Variations in the feedback of hearing aids

Variations in the loop response of hearing aids caused by jaw movements, variations in acoustics outside the ear, and variations of vent size have been identified. Behind The Ear (BTE) and In The Ear Canal (ITEC) hearing aids were considered. The largest variations among the variations of the acoustics outside the ear, except when the hearing aid was partly removed, were found with the ITEC when a telephone set was placed by the ear. The variations of the loop response caused by changes in vent size were compared with the variations of a theoretical model of the feedback path. The theoretical model was also used to compare the feedback of different designs of the vent that gives the same acoustic impedance at low frequencies. The calculated feedback was less with the short vents (12 mm) than the long vents (24 mm).     

Constrained adaptation for feedback cancellation in hearing aids.

In feedback cancellation in hearing aids, an adaptive filter is used to model the feedback path. The output of the adaptive filter is subtracted from the microphone signal to cancel the acoustic and mechanical feedback picked up by the microphone, thus allowing more gain in the hearing aid. In general, the feedback-cancellation filter adapts on the hearing-aid input signal, and signal cancellation and coloration artifacts can occur for a narrow-band input. In this paper, two procedures for LMS adaptation with a constraint on the magnitude of the adaptive weight vector are derived. The constraints greatly reduce the probability that the adaptive filter will cancel a narrow-band input. Simulation results are used to demonstrate the efficacy of the constrained adaptation.     

Adaptive feedback cancellation in hearing aids with clipping in the feedback path

Adaptive linear filtering algorithms are commonly used to cancel feedback in hearing aids. The use of these algorithms is based on the assumption that the feedback path is linear, so nonlinearities in the feedback path may affect performance. This study investigated the effect on feedback canceller performance of clipping of the feedback signal arriving at the microphone, as well as the benefit of applying identical clipping to the cancellation signal so that the cancellation path modeled the nonlinearity of the feedback path. Feedback signal clipping limited the amount of added stable gain that the feedback canceller could provide, and caused misadjustment in response to high-level inputs, by biasing adaptive filter coefficients toward lower magnitudes. Cancellation signal clipping mitigated these negative effects, permitting higher amounts of added stable gain and less misadjustment in response to high-level inputs, but the benefit was reduced in the presence of the highest-level inputs.     

System identification of feedback in hearing aids.

The feedback problems of behind the ear (BTE), in the ear (ITE), and in the ear canal (ITEC) hearing aid categories have been investigated. All possible feedback paths (acoustical via vent, via tubing wall, mechanical, etc.) were converted to a single transfer function from the ear canal to the hearing aid microphone, here called the acoustic feedback equivalent (AFE). The attenuation of the AFE represents the maximum gain that can be used without the hearing aid starting to howl. Magnitude and phase responses of the AFE were identified on ten human subjects and on a Knowles ear manikin (KEMAR). The acoustic feedback via vent and leak between earmould and ear canal dominated the AFE. The transfer function from a reference point under the ear to the position of microphone of the different hearing aid categories was identified and used together with the AFE to calculate the maximum real ear aided gain (REAG) for the hearing aid categories. A model of the AFE, consisting of a fourth-order filter together with a delay, showed good agreement with the measured data.     

Extracting the invariant model from the feedback paths of digital hearing aids

Feedback whistling is a severe problem with hearing aids. A typical acoustical feedback path represents a wave propagation path from the receiver to the microphone and includes many complicated effects among which some are invariant or nearly invariant for all users and in all acoustical environments given a specific type of hearing aids. Based on this observation, a feedback path model that consists of an invariant model and a variant model is proposed. A common-acoustical-pole and zero model-based approach and an iterative least-square search-based approach are used to extract the invariant model from a set of impulse responses of the feedback paths. A hybrid approach combining the two methods is also proposed. The general properties of the three methods are studied using artificial datasets, and the methods are cross-validated using the measured feedback paths. The results show that the proposed hybrid method gives the best overall performance, and the extracted invariant model is effective in modeling the feedback path.     

