聲學回授音(Acoustic Feedback)對於助聽器使用者是主要干擾聲音辨識的因素之一，本研究提出新的適應性回授消除 (Adaptive Feedback Cancellation, 簡稱AFC) 演算法，使用雙麥克風架構，利用副麥克風收入語音作為主麥克風輸入語音的估測值，在適應性濾波器更新時去除輸入訊號的影響，一方面解決輸出訊號與輸入訊號相關性過高造成估測誤差提升的問題，另一方面也使適應性濾波器更新時不隨輸入訊號能量大小而波動，除此之外，本架構利用兩組適應性濾波器對雙麥克風做回授消除，改善原始架構在放大增益較大的情況會失效的缺點，滿足重度聽損者對於放大增益的需求。經過模擬，在不同回授路徑及不同放大增益下皆能有效的消除回授音，所估測出的回授路徑跟外部回授路徑相比，係數估測誤差(Misalignment)能減少7dB，此外，能多提供7dB的增加穩定增益(Added Stable Gain, 簡稱ASG)也加寬了助聽器所能調整的增益範圍，使需要較多補償的配戴者能有足夠的補償且不受回授音干擾影響，提升輸出音訊品質讓使用者能有較好的聽覺感受。

In this thesis, a dual microphone adaptive feedback cancellation algorithm (AFC) is proposed. The reason of employing an additional microphone is to provide added information about the signal which is then utilized to obtain an incoming signal estimate. This estimate is removed from the primary microphone signal to create the error signal which adapts the adaptive filter’s coefficients. The purpose of using dual microphone is to overcome the problems remaining in using adaptive filters for feedback cancellation. One is the biased estimate of filter’s coefficients because of the correlation between the loudspeaker and incoming signal. The other is that filter’s coefficients may variate according to the change of incoming signal. Besides, in comparison to traditional work of dual microphone structure, the proposed can provide more gain compensation for a large class of hearing loss degree by making use of two adaptive filters to remove the feedback. That is, it can still cancel feedback noise without being out of control while the power of feedback noise increases. Finally, the results show that the proposed AFC is better than other AFC by 7 dB in misalignment and 7 dB extra added stable gain (ASG) is obtained. Those statistical indexes mean that proposed AFC can estimate feedback path more accurately and produce more precise output signal. Therefore, it is very suitable for future applications in the area of hearing aids.

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