Low-Delay Signal Processing for Digital Hearing Aids

• Published in: IEEE transactions on audio, speech, and language processing speech, and language processing, 2011-05, Vol.19 (4), p.699-710
• Main Authors: Pandey, Ashutosh, Mathews, V John
• Format: Article
• Language: English
• Subjects: Adaptive filters; Applied sciences; Auditory system; Deafness; Delay; Detection, estimation, filtering, equalization, prediction; Digital signal processing; Ears & hearing; Exact sciences and technology; Filters; Frequency; Hearing aids; Information, signal and communications theory; low-delay; Miscellaneous; Performance analysis; Signal analysis; Signal and communications theory; Signal processing; Signal, noise; Telecommunications and information theory; Vents
• ISSN: 1558-7916; 2329-9290; 1558-7924; 2329-9304
• DOI: 10.1109/TASL.2010.2060193

Digital signal processing in modern hearing aids is typically performed in a subband or transform domain that introduces analysis-synthesis delays in the forward path. Long forward-path delays are not desirable because the processed sound combines with the unprocessed sound that arrives at the cochlea through the vent and changes the sound quality. Nonetheless, subband domain processing for digital hearing aids is the most popular choice for hearing aids because of the associated computational simplicity. In this paper, we present an alternative digital hearing aid structure with low-delay characteristics. The central idea in the paper is a low-delay spectral gain shaping method (SGSM) that employs parallel parametric equalization (EQ) filters. The low-delay SGSM provides frequency-dependent amplification for hearing loss compensation with low forward path delays and performs dynamic signal processing such as noise suppression and dynamic range compression. Parameters of the parametric EQ filters and associated gain values are selected using a least-squares approach to obtain the desired spectral response. The low-delay structure also employs an off-the-forward-path, frequency domain adaptive filter to perform acoustic feedback cancellation. Extensive MATLAB simulations and subjective evaluations of the results indicate that the method of this paper is competent with a state-of-the-art digital hearing aid system, but exhibits much smaller forward-path delays.