A robust adaptive hybrid feedback cancellation scheme for hearing aids in the presence of outliers

It has been demonstrated that the Filtered-x Wilcoxon LMS (FxWLMS) based adaptive filter mitigates the effect of the outliers acquired by the microphone signal of hearing aids by minimizing the Wilcoxon norm and hence shows better cancellation performance than the existing Filtered-x LMS (FxLMS) algorithm. The prediction error method based adaptive feedback canceller (PEMAFC) reduces the bias present in the estimate of the feedback path due to the continuous adaptive filtering (CAF). However, the impulse response of the measured feedback path is close to zero for the first many samples due to the delay introduced by ADC converters and then contains few significant values, which results in slow convergence rate when an adaptive filter is used to model the same. To overcome this limitation, we propose a proportionate normalized WLMS (PNWLMS) algorithm based PEMAFC (P-PNWLMS) for feedback cancellation in hearing aid in the presence of outliers. Further, with an objective to improve the convergence rate and performance accuracy simultaneously, this paper proposes a novel convex PNWLMS (CPNWLMS) algorithm which incorporates convex combination of PNWLMS and WLMS algorithms. The weight update equations are derived for PEMAFC trained by PNWLMS (P-PNWLMS) and CPNWLMS (P-CPNWLMS) algorithms respectively. The results of the simulation study show improved performance of the proposed CPNWLMS based adaptive filter over its component filters.     

Study on the applicability of instrumental measures for black-box evaluation of static feedback control in hearing aids

Presented is a report on black-box evaluation of feedback control systems for commercial hearing aids. The aim of the study is to examine the ability of existing instrumental measures to quantify the performance of the feedback control system in black-box settings and on realistic signals, when more than one element of the signal processing chain may be active (compression, noise suppression, microphone directionality, etc.). The evaluation is carried out on 6 different hearing aids and for 10 measures. Thereby it is possible to see which measure is best suited to measuring which specific characteristic of the feedback control system, and serves as a beginning for conducting perceptual tests. The study uses static (but variable) feedback paths and is based on signals recorded from the in-ear microphone of an artificial head, on which the hearing instruments are mounted.     

The Pohlman cell as as a means of producing acoustic holograms

This paper reports the application of acoustic holography to a previously used non-destructive testing technique — the Pohlman Cell.     

A fast transfer function algorithm for nonuniform acoustic tubes

A fast algorithm for computing the transfer function for a nonuniform acoustic tube with acoustical losses and branches has been developed. It is based on the derived analytical correlations between the cross-section areas of such a tube and the coefficient of a corresponding autoregressive model with a moving average (ARMA-model). Comparison of transfer functions calculated with the use of the derived correlations with the results of a standard transmission-line method shows the accuracy and efficiency of the developed algorithm.     

Delayed feedback as a means of control of noise-induced motion

Time-delayed feedback is exploited for controlling noise-induced motion in coherence resonance oscillators. Namely, under the proper choice of time delay, one can either increase or decrease the regularity of motion. It is shown that in an excitable system, delayed feedback can stabilize the frequency of oscillations against variation of noise strength. Also, for fixed noise intensity, the phenomenon of entrainment of the basic oscillation period by the delayed feedback occurs. This allows one to steer the time scales of noise-induced motion by changing the time delay.     

Finite difference computation of head-related transfer function for human hearing

Modeling the head-related transfer function (HRTF) is a key to many applications in spatial audio. To understand and predict the effects of head geometry and the surrounding environment on the HRTF, a three-dimensional finite-difference time domain model (3D FDTD) has been developed to simulate acoustic wave interaction with a human head. A perfectly matched layer (PML) is used to absorb outgoing waves at the truncated boundary of an unbounded medium. An external source is utilized to reduce the computational domain size through the scattered-field/total-field formulation. This numerical model has been validated by analytical solutions for a spherical head model. The 3D FDTD code is then used as a computational tool to predict the HRTF for various scenarios. In particular, a simplified spherical head model is compared to a realistic head model up to about 7 kHz. The HRTF is also computed for a realistic head model in the presence of a wall. It is demonstrated that this 3D FDTD model can be a useful tool for spatial audio applications.     

Speech recognition in noise as a function of the number of spectral channels: comparison of acoustic hearing and cochlear implants

