Description
Download Manual Please NOTE: This listing is for B-Stock item. It means that this unit was refurbished by manufacturer and holds full manufacturer warranty, however it may have scratches, signs of wear and tear or come not in original box.
The BG-A1616MD Digital Audio Processor is a versatile solution designed for video conferences, distance learning, and telemedicine applications. With its advanced features and intelligent algorithms, this processor delivers exceptional audio quality and seamless control. It is equipped with 12-ch MIC/linear inputs, 12-ch linear outputs, and two DANTE ports for low-latency audio processing. This powerful processor is ready to use immediately, ensuring hassle-free installation and project implementation.
Advanced Audio Processing Features
The BG-A1616MD Digital Audio Processor incorporates a range of advanced audio processing features. It includes full-band Adaptive Echo Cancellation (AEC), which effectively eliminates echoes for clear and resonant sound. The Adaptive Noise Suppression (ANS) technology reduces background noise, ensuring clean audio signals. The Automatic Gain Control (AGC) feature adjusts the microphone input levels automatically, maintaining a consistent volume. Additionally, the Auto Mixer function allows you to prioritize microphone inputs when multiple microphones are used simultaneously. These features work together to provide a high Signal-to-Noise ratio and deliver exceptional audio quality.
Seamless Audio Signal Routing
With the BG-A1616MD, seamless audio signal routing is made easy. It offers 12-ch balanced/unbalanced MIC/linear inputs and 12-ch balanced/unbalanced linear outputs, allowing for efficient and flexible audio routing. The processor is equipped with standard DANTE network audio interfaces, enabling low-latency communication. It also incorporates adaptive feedback suppression, full-band acoustic echo cancellation, and dynamic adaptive noise reduction technologies. These technologies effectively eliminate unwanted audio artifacts and background noise, ensuring clear and pristine audio during conferences, learning sessions, or telemedicine consultations.
Fine-tuning Audio Signals with DSP Modules
The BG-A1616MD provides an array of digital signal processing modules that allow you to fine-tune your audio signals. The processor includes modules such as Expander, Equalizer, Compressor, Auto Gain Control, Limiter, High Pass Filter, Low Pass Filter, and Delay. These modules enable you to customize and optimize the audio to your specific requirements. The intuitive interface facilitates easy configuration, volume adjustment, audio meter monitoring, and scene control. With these features, you have precise control over your audio signals and can create an optimal listening experience.
Versatile Connectivity Options
The BG-A1616MD offers versatile connectivity options to accommodate various setups. It supports a 48V phantom power supply for all 12-ch MIC inputs, ensuring compatibility with a wide range of microphone types. The processor operates at a 48KHz sampling rate and performs 24-bit A/D or D/A conversion, guaranteeing high-fidelity audio reproduction. Additionally, it provides an 8-ch programmable GPIO function, enhancing flexibility and control in your audio setup. The processor seamlessly integrates with both Windows 7 and Windows 10 platforms, and its standard RJ45 interface control allows for easy integration into existing setups.
Advanced Control Capabilities
The BG-A1616MD is equipped with advanced control capabilities through RS-232 serial command control. This feature enables precise and comprehensive control over the audio processor, allowing for fine-grained adjustments and customization.
What is Dante Audio?
Dante (Digital Audio Network Through Ethernet) is a proprietary technology developed by Audinate that allows uncompressed, multi-channel digital audio to be transmitted over standard Ethernet networks with extremely low latency. Instead of relying on traditional analog cables or point-to-point digital connections, Dante uses standard IP networking, making audio routing more flexible and scalable.
Key Features:
- Uncompressed, high-quality audio: Delivers pristine, professional-grade sound without degradation.
- Low latency: Typically less than 1 millisecond, suitable for live sound and broadcast applications.
- Channel capacity: Can handle hundreds of audio channels over a single network cable.
- Synchronization: Uses precise clocking for sample-accurate playback across all devices (no drift).
- Plug-and-play: Devices automatically discover each other on the network using Dante Controller software.
- Scalability: Works over standard switches and routers, allowing systems to grow from a few channels to thousands.
Typical Application
- Live sound reinforcement (concerts, theaters, stadiums)
- Broadcast and recording studios
- Installed AV systems in corporate, education, and government spaces
- Houses of worship
- Anywhere multi-channel audio needs to be routed reliably and flexibly over IP
In short: Dante Audio replaces bulky analog and digital cabling with a single network cable, giving AV professionals a simple, scalable, and rock-solid way to manage audio across devices and locations.
Features
- DANTE Ports – Two DANTE ports for low-latency audio processing
- Auto Mixer – for prioritizing microphone inputs
- Quality Control – Adaptive feedback suppression, full-band acoustic echo cancellation, and dynamic adaptive noise reduction technologies
- DSP Modules – Fine-tuning audio signals with DSP modules (Expander, Equalizer, Compressor, Auto Gain Control, Limiter, High Pass Filter, Low Pass Filter, and Delay)
- Windows Compatibility – Seamless compatibility with Windows 7 and Windows 10 platforms
- Advanced Control – Advanced control capabilities through RS-232 serial commands control
- 12-ch MIC/linear inputs
- 12-ch linear outputs
- Seamless audio signal routing with 12-ch balanced/unbalanced MIC/linear inputs and 12-ch balanced/unbalanced linear outputs
- DANTE network audio interfaces
- Intuitive interface for easy configuration, volume control, monitoring, and scene management
- 48V phantom power supply for all 12-ch MIC inputs
- 48KHz sampling rate and 24-bit A/D or D/A conversion for high-fidelity audio reproduction
- 8-ch programmable GPIO function for enhanced flexibility and control
- Standard RJ45 interface control for easy integration into existing setups
- High Signal-to-Noise ratio for a clear, clean, and resonant sound
