N. Subbulakshmi,R. Manimegalai,



Finite Impulse Response,Filter bank algorithms,Digital hearing aid,DSP algorithms,


Designing an electronic circuit with low power and small area are two important concerns for signal processing designers. Though fast emergence of the new technologies and several reviews over signal and speech processing, the difficulty cannot be fulfilled for the hearing impaired people. Many filter bank algorithms have been discussed on the hearing aid design to extend the efficiency. The conventional design of cascaded Direct Truncation (DT) data path is mainly based on the design of Full Precision Static Floating Point. In this paper, we introduce Static Floating Point Sample Rate Converter (SFP-SRC) with Linear Phase Finite Impulse Response (LPFIR) for hearing aid applications. The Sampling Rate Conversion is done before or after the LPFIR filter with upsampling and downsampling factors. In order to increase efficiency of DSP systems, filter bank algorithms need more than one sampling rate. The proposed method provides minimum delay and excellent Signal to Noise Ratio (SNR) performance when compared to Post Truncation (PT) data path. In order to obtain better performance, many experiments have been conducted. The proposed SFP-SRC is suitable for hearing assistance applications. Hence, it is implemented on 1/3 octave analysis filter bank with umc-90nm CMOS technology at 24 KHz.


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