Speech recognition was measured as a function of spectral resolution (number of spectral channels) and speech-to-noise ratio in normal-hearing (NH) and cochlear-implant (CI) listeners. Vowel, consonant, word, and sentence recognition were measured in five normal-hearing listeners, ten listeners with the Nucleus-22 cochlear implant, and nine listeners with the Advanced Bionics Clarion cochlear implant. Recognition was measured as a function of the number of spectral channels (noise bands or electrodes) at signal-to-noise ratios of + 15, + 10, +5, 0 dB, and in quiet. Performance with three different speech processing strategies (SPEAK, CIS, and SAS) was similar across all conditions, and improved as the number of electrodes increased (up to seven or eight) for all conditions. For all noise levels, vowel and consonant recognition with the SPEAK speech processor did not improve with more than seven electrodes, while for normal-hearing listeners, performance continued to increase up to at least 20 channels. Speech recognition on more difficult speech materials (word and sentence recognition) showed a marginally significant increase in Nucleus-22 listeners from seven to ten electrodes. The average implant score on all processing strategies was poorer than scores of NH listeners with similar processing. However, the best CI scores were similar to the normal-hearing scores for that condition (up to seven channels). CI listeners with the highest performance level increased in performance as the number of electrodes increased up to seven, while CI listeners with low levels of speech recognition did not increase in performance as the number of electrodes was increased beyond four. These results quantify the effect of number of spectral channels on speech recognition in noise and demonstrate that most CI subjects are not able to fully utilize the spectral information provided by the number of electrodes used in their implant.     

Evaluation of two multichannel tactile aids for the hearing impaired

Two multichannel tactile devices for the hearing impaired were compared in speech perception tasks of varying levels of complexity. Both devices implemented the "vocoder" principle in their stimulus processing: One device had a 16-element linear vibratory array worn on the forearm and displayed activity in 16 overlapping frequency channels; the other device delivered tactile stimulation to a linear array of 16 electrodes worn on the abdomen. Subjects were tested in several phoneme discrimination tasks, ranging from discrimination of pairs of words differing in only one phoneme under tactile aid alone conditions to identification of stimuli in a larger set under tactile aid alone, lipreading alone, and lipreading plus tactile aid conditions. Results showed both devices to be better transmitters of manner and voicing features of articulation than of place features, when tested in single-item tasks. No systematic differences in performance with the two devices were observed. However, in a connected discourse tracking task, the vibrotactile vocoder in conjunction with lipreading yielded much greater improvements over lipreading alone than did the electrotactile vocoder. One possible explanation for this difference in performance, the inclusion of a noise suppression circuit in the electrotactile aid, was evaluated, but did not appear to account for the differences observed. Results are discussed in terms of additional differences between the two devices that may influence performance.     

A technique for simulating the amplifier-to-eardrum transfer function of an in situ hearing aid

There are numerous articles wherein mathematical models of various parts of an in situ hearing aid have been reported. Such parts include, for example, the microphone, receiver, cylindrical tubes carrying sound to the eardrum and out through the earmold vent, and the external path from the vent back to the microphone. This article extends these earlier works to include the hearing-aid amplifier. In particular, a mathematical technique for characterizing the amplifier in combination with the receiver is reported. Cascade parameters of a two-port model of one particular amplifier/receiver combination are obtained by this method. The cascade-parameter data and the method of obtaining this data are verified by two different experimental procedures. One procedure involves both computing and measuring the input driving-point impedance of the amplifier/receiver combination. In the second procedure, the amplifier-to-eardrum transfer function of a hearing aid incorporating this same amplifier/receiver combination and mounted on an artificial ear is both computed and measured. Experimental and computed values of this transfer function for three different earmold geometries are in reasonably close agreement. The amplifier/receiver model reported herein will be used in future studies of acoustic feedback in hearing aids.     

The negative effect of amplitude compression in multichannel hearing aids in the light of the modulation-transfer function

The article deals with the question of why multichannel amplitude compression appears to have a negative rather than a positive effect on speech intelligibility by hearing-impaired listeners. It is argued that the small time constants of amplitude compression diminish the temporal as well as the spectral contrasts in the speech signal. According to the modulation-transfer function concept, this results in reduced intelligibility scores. Experimental evidence is reviewed indicating that the following two arguments in favor of amplitude compression in case of sensorineural hearing loss are not valid: (1) to compensate for the effects of loudness recruitment and (2) to get weak consonants above threshold. The author concludes that, in multichannel hearing aids, automatic gain control with time constants of 0.25-0.5 s should be given preference to amplitude compression.     

