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Biomedical subjects

J M Kates

Publications and source records attributed to J M Kates.

At least 19 recordsLinked to original sources

Speech intelligibility enhancement using hearing-aid array processing.

Microphone arrays can improve speech recognition in the noise for hearing-impaired listeners by suppressing interference coming from other than desired signal direction. In a previous paper [J. M. Kates and M. R. Weiss, J. Acoust. Soc. Am. 99, 3138-3148 (1996)], several array-processing techniques were evaluated in two rooms using the AI-weighted array gain as the performance metric. The array consisted of five omnidirectional microphones having uniform 2.5-cm spacing, oriented in the endfire direction. In this paper, the speech intelligibility for two of the array processing techniques, delay-and-sum beamforming and superdirective processing, is evaluated for a group of hearing-impaired subjects. Speech intelligibility was measured using the speech reception threshold (SRT) for spondees and speech intelligibility rating (SIR) for sentence materials. The array performance is compared with that for a single omnidirectional microphone and a single directional microphone having a cardioid response pattern. The SRT and SIR results show that the superdirective array processing was the most effective, followed by the cardioid microphone, the array using delay-and-sum beamforming, and the single omnidirectional microphone. The relative processing ratings do not appear to be strongly affected by the size of the room, and the SRT values determined using isolated spondees are similar to the SIR values produced from continuous discourse.

Aged

Quality ratings for frequency-shaped peak-clipped speech: results for listeners with hearing loss.

Peak clipping is a common form of distortion in hearing aids and can reduce the subjective quality of the amplified speech. In a previous study involving listeners with normal hearing (Kates & Kozma-Spytek, 1994), the effect of peak clipping on speech quality ratings was studied using sentence test materials that were filtered using three different frequency response contours and then clipped at four different clipping levels. The present study extends the quality ratings to include those from a group of listeners having moderate to profound hearing impairments. The experimental results indicate that the clipping level, and the interaction of the frequency-response shaping with the clipping level, significantly affects speech quality. It is also shown that the distortion effects on speech quality for the listeners with impaired hearing can be modeled by a distortion index computed from the magnitude-squared coherence of the speech-processing system in response to a shaped-noise input signal. The distortion-index weights derived for the group of listeners with impaired hearing, however, differ substantially from those derived for listeners with normal hearing, and substantial inter-listener variation was also observed.

Audiometry, Speech

A comparison of hearing-aid array processing techniques.

Microphone arrays have proven effective in improving speech intelligibility in noise for hearing-impaired listeners, and several array processing techniques have been proposed for hearing aids. Among the signal-processing approaches are classical delay-and-sum beamforming, superdirective arrays, and adaptive arrays. To directly compare the effectiveness of these different processing strategies, a 10-cm-long linear array was built using five uniformly spaced omnidirectional microphones. This array was used in the end-fire orientation to acquire speech and noise signals for a variety of array placements in two representative rooms. Both digital and simulated analog processing techniques were considered, with the array processing implemented in the frequency domain. The performance metric was the steady-state array gain weighted to represent the relative importance of the different frequency regions in understanding speech. The processing comparison indicates that digital systems are more effective than the simulated analog processing, and that both superdirective and adaptive digital array processing can provide more than 9 dB of weighted array gain.

Deafness

Classification of background noises for hearing-aid applications.

A background-noise classification procedure is being developed for hearing-aid applications, wherein the hearing-aid response would be adjusted in response to changes in the noise environment. The classification procedure is based on measuring four signal features giving the fluctuations of the signal envelope and the mean frequency and low- and high-frequency slopes of the average spectrum. A more complicated procedure, based on determining the envelope modulation spectra in auditory critical bands, was also investigated and was found to offer no advantages over the simpler procedure. The accuracy of the classification procedure was determined for eleven everyday background noises under optimal conditions where the training and test noise sequences were different portions of the same short noise recording. A cluster analysis was used to determine the similarities among the feature vectors for the noises, and when the noises are grouped into a reduced number of clusters the noise-classification accuracy using the four features exceeds 90%.

Cluster Analysis

On the feasibility of using neural nets to derive hearing-aid prescriptive procedures.

