Design and Develop a PC-Based Hearing Aid Platform

• Main Authors: Jen-Jen Chang, 張人仁
• Other Authors: Kao-Chi Chung
• Format: Others
• Language: zh-TW
• Published: 2005
• Online Access: http://ndltd.ncl.edu.tw/handle/42291077736977366761

碩士 === 國立成功大學 === 醫學工程研究所碩博士班 === 93 === 　　An increasing of the elderly and serious noise pollution on living environments have led more and more people to suffer from hearing impairments with various types and different degrees of hearing loss. In Taiwan, the prevalence of hearing impairments are reported that about 18 % of adults between the ages of 65 and 74 years have hearing loss ,and approximately 35% of the person over age 75 have hearing loss. The statistical data of National Institute on Deafness and Other Communication Disorders shows that 54% of the people over age 65 have hearing loss. Only 20% of the hearing loss wear hearing aids in America, which indicates many problems of hearing aids needed to be solved at present, such as speech enhancement, echo cancellation and automatic gain control. The aim of this study is to design and develop a PC-based hearing aid platform for hearing evaluation and prescription on the hearing loss. 　　This PC-based platform is developed to include pure tone audiogram (PTA) module and prescriptive gain fitting module by using MATLAB 6.5 and Simulink /DSP blockset software packages. Pure tone sound is generated by temporal sinusoidal sequences multiplying with various amplitudes in PTA module .With speech analysis-synthesis theory, the speech signal is sampled and framed by Hamming window and equalized by FIR filter-banks (low-pass, band-pass and high-pass filters) in prescriptive gain fitting module . 　　The evaluation and prescription platform for hearing aid are completely developed. There are pure tones of 250, 500, 1k, 2k, 4k, 8k and 16k Hz in PTA module to measure subject’s hearing threshold. The prescriptive gain fitting module provides 10 sub-bands speech equalizers which frequency range from 20~20k Hz, and the volume gain fitted according to gain fitting strategy with maximum volume at 105 dB SPL. The frequency calibration of the platform shows the relative error is under 0.1% .The volume mapping table of the platform is completed. The calibration results of volume output in the system relative to sound level meter shows the mean value of correlation coefficients is 0.976. 　　The phonetically balanced word list (PBWL) consists of 50 words is developed by analyzing the phonetically balanced Chinese sentences. According to the PBWL, the speech intelligibility(SI) test on Eight adults with normal hearing shows that the mean and standard deviation of SI score is 92 ± 2.33%. 　　The future work is suggested as follow: (1) frequency division: the band splitting based on multirate-signal processing is likely to decrease computational complexity and increase efficiency；(2) noise reduction: increasing SNR by using adaptive filter algorithms；(3) running simulation in Simulink/DSP blockset on PC can generate C codes from the model and down-load the C codes into the platform for real-time simulation. Finally, simulation algorithm can be implemented by DSP board.