AN ADAPTIVE MICROPHONE ARRAY-PROCESSING SYSTEM

被引:11
|
作者
DOWLING, EM
LINEBARGER, DA
TONG, Y
MUNOZ, M
机构
[1] The University of Texas at Dallas, Erik Jonsson School of Engineering and Computer Science, Richardson
关键词
MICROPHONE ARRAYS; SPEECH ACQUISITION; SIGNAL PROCESSING;
D O I
10.1016/0141-9331(92)90080-D
中图分类号
TP3 [计算技术、计算机技术];
学科分类号
0812 ;
摘要
A microphone array processing system has been developed for enhanced speech acquisition. The key aim is to allow the microphone array to direct its own pattern to selectively enhance a desired signal while attenuating interference. Basic design considerations are to keep the part count low and system cost down. While the prototype system is designed for research use, the design readily extends to production systems. Each channel of the eight element array consists of an omnidirectional microphone, analogue signal conditioning and amplification circuitry, a TLC32044 A/D subsystem and a serial buffer. Programmable logic circuits are used to force the eight channels to sample the inputs simultaneously, and to multiplex and send the data to a TMS320C30 development system using the TMS320C30 serial port protocol. The microphone array data is processed in order to form beams, null out interfering sources and to determine the direction of arrival of desired speech signals. Experimental results with the generalized sidelobe canceller beamformer, and Capon's direction finding technique, demonstrate the array's utility and functionality.
引用
收藏
页码:507 / 516
页数:10
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