A neural net is a "black box" information processing system that can be used for pattern matching, optimal prediction, or functional approximation. A neural net requires a minimal amount of a priori knowledge about the problem to be solved, but can require large amounts of data to converge to a solution. For a hearing-aid fitting procedure, a multilayer perceptron net was trained to generate an optimum match between a set of input pure-tone audiograms and the corresponding best frequency response and gain for each subject. The feasibility of using neural nets to select hearing-aid response characteristics was tested using both simulated and real audiometric data. The simulation results indicate that a neural net can be successfully trained to reproduce a fitting rule such as the NAL-R procedure, and that a minimum of about 50 sets of audiometric response data are needed for the net to converge to a generalized solution. When used to predict, from the pure-tone audiograms, the best frequency response characteristics determined for subjects having severe-to-profound hearing losses, the neural net was more accurate than the NAL-R fitting procedure derived from the same data.

Audiometry, Pure-Tone

Speech enhancement based on a sinusoidal model.

Sinusoidal modeling is a new procedure for representing the speech signal. In this approach, the signal is divided into overlapping segments, the Fourier transform computed for each segment, and a set of desired spectral peaks is identified. The speech is then resynthesized using sinusoids that have the frequency, amplitude, and phase of the selected peaks, with the remaining spectral information being discarded. Using a limited number of sinusoids to reproduce speech in a background of multi-talker speech babble results in a speech signal that has an improved signal-to-noise ratio and enhanced spectral contrast. The more intense spectral components, assumed to be primarily the desired speech, are reproduced, whereas the less intense components, assumed to be primarily background noise, are not. To test the effectiveness of this processing approach as a noise suppression technique, both consonant recognition and perceived speech intelligibility were determined in quiet and in noise for a group of subjects with normal hearing as the number of sinusoids used to represent isolated speech tokens was varied. The results show that reducing the number of sinusoids used to represent the speech causes reduced consonant recognition and perceived intelligibility both in quiet and in noise, and suggests that similar results would be expected for listeners with hearing impairments.

Acoustic Stimulation

Quality ratings for frequency-shaped peak-clipped speech.

Peak clipping is a common form of distortion in hearing aids and can reduce the subjective quality of the amplified speech. In a typical hearing aid, frequency-response shaping precedes symmetric peak clipping. The effect of peak clipping on speech quality ratings was therefore studied using sentence test materials that were processed using different frequency response contours and then clipped at different clipping thresholds. The quality of each processed sentence was rated on a ten-point scale by normal-hearing subjects. The experimental results indicate that the clipping threshold, and the interaction of the frequency-response shaping with the clipping threshold, significantly affect speech quality. It is also shown that the distortion effects on speech quality can be modeled by a distortion index computed from the magnitude-squared coherence of the speech-processing system in response to a shaped-noise input signal.

Acoustic Stimulation

Optimal estimation of hearing-aid compression parameters.

A new procedure for measuring the attack and release time constants of a hearing-aid compression circuit is presented in this paper. The procedure is based on a mathematical model of the response of a compression system to sudden increments or decrements in the amplitude of a sinusoidal excitation. The parameters of the model, which include the attack and release time constants, are fitted to the measured hearing-aid test-signal response using a minimum mean-squared error criterion. A computer simulation of a compression hearing aid is used to illustrate the behavior of the compression amplifier and to assess the accuracy of the estimated attack and release time constants.

Amplifiers, Electronic

Superdirective arrays for hearing aids.

Microphone arrays are the most effective of the techniques that have been proposed for improving speech intelligibility in noise for the hearing impaired. However, classical delay-and-sum beamforming provides very small amounts of array gain at low frequencies, while adaptive array processing has been shown to cancel the desired signal in the presence of strong room reflections. Superdirective arrays offer a heretofore overlooked solution in which optimal performance can be obtained for a stationary random noise field, but where the desired signal will not be canceled. A short constrained superdirective array suitable for hearing-aid applications is proposed in this paper, and its theoretical performance is evaluated.

Correction of Hearing Impairment

Toward a theory of optimal hearing aid processing.

An ideal hearing aid for a peripheral hearing loss would process the incoming signal in order to give a perfect match between the cochlear outputs of the impaired ear and a reference normal ear. As a first step toward this objective, a model of the normal and impaired peripheral auditory system was used to derive the optimal hearing-aid processing filter based on a minimum mean-squared error criterion. The auditory model includes the compression and suppression effects of the cochlear mechanics and the sensitivity of the neural transduction process. Simplifying assumptions were then incorporated into the processing to yield a practical frequency-dependent adaptive gain system. Processing examples of several individual speech sounds are presented for a flat hearing loss, and the results indicate that a three-channel compression system with adjustable gains and band edges will be close to the optimal solution for this case.

Adaptation, Physiological

On using coherence to measure distortion in hearing aids.

