Evaluation of array-processing algorithms for a headband hearing aid

Several array-processing algorithms were implemented and evaluated with experienced hearing-aid users. The array consisted of four directional microphones mounted broadside on a headband worn on the top of the listener's head. The algorithms included two adaptive array-processing algorithms, one fixed array-processing algorithm, and a reference condition consisting of binaural directional microphones. The algorithms were evaluated under conditions with both one and three independent noise sources. Performance metrics included quantitative speech reception thresholds and qualitative subject preference ratings for ease-of-listening measured using a paired-comparison procedure. On average, the fixed algorithm improved speech reception thresholds by 2 dB, while the adaptive algorithms provided 7-9-dB improvement over the reference condition. Subjects judging ease-of-listening generally preferred all array-processing algorithms over the reference condition. The results suggest that these adaptive algorithms should be evaluated further in more realistic acoustic environments.     

Signal-processing techniques in a computerized hearing aid test system

General background is given to describe the factors leading up to the implementation of a computerized hearing aid test system in a production environment. The digital measurement methods for determination of the required acoustal components such as fundamental, rms, distortion, etc., in the presence of noise are discussed. The use of the concept of the average cycle for repetitive signals is described with its advantages in a Fourier type system. An algorithm for measuring rms for nonrepetitive signals that trades off resolution for memory size is described. Features and advantages of computerized testing of hearing aids are listed.     

Effect of handset proximity on hearing aid feedback

The increased sensitivity of hearing aids to feedback as a telephone handset is brought near has been studied experimentally and numerically. For the measurements, three different hearing aids were modified so that the open-loop transfer function could be measured. They were mounted in the pinna of a mannikin and the change in open-loop transfer function determined as a function of handset proximity. Increases of over 20 dB were observed, most of this change occurring within the first 10 mm of separation between pinna and handset. Numerical calculations performed using a boundary element technique were in good agreement with the measurements.     

Room reverberation effects in hearing aid feedback cancellation

Room reverberation can affect feedback cancellation in hearing aids, with the strength of the effects depending on the acoustical conditions. These effects were studied using a behind the ear (BTE) hearing aid mounted on a dummy head and coupled to the ear canal via an open fitting. The hearing aid impulse response was measured for the dummy head placed at eight closely spaced locations in a typical office. The feedback cancellation in the hearing aid used a set of filter coefficients that were initialized for one location within the room, and then allowed to adapt to the feedback path measured at the same or to a different location. The maximum stable gain for the hearing aid was then estimated without feedback cancellation, for the initial set of feedback cancellation filter coefficients prior to adaptation, and for the feedback cancellation filter after adaptation. A low-order ARMA model combining a fixed set of poles with an adaptive FIR filter is shown to be effective in representing the feedback path exclusive of reverberation. Increasing the adaptive filter length has only a small benefit in improving the feedback cancellation performance due to the inability of the system to model the room reverberation. The mismatch between the modeled and actual feedback paths limits the headroom increase that can be achieved when using feedback cancellation, and varies with the location within the room.     

An evaluation of three adaptive hearing aid selection strategies

Paired-comparison judgments of intelligibility of speech in noise were obtained from eight hearing-impaired subjects on a large number of hearing aids simulated by a digital master hearing aid. The hearing aids which comprised a 5 X 5 matrix differed systematically in the amount of low-frequency and high-frequency gain provided. A comparison of three adaptive strategies for determining optimum hearing aid frequency-gain characteristics (an iterative round robin, a double elimination tournament, and a modified simplex procedure) revealed convergence on the same or similar hearing aids for most subjects. Analysis revealed that subjects for whom all three procedures converged on the same hearing aid showed a single pronounced peak in the response surface, while a broader peak was evident for the subjects for whom the three procedures identified similar hearing aids. The modified simplex procedure was found to be most efficient and the iterative round robin least efficient.     

