Poor sound pick up by remote microphones in multimedia applications, conference rooms and auditoria has traditionally hampered speech recognition and communication among spatially-separated groups. The problems are reverberation, acoustic noise, and the variability of the radiation pattern of unconstrained talkers. One potential solution that is becoming increasingly practical is to use an array of microphones and sophisticated signal processing. In this paper a brief description of a large, real-time, working system is presented and its measured beamforming performance is compared to what is predicted from a mathematical model. A combination of a synchronized test signal/system and a careful mathematical model results in the performances matching surprisingly well. From this match of theory to practice, we are able to draw some important inferences about future system improvements.
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