Coherence is a frequency-domain measure of the degree to which the output of a system is linearly related to the system input. The signal-to-distortion ratio (SDR), where the distortion term includes all nonlinear effects and noise in the system, can be computed from the coherence. The coherence estimate, however, is subject to sources of variance and bias that reduce the accuracy of the measured SDR. The origins of the variance and bias and their effects on distortion measurements are presented. New procedures for reducing the variance and bias effects are described, and the processing effectiveness is demonstrated for a simulated hearing-aid response.

Fourier Analysis

The problem of feedback in hearing aids.

One of the factors that limits hearing-aid performance is feedback. This paper discusses the problem of feedback in hearing aids, illustrated with examples based on a computer simulation of hearing-aid behavior. The available technology for dealing with the problem of feedback is then reviewed, and new digital signal-processing approaches are described that may finally solve the feedback problem.

Equipment Design

Modeling normal and impaired hearing: implications for hearing aid design.

A cochlear simulation has been developed to model normal and impaired hearing. The simulation includes the middle ear, the mechanical motion of the cochlear partition, and the mechanical to neural transduction of the inner hair cells. The outer hair cells are postulated to provide an active feedback mechanism that adjusts the gain and shape of the auditory filters. Auditory impairment is simulated by reducing the efficacy of the outer hair cells and by modifying the inner hair cell transduction process. The effects of simulated impairment are illustrated for two speech sounds, /ba/ and /ka/, with the neural firing patterns from the impaired ear compared with those of a normal ear. The differences in the neural firing patterns are interpreted in the context of hearing aid signal processing.

Cochlea

A test suite for hearing aid evaluation.

A test suite has been developed for evaluating hearing aids. The tests in the suite are frequency response, number of processing bands and type of processing, input/output characteristics, processing attack and release times, and broadband distortion. The test suite produces a more complete evaluation of a hearing aid than any previous set of tests, and is suitable for the automatic evaluation of a hearing aid containing unknown processing. The test procedures are described, and sample test results are presented for simulated linear and two-channel compression hearing aids.

Acoustics

A time-domain digital simulation of hearing aid response.

A time-domain digital simulation of an in-the-ear (ITE) hearing aid has been developed. The stimulation allows modeling of nonlinear effects such as compression and amplifier distortion in addition to linear processing and acoustics. The simulation includes a microphone, two-channel compression processing, an amplifier with clipping distortion, a receiver, an ear canal and ear drum, and feedback and direct sound transmission through the vent. Simulation results for a linear hearing aid are similar to those obtained for frequency-domain representations of the analog system. Examples of responses for nonlinear systems are also provided.

Acoustics

Acoustic effects in in-the-ear hearing aid response: results from a computer simulation.

The response of a hearing aid depends on the design of the instrument and on the characteristics of the individual ear. In this paper a computer simulation of an in the ear (ITE) hearing aid is used to determine the effects on the hearing aid response caused by variations in the size of the ear canal, the magnitude of the eardrum impedance, and the vent size and damping. The simulation results indicate that, for an unvented hearing aid, changes in the size of the ear canal or the eardrum impedance shift the average sound pressure level at the eardrum but have relatively small effects on the overall shape of the frequency response. A vented instrument presents a more complicated situation since the vent modifies the low-frequency response in a predictable manner but can have unexpectedly pronounced effects at high frequencies due to the acoustic feedback.

Acoustic Impedance Tests

A computer simulation of hearing aid response and the effects of ear canal size.

The response of a hearing aid is affected by many factors which include the head and outer ear, the microphone, amplifier, and receiver used in the hearing aid, the properties of the ear canal and the eardrum, and acoustic feedback through the vent. This article presents a computer simulation of an in-the-ear (ITE) hearing aid that includes all of the above factors. The simulation predicts the pressure at the eardrum for a frontal free-field sound source. The computer model was then used to determine the effects on the hearing aid response due to variations in the size of the ear canal. The simulation indicates that, for an unvented hearing aid, changes in the size of the ear canal shift the overall sound-pressure level at the eardrum but have only small effects on the shape of the frequency response. The situation is more complicated when a vent is present, however, since changes in the size of the ear canal that cause apparently small perturbations in the acoustic feedback signal may, nonetheless, have large effects on the overall system response.

Computer Simulation

The short-time articulation index.

In this paper we introduce the concept of the short-time articulation index. This is a procedure for calculating a time-varying articulation index from data on a block-by-block basis. The short-time articulation index can be used to give a running measure of the speech intelligibility for an adaptive noise-cancellation system as it converges. We present an algorithm for calculating the short-time articulation index and give some examples of its use.

Acoustics