Cross-correlation procedures for measuring noise and distortion in AGC hearing aids

The magnitude-squared coherence function (MSC) has been used to measure noise and distortion in linear and compression hearing aids. However, the MSC will overestimate the distortion in a linear time-varying system such as a compression amplifier. The reduction in coherence caused by varying the gain in an otherwise linear system can be substantial, and can lead to large errors in estimating the distortion present in a compression hearing aid. The effects of gain changes in a linear system can be reduced by measuring the normalized system input-output cross correlation, which emphasizes the variance in the system phase response and deemphasizes the system gain fluctuations. Estimates of the total noise and distortion produced using the MSC, phase variance, and notched-noise measurement techniques are compared for additive noise, clipping distortion, and compression amplification. The MSC is found to give the most accurate results for estimating the noise and distortion in a linear time-invariant system, and the notched noise measurements are the most accurate for a compression system. The phase variance is found to give reasonable measurements for a time-varying gain as long as the system variations are slow relative to the length of the analysis data segments.     

The acoustic analysis of tone differentiation as a means for assessing tone production in speakers of Cantonese

This paper reports on a methodology for acoustically analyzing tone production in Cantonese. F0 offset versus F0 onset are plotted for a series of tokens for each of the six tones in the language. These are grouped according to tone type into a set of six ellipses. Qualitative visual observations regarding the degree of differentiation of the ellipses within the tonal space are summarized numerically using two indices, referred to here as Index 1 and Index 2. Index 1 is a ratio of the area of the speaker's tonal space and the average of the areas of the ellipses of the three target tones making up the tonal space. Index 2 is a ratio of the average distance between all six tonal ellipses and the average of the sum of the two axes for all six tone ellipses. Using this methodology, tonal differentiation is compared for three groups of speakers; normally hearing adults; normally hearing children aged from 4-6 years; and, prelinguistically deafened cochlear implant users aged from 4-11 years. A potential conundrum regarding how tone production abilities can outstrip tone perception abilities is explained using the data from the acoustic analyses. It is suggested that young children of the age range tested are still learning to normalize for pitch level differences in tone production. Acoustic analysis of the data thus supports results from tone perception studies and suggests that the methodology is suitable for use in studies investigating tone production in both clinical and research contexts.     

Influence of acoustic impedance of multilayer acoustic systems on the transfer function of ultrasonic airborne transducers

In different solutions of ultrasonic transducers radiating acoustic energy into the air there occurs the problem of the proper selection of the acoustic impedance of one or more matching layers. The goal of this work was a computer analysis of the influence of acoustic impedance on the transfer function of piezoceramic transducers equipped with matching layers. Cases of resonance and non-resonance matching impedance in relation to the transfer function and the energy transmission coefficient for solid state-air systems were analysed. With stable thickness of matching layers the required shape of the transfer function can be obtained through proper choice of acoustic impedance were built (e.g. maximal flat function). The proper choice of acoustic impedance requires an elaboration of precise methods of synthesis of matching systems. Using the known matching criteria (Chebyshev's, DeSilets', Souquet's), the transfer function characteristics of transducers equipped with one, two, and three matching layers as well as the optimisation methods of the energy transmission coefficient were presented. The influence of the backside load of the transducer on the shape of transfer function was also analysed. The calculation results of this function for different loads of the transducer backside without and with the different matching layers were presented. The proper load selection allows us to obtain the desired shape of the transfer function, which determines the pulse shape generated by the transducer.     

Frequency resolution as a function of hearing threshold level and age

Frequency resolution ability was measured using a psychoacoustical tuning curve (PTC) or a notch-noise technique in two population samples. The first sample incorporated 1764 subjects with various degrees of sensorineural hearing impairment and ranging in age from 17-80 years. The second sample included 240 subjects aged between 50 and 75 years, carefully balanced in terms of impairment and age to avoid confounding between these two variables. In both samples, frequency resolution ability declined with increasing hearing threshold level (HTL), as measured by either method. In a subsample tested with both methods, the correlation between the two was only modest. After accounting for HTL, there was a minor dependence of frequency resolution on age, older subjects having poorer frequency resolution once HTL had been accounted for. No addition to the explained variance was achieved by taking sex, occupational group, or audiogram slope into account. Despite the documented reproducibility of the measures, much of the variance in the frequency resolution measurements remained unrelated to HTL or age.     

The ear effect as a function of age and hearing loss

Many studies have shown that the right ear statistically is slightly more sensitive than the left ear, particularly in the male adult population. In this study, we examined the lateral difference in hearing sensitivity, termed the ear effect here, in an industrial noise-exposed, nonshooting population, by sex, age, and hearing level. It was found that the male population had a larger ear effect (right ear being more sensitive) than the female population. The magnitude of the ear effect was found to be significantly related to the hearing threshold level. The ear effect was highest when the threshold was between 30- and 40-dB HL. Several possible causes for the ear effect are discussed.