Study on adaptive microphone array system for concurrent speech segregation

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Two algorithms are proposed here ; one uses only two microphone elements, other uses multiple elements.Two elements :Frequency Domain Binaural Model (FDBM) is proposed. The algorithm performs segregation based on binaural information ; inter-aural phase and level differences. Advantage of the algorithm is archive to detect both azimuth and elevation for direction of arrival with only two elements. Moreover, real time processing using a laptop computer is performed due to low calculation cost.Multiple elements (over 2) :This model is proposed for computer users. Eight microphone elements are attached to frames of a computer display. Blind signal separation based on AMUSE method is performed as pre-processing. In post-processing, adaptive notch filter is designed for each sound source based on each direction of arrival in order to segregate signals simultaneously.In this research project, two microphone array algorithms are proposed for the future human interface. In particular, the former method, FDBM, is expected to apply to hearing-aid system that takes into account of binaural information.In future, improvement and arrangement of the algorithms will be discussed to use the algorithm in a daily life.