적응 후처리 과정을 갖는 마이크로폰 배열을 이용한 잡음제거기의 DSP 구현

DSP Implementation of Speech Enhancement System Using Microphone Array with Adaptive Post-processing

  • 권홍석 (경북대학교 전자전기공학부) ;
  • 김시호 (경북대학교 전자전기공학부) ;
  • 배건성 (경북대학교 전자전기공학부)
  • 발행 : 2002.06.01

초록

In this paper, a speech enhancement system using microphone array with adaptive Post-Processing is implemented in real-lime with TMS320C6201 DSP. It consists of delay-and-sum beamformer and adaptive post-processing filters with NLMS (Normalized Least Mean Square) algorithm. THS1206 ADC is used for collection of 4-channel microphone signals. Sizes of program memory, data ROM and data RAM of the implemented system are 15,744, 748 and 47,540 bytes, respectively. Finally 21.839${\times}$106 clocks per second is required for real-time operation.

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