Adaptive microphone‐array system for noise reduction

Abstract

A new adaptive microphone‐array system for noise reduction (AMNOR system) is introduced. The AMNOR system detects the arrival directions of noise and forms a directional pattern possessing low sensitivities to those directions using a digital filtering technique. With pre‐informed data on the direction of desired signal source, the AMNOR system learns the noise arrival directions previously. Optimum filter coefficients are determined based on a new criterion, whose concept is to minimize output noise power while maintaining the degradation in the frequency response to the desired signal below the predetermined value. An algorithm for calculating the optimum filter is proposed. To confirm the effectiveness of the AMNOR system, experiments on the noise reduction processing were carried out in a room with a reverberation time of 0.4 s. A small circular microphone array (radius of 8.5 cm) with four microphone elements and four FIR filters with 16 taps were used. The convergence time of the algorithm at this experimental condition was 0.3 s. Improvement in SNR as a result of the AMNOR system by more than 15 dB in the frequency range of 300–3300 Hz. Superiority over the conventional LMS criterion for noise reduction of speech signals was also confirmed by subjective preference tests.