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The Sony MAS-A100 is an IP-based beamforming microphone designed to provide clear audio for both speech reinforcement and recording in corporate meetings, lectures, and presentations. The microphone is simply affixed to the ceiling and ready to use, saving you the inconvenience of setting up a lavalier mic or holding a handheld mic.
The beamforming technology provides a pickup pattern that changes with the location of the sound source, allowing the presenter to move around while maintaining consistent sound quality, great for hands-free presentations.
With Dante networking capability, the MAS-A100 can be integrated into existing A/V setups and used with a wide range of products supporting the Dante digital audio-over-IP networking standard including interfaces, switches, mixing consoles, and more. Power over Ethernet (PoE) enables installation and power management with a single network cable.
The MAS-A100 has two output channels. One for speech reinforcement (with a PA system required) and one for simultaneous recording. The dedicated recording channel provides a wider capture range and optimal intelligibility, allowing for crisp recordings that include not only the speaker’s voice but also the students or participants.
- The MAS-A100 is equipped with proprietary Intelligent Feedback Reducer technology designed to extract speech sound while suppressing unwanted feedback via digital signal processing.
- With advanced noise-reduction technology the noises arising from projectors and air conditioning are automatically detected and reduced, providing better intelligibility.
- After capturing the voice, the microphone’s Automatic Gain Control function automatically adjusts the output volume to remain consistent regardless of the presenter’s location, making it easier for your message to get through to your listeners.
- Offered as a free download, Sony’s Microphone Array System Manager software (MASM-1) supports the installation and management of one or multiple microphones centrally via the local network. The software is easy to set up and operate, and assists with initial setup, calibration, general settings, operations, and firmware management.
- To minimize setup time during the initial installation, the automatic calibration function generates and captures a test signal, which is used to automatically optimize the parameters of audio processing for speech-reinforcement applications.
- The unit can also be operated via API for simple external control and customization.
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