhen Tongchuang Audio Technology Co., Ltd., as a national high-tech enterprise actively responding to the trend of domestic production, adheres to independent innovation and is dedicated to the research, development, and application of high-performance audio digital signal processor software and hardware. Our products cover diverse scenarios such as commercial background music, conferences, education, courtroom trials, theme parks, aiming to provide excellent DSP solutions to a wide range of users and lead the industry in cutting-edge product development and technical services.

The company gathers industry elites and boasts a group of highly qualified research and application technology talents. Leveraging rich industry experience and professional knowledge, they continuously push the boundaries of technology. The configurable DSP series and open architecture platforms launched have established a leading position in the industry, excelling in sound enhancement, noise reduction, automatic mixing, echo processing, signal distribution, and routing. Our supporting software is also outstanding, highly praised in the market.

We closely cooperate with domestic authoritative institutions and collaborate with well-known enterprises to jointly promote technological innovation and domestic production. Over the years, we have accumulated fruitful results, owning over 10 invention patents, more than 20 appearance patents, and hundreds of software copyrights, fully demonstrating strong independent R&D capabilities. Meanwhile, we flexibly adapt to market demands and have successfully provided personalized OEM/ODM services to over 1000 domestic and international brands.

In the future, Shenzhen Tongchuang Audio Technology Co., Ltd. will continue to uphold the spirit of innovation, develop more high-quality audio solutions, and contribute to the cause of domestic production.

Technical Advantages

Automatic Mixing Technology based on ATS Noise Perception

ATS automatic mixing technology is based on adaptive noise thresholds and adaptive coherence of speech signals to automatically activate relevant microphones. On one hand, it perfectly resolves issues present in traditional threshold-based automatic mixers where speaker sound can be inadvertently triggered by microphones picking up from non-speaking positions. On the other hand, it effectively addresses the comb filtering phenomenon that occurs when two adjacent microphones receive speech signals. It also solves problems like dropout during traditional voice activation, ultimately achieving maximum vocal gain when all microphones are activated.

Neural Network-based Echo Cancellation Technology

Echo cancellation technology based on a neural network framework addresses the challenge faced by traditional echo cancellation methods in accurately determining near-end, far-end, and double-talk scenarios, thereby reducing the interference of echo in bidirectional sound transmission.

This technology, leveraging the strong nonlinear fitting capability of neural networks, eliminates linear and nonlinear echoes directly without relying on traditional determinations. It can effectively handle uncertainties such as varying speech durations during meetings, as well as issues arising from near-end, far-end, or double-talk situations, improving echo cancellation quality and ensuring integrity in bidirectional sound transmission.

Adaptive Acoustic Feedback Technology

Adaptive acoustic feedback technology is designed to address the issue of local sound reinforcement gain caused by acoustic feedback. Techniques such as frequency shifting, notch filtering, and adaptive filtering are employed to reduce gain and alter system phase to mitigate these issues. In practical applications, adaptive acoustic feedback technology from our company utilizes prediction error methods to pre-whiten the source signal components of microphone and speaker signals, significantly enhancing sound quality by accounting for the correlation between feedback signals and sound sources.

AI Intelligent Noise Reduction Technology

Based on Wide Learning AI algorithms, this technology utilizes a network structure that dynamically adjusts its node count as it learns, facilitating online learning with extremely low latency. It intelligently removes non-human noises present during voice amplification processes, such as flipping pages, writing, tapping on desks, and fan noises, enhancing speech clarity by effectively eliminating non-speech signals.

Plug-in Programmable DSP

Featuring 8 customizable I/O slots, 128-channel Dante network audio, a vast 256x256 channel matrix, a 16-channel player paired with Lua script editor, dual-machine hot backup, and other functionalities, this system perfectly meets the demands of large-scale emergency command centers and multi-conference clusters.

Configurable Software

The configurable operating software boasts a rich database where processing modules can be directly accessed and edited. During editing, it's possible to replace or remove processing modules for input and output channels. Additionally, CPU and memory usage can be monitored, supporting centralized debugging across 8 devices. The software is robust, featuring a customizable user interface, online software upgrades, level alarms, dual-machine hot backup, permission management, real-time saving, and the ability to control the entire system with a single set of software.

Customizable User Interface

The customizable user interface is designed specifically for end users, allowing for easy device management and flexible editing based on different project requirements. It supports mobile app control, enabling convenient management of multiple venues within the same network using a tablet. Real-time status monitoring and fault alarms are available, along with centralized control functionality for overall system management including power, signal switching, environmental control, and audio. This allows for one-touch activation of system functions and supports wireless remote control via iPad, tablets, and smartphones, compatible with Windows, Android, and iOS systems. Suitable for various small to medium-sized conference scenarios.

ADI SHARC Processing Chip

The processor utilizes the ADI SHARC processing chip as its computational core, featuring a 1GHz core clock speed, 40-bit floating-point operation capability, and support for 12S, left-aligned sample pairs, DSP serial and TDM mode FIR, IIR, and FFT accelerators. It is equipped with 256M DDR3 memory, striking a balance between exceptional core and memory performance and outstanding I/O throughput capabilities.