Main Article Content
This paper presents adaptive filtering algorithm for processing noise signal which are present in ambient environment for digital hearing aid system. Conventional hearing aid system suffers from noise signal due to imbalance in the signal boost which degrades the original speech signal and reduce the capability of the hearing signals in noise environment. This can be reduced by utilizing adaptive filtering techniques in the FIR filter bank. However this method is very unstable since their weights are modified frequently. The proposed model design is written in Verilog and realized using Synopsys design compiler using 90 nm technology. The model is realized for 32, 64 and 128 adaptive filters coefficient lengths and are compared with traditional pipelined DA based adaptive FIR filter. The model achieves42.61% lesser area and 29.42% lesser ADP along65.13% power reduction and 41% of PDP compared with the traditional designs.