8x8 Web Based Dante Audio Processor

Tendzone's WebGem 0808D DSP provides an extensive range of adaptable audio DSP tools specifically designed for AV integrators. It comes with eight channels for analog audio input and output, in addition to a 16x16 model supporting Dante™ signal extension, which caters to diverse audio processing needs. The DSPs feature an easy-to-use Web-UI and support standard bidirectional control via RS-232, TCP/IP, and GPI, allowing for seamless integration with various third-party control systems. Users can also create a custom web control interface for local room management that works across various platforms such as Windows, iOS, macOS, and Android. The WebGem series from Tendzone combines impressive audio processing capabilities with flexible control solutions, making it perfect for AV integration in settings like conference rooms, educational facilities, hotels, and courtrooms.

• This unit offers 8 channels of balanced microphone/line inputs equipped with a soft switch for 48V phantom power.
• It is designed with 8 balanced line outputs and provides 16x16 Dante network audio compatibility with AES67.
• Enjoy remarkable sound quality at 48 kHz and 24-bit resolution.
• The preamp includes gain optimization with settings at 0dB, 10dB, 20dB, 30dB, and 40dB, accommodating both MIC and LINE levels.
• There is a USB2.0 audio interface for recording and playback using a USB drive.
• The USB-B sound card interface supports 2x2 audio and can connect with various devices on Windows, iOS, Android, and HarmonyOS for two-way audio transmission.
• The device incorporates Adaptive Echo Cancellation (AEC), which swiftly removes echoes that arise during video conferencing.
• Additionally, Adaptive Feedback Cancellation (AFC) applies advanced algorithms to each microphone for efficient feedback suppression.
• Integrated Automatic Gain Control (AGC) maintains steady output volume, regardless of the speaker's proximity to the microphone.
• The intelligent Ducker function helps to reduce background music levels easily, placing emphasis on the vocals.
• A proactive limiter responds instantly, ensuring protection for downstream equipment.
• The system features gain-sharing automatic mixing (AM) and threshold-based automatic mixing (GAM) for easy oversight of multiple microphones.
• Users can define ranges for input and output levels, setting maximum and minimum values for better usability.
• The system includes a user-friendly graphical software control interface structured on B/S architecture, allowing adjustments via a browser in English or Chinese.
• The user interface can be customized and operates effectively across multiple platforms including Android, iPad, Windows, HarmonyOS, and KylinOS.
• A 1.8-inch color LCD provides valuable information such as device name, IP address, input/output levels, mute status, current scene, and software version.
• It features 8 configurable GPIO interfaces, which can be assigned as inputs or outputs through software.
• The open design allows for third-party control protocols using UDP or RS-232 for device management and control.