Doctor of Philosophy

dissertationHearing aids suffer from the problem of acoustic feedback that limits the gain provided by hearing aids. Moreover, the output sound quality of hearing aids may be compromised in the presence of background acoustic noise. Digital hearing aids use advanced signal processing to reduce acoustic feedback and background noise to improve the output sound quality. However, it is known that the output sound quality of digital hearing aids deteriorates as the hearing aid gain is increased. Furthermore, popular subband or transform domain digital signal processing in modern hearing aids 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. In this dissertation, we employ a variable, frequency-dependent gain function that is lower at frequencies of the incoming signal where the information is perceptually insignificant. In addition, the method of this dissertation automatically identifies and suppresses residual acoustical feedback components at frequencies that have the potential to drive the system to instability. The suppressed frequency components are monitored and the suppression is removed when such frequencies no longer pose a threat to drive the hearing aid system into instability. Together, the method of this dissertation provides more stable gain over traditional methods by reducing acoustical coupling between the microphone and the loudspeaker of a hearing aid. In addition, the method of this dissertation performs necessary hearing aid signal processing with low-delay characteristics. The central idea for the low-delay hearing aid signal processing is a spectral gain shaping method (SGSM) that employs parallel parametric equalization (EQ) filters. Parameters of the parametric EQ filters and associated gain values are selected using a least-squares approach to obtain the desired spectral response. Finally, the method of this dissertation switches to a least-squares adaptation scheme with linear complexity at the onset of howling. The method adapts to the altered feedback path quickly and allows the patient to not lose perceivable information. The complexity of the least-squares estimate is reduced by reformulating the least-squares estimate into a Toeplitz system and solving it with a direct Toeplitz solver. The increase in stable gain over traditional methods and the output sound quality were evaluated with psychoacoustic experiments on normal-hearing listeners with speech and music signals. The results indicate that the method of this dissertation provides 8 to 12 dB more hearing aid gain than feedback cancelers with traditional fixed gain functions. Furthermore, experimental results obtained with real world hearing aid gain profiles indicate that the method of this dissertation provides less distortion in the output sound quality than classical feedback cancelers, enabling the use of more comfortable style hearing aids for patients with moderate to profound hearing loss. Extensive MATLAB simulations and subjective evaluations of the results indicate that the method of this dissertation exhibits much smaller forward-path delays with superior howling suppression capability

The impact of spectrally asynchronous delay on the intelligibility of conversational speech

Conversationally spoken speech is rampant with rapidly changing and complex acoustic cues that individuals are able to hear, process, and encode to meaning. For many hearing-impaired listeners, a hearing aid is necessary to hear these spectral and temporal acoustic cues of speech. For listeners with mild-moderate high frequency sensorineural hearing loss, open-fit digital signal processing (DSP) hearing aids are the most common amplification option. Open-fit DSP hearing aids introduce a spectrally asynchronous delay to the acoustic signal by allowing audible low frequency information to pass to the eardrum unimpeded while the aid delivers amplified high frequency sounds to the eardrum that has a delayed onset relative to the natural pathway of sound. These spectrally asynchronous delays may disrupt the natural acoustic pattern of speech. The primary goal of this study is to measure the effect of spectrally asynchronous delay on the intelligibility of conversational speech by normal-hearing and hearing-impaired listeners. A group of normal-hearing listeners (n = 25) and listeners with mild-moderate high frequency sensorineural hearing loss (n = 25) participated in this study. The acoustic stimuli included 200 conversationally-spoken recordings of the low predictability sentences from the revised speech perception in noise test (r-SPIN). These 200 sentences were modified to control for audibility for the hearing-impaired group and so that the acoustic energy above 2 kHz was delayed by either 0 ms (control), 4ms, 8ms, or 32 ms relative to the low frequency energy. The data were analyzed in order to find the effect of each of the four delay conditions on the intelligibility of the final key word of each sentence. Normal-hearing listeners were minimally affected by the asynchronous delay. However, the hearing-impaired listeners were deleteriously affected by increasing amounts of spectrally asynchronous delay. Although the hearing-impaired listeners performed well overall in their perception of conversationally spoken speech in quiet, the intelligibility of conversationally spoken sentences significantly decreased when the delay values were equal to or greater than 4 ms. Therefore, hearing aid manufacturers need to restrict the amount of delay introduced by DSP so that it does not distort the acoustic patterns of conversational speech

Frequency Controlled Noise Cancellation for Audio and Hearing Purposes

Methods for hearing aids sought to compensate for loss in hearing by amplifying signals of interest in the audio band. In real-world, audio signals are prone to outdoor noise which can be destructive for hearing aid.  Eliminating interfering noise at high speed and low power consumption became a target for recent researches. Modern hearing compensation technologies use digital signal processing which requires minimum implementation costs to reduce power consumption, as well as avoiding delay in real time processing. In this paper, frequency controlled noise cancellation (FCNC) strategy for hearing aid and audio communication is developed with low complexity and least time delay. The contribution of the current work is made by offering a method that is capable of removing inherent distortion due filter-bank insertion and assigning adaptive filtering to a particular sub-band to remove external noise. The performance of the proposed FCNC was examined under frequency-limited noise, which corrupts particular parts of the audio spectrum. Results showed that the FCNC renders noise-immune audio signals with minimal number of computations and least delay. Mean square error (MSE) plots of the proposed FCNC method reached below -30 dB compared to -25 dB using conventional sub-band method and to -10 dB using standard full-band noise canceller. The proposed FCNC approach gave the lowest number of computations compared to other methods with a total of 346 computations per sample compared to 860 and 512 by conventional sub-band and full-band methods respectively. The time delay using FCNC is the least compared to the other methods

A low-power transmission-gate-based 16-bit multiplier for digital hearing aids

The most widespread 16-bit multiplier architectures are compared in terms of area occupation, dissipated energy, and EDP (Energy-Delay Product) in view of low-power low-voltage signal processing for digital hearing aids and similar applications. Transistor-level simulations including back-annotated wire parasitics confirm that the propagation of glitches along uneven and re-convergent paths results in large unproductive node activity. Because of their shorter full-adder chains, Wallace-tree multipliers indeed dissipate less energy than the carry-save (CSM) and other traditional array multipliers (6.0µW/MHz versus 10.9µW/MHz and more for 0.25µm CMOS technology at 0.75V). By combining the Wallace-tree architecture with transmission gates (TGs), a new approach is proposed to improve the energy efficiency further (3.1µW/MHz), beyond recently published low-power architectures. Besides the reduction of the overall capacitance, minimum-sized transmission gate full-adders act as RC-low-pass filters that attenuate undesired switching. Finally, minimum size TGs increase the V dd to ground resistance, hence decreasing leakage dissipation (0.55nW versus 0.84nW in CSM and 0.94nW in Wallace

Designs of low delay cosine modulated filter banks and subband amplifiers

This paper proposes a design of a low delay cosine modu-lated filter bank and subband amplifier coefficients for digi-tal audio hearing aids denoising applications. The objective of the design is to minimize the delay of the filter bank. Speci-fications on the maximum magnitude of both the real and the imaginary parts of the transfer function distortion and the aliasing distortion of the filter bank are imposed. Also, the constraint on the maximum absolute difference between the desirable magnitude square response and the designed mag-nitude square response of the prototype filter over both the passband and the stopband is considered. The subband am-plifier coefficients are designed based on a least squares training approach. The average mean square errors between the noisy samples and the clean samples is minimized. Com-puter numerical simulation results show that our proposed approach could significantly improve the signal-to-noise ratio of digital audio hearing aids