Title :
Acoustic feedback reduction based on FIR and IIR adaptive filters in ITE digital hearing aids
Author :
Martínez-Leira, A. ; Vicen-Bueno, R. ; Gil-Pita, R. ; Rosa-Zurera, M.
Author_Institution :
Dimetronic-Signals-San-Fernando-Bus.-Park, Madrid
Abstract :
Acoustic feedback phenomenon can deteriorate the performance of digital hearing aids performance working at high gains, causing instability and speech degradation. In order to restore a stable situation, it is needed a feedback reduction system based on adaptive algorithms. Depending on the adaptive flter impulse response, a feedback reduction algorithm based on a finite impulse response (FIR) or an infinite impulse response (IIR) adaptive filter can be used. The commonly used adaptive algorithm based on FIR filters is the least-mean square (LMS) algorithm, but it becomes unstable if some conditions are not fulfilled. In order to avoid this problem, a feedback reduction subsystem based on IIR filters is proposed. In this case, the IIR filter is adapted with a modified version of the LMS algorithm for this kind of filters, which is called the IIR-LMS algorithm. These two adaptive algorithms are tested in the In-The-Ear (ITE) hearing aid category, which the category that most suffer from acoustic feedback. In order to determine which feedback reduction technique (FIR or IIR) is the most suitable for the ITE hearing aid, the added stable gain (ASG) value over its limit gain of performance is obtained. The ASG value is achieved as a tradeo between the speech quality (subjective parameter) and the segmented signal-to-noise ratio (objective parameter). The results show how the digital hearing aid working with an acoustic feedback reduction subsystem based on IIR filters adapted with the IIR-LMS algorithm achieves an increase of 8 dB in its ASG.
Keywords :
FIR filters; IIR filters; acoustic signal processing; adaptive filters; hearing aids; least mean squares methods; FIR adaptive filters; IIR adaptive filter; IIR-LMS algorithm; ITE digital hearing aids; acoustic feedback reduction; added stable gain; finite impulse response filter; infnite impulse response filter; least-mean square algorithm; segmented signal-to-noise ratio; Adaptive algorithm; Adaptive filters; Auditory system; Feedback; Finite impulse response filter; Hearing aids; IIR filters; Least squares approximation; Performance gain; Speech;
Conference_Titel :
Audio, Language and Image Processing, 2008. ICALIP 2008. International Conference on
Conference_Location :
Shanghai
Print_ISBN :
978-1-4244-1723-0
Electronic_ISBN :
978-1-4244-1724-7
DOI :
10.1109/ICALIP.2008.4590247