Digital hearing aids identify acoustic feedback signals and cancel them continuously in a closed loop with an adaptive filter. This scheme facilitates larger hearing aid gain and improves the output sound quality of hearing aids. However, the output sound quality deteriorates as the hearing aid gain is increased. This paper presents two methods to modify the forward path gain in digital hearing aids. The first approach employs a variable, frequency-dependent gain function that is lower at frequencies of the incoming signal where the information is perceptually insignificant. The second method of this paper 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 gain processing methods of this paper provide 8 to 12 dB more hearing aid gain than feedback cancelers with fixed gain functions. Furthermore, experimental results obtained with real world hearing aid gain profiles indicate that the gain processing methods of this paper, individually and combined, provide 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.

Acoustic feedback limits the gain provided by hearing aids. Digital hearing aids identify acoustic feedback signals and cancel them continuously in a closed loop with an adaptive filter in the digital domain. This scheme facilitates larger hearing aid gain and improves the output sound quality of hearing aids. However, the output sound quality of hearing aids deteriorates as the hearing aid gain is increased. This paper 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. Experimental results obtained with real world hearing aid gain profiles using speech and music signals indicate that the method of this paper 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.

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.

Adaptive filters are commonly used to cancel acoustic feedback in hearing aids. The sound quality of hearing aids deteriorates as the hearing aid gain is increased. This paper presents a method that increases gain possible in hearing aids by automatically identifying and suppressing residual acoustical feedback components that have the potential to drive the system to instability. The increase in the hearing aid gain over traditional methods and the output sound quality were evaluated using subjective tests. The results indicate that the method of this paper provides 6 to 8 dB more hearing aid gain and less perceptual distortion in the output sound quality than several competing algorithms to improve the performance of adaptive feedback cancellers.

Adaptive filters are commonly used to cancel acoustic feedback in hearing aids. The sound quality of hearing aids deteriorates as the hearing aid gain is increased. This paper presents a method to alter the gain function in digital hearing aids to provide additional amplification and better output sound quality. This approach employs a variable, frequency-dependent gain function that is lower at frequencies of the incoming signal where the information is perceptually insignificant. The increase in stable gain over traditional methods and the output sound quality were evaluated with a psychoacoustic experiment on normal-hearing listeners. The results indicate that the method of this paper provides more hearing aid gain and less distortion in the output sound quality than feedback cancelers with fixed gain functions.

Howling is a significant problem even in digital hearing aids equipped with adaptive feedback cancellation. Among the many causes of howling is the inability of the adaptive filter to track rapid changes in the feedback path. Many systems use howling detectors to detect the start of howling and reduce the hearing aid gain for several seconds to avoid prolonged howling. Unfortunately the inadequate speech pressure levels (SPL) during times when the gain is reduced causes loss of information and reduced intelligibility of speech signals arriving at the patient's ears. This paper presents a new method that 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. In addition, the gain function is changed immediately after howling detection in such a way that the system operates in a stable manner and the distortions caused are not perceived because of temporal masking. Simulation results comparing with a conventional method is presented in the paper to demonstrate the superior howling suppression capabilities of the method