Transverse pressure distributions in a simple model ear canal occluded by a hearing aid test fixture

The sound field in a model ear canal with a hearing aid test fixture has been investigated experimentally and theoretically. Large transverse variations of sound pressure level, as much as 20 dB at 8 kHz, were found across the inner face of the hearing aid. Variations are greatest near the outlet port of the receiver and the vent port. Deeper into the canal, the transverse variations are less significant and, at depths greater than 4 mm, only a longitudinal variation remains. The model canal was cylindrical, 7.5 mm diameter, and terminated with a Zwislocki coupler to represent absorption by the human middle ear. The outer end of the canal was driven by the receiver in the hearing aid test fixture, with the acoustic output entering the canal through a 1 mm port. The hearing aid was provided with a 20-mm-long vent, either 1 or 2 mm in diameter. The sound field inside the canal was measured using a specially designed 0.2-mm-diam probe microphone [Daigle and Stinson, J. Acoust. Soc. Am. 116, 2618 (2004)]. In parallel, calculations of the interior sound field were performed using a boundary element technique and found to agree well with measurements.     

Advantage of bimodal fitting in prosody perception for children using a cochlear implant and a hearing aid

Cochlear implants are largely unable to encode voice pitch information, which hampers the perception of some prosodic cues, such as intonation. This study investigated whether children with a cochlear implant in one ear were better able to detect differences in intonation when a hearing aid was added in the other ear ("bimodal fitting"). Fourteen children with normal hearing and 19 children with bimodal fitting participated in two experiments. The first experiment assessed the just noticeable difference in F0, by presenting listeners with a naturally produced bisyllabic utterance with an artificially manipulated pitch accent. The second experiment assessed the ability to distinguish between questions and affirmations in Dutch words, again by using artificial manipulation of F0. For the implanted group, performance significantly improved in each experiment when the hearing aid was added. However, even with a hearing aid, the implanted group required exaggerated F0 excursions to perceive a pitch accent and to identify a question. These exaggerated excursions are close to the maximum excursions typically used by Dutch speakers. Nevertheless, the results of this study showed that compared to the implant only condition, bimodal fitting improved the perception of intonation.     

Computer models of hearing aid transducers for integrated circuit design.

Electronic circuit modeling using computer-based simulation tools is well established and device models are available for common electronic components. However, acoustic models of audio transducers for use during integrated circuit design are not readily available. This causes difficulty for designers of audio amplifiers, and increases the uncertainties of a successful silicon integration of a circuit design. This paper reports on a technique for the creation of electroacoustic models of hearing aid microphones and receivers that can be connected to an amplifier under design, and incorporated into PSPICE simulations. Verification of the technique and models was performed by comparing measured frequency response data with graphs created by PSPICE modeling. The conclusions were that the method developed for creating these models, and the models themselves, were accurate enough to be used for acoustic simulations of frequency response performance during amplifier design, and gave results comparable to data obtained from breadboard measurements of the same circuits.     

Intelligibility ratings of continuous discourse: application to hearing aid selection

Twelve normal-hearing subjects rated the intelligibility of 35-s, hearing-aid-processed continuous discourse (CD) passages. Three talkers (two male, one female), four hearing aids, and two signal-to-babble (S/B) ratios were used in a completely crossed design. Research questions concerned: (1) ability of listeners to rate intelligibility, (2) sensitivity of hearing aid rankings were based on intelligibility ratings for three CD passages per instrument, and (3) dependence of hearing aid rankings on (a) S/B ratio, and (b) talker characteristics. Results were: (1) listeners were able to rate intelligibility, (2) rankings based on intelligibility ratings of three CD passages per hearing aid were capable of identifying two superior instruments within a group of four hearing aids that were similar in frequency/gain function, (3) listening in a more difficult S/B ratio substantially decreased the sensitivity of the hearing aid rankings for the female talker but had only minor effects on the rankings for the male talkers, and (4) hearing aid intelligibility rankings were found to be different for different talkers. Applications to hearing aid selection are discussed.     

The hearing aid feedback path: mathematical simulations and experimental verification

Acoustic feedback in hearing aids has received little attention in the literature. Feedback occurs when stability conditions of the open-loop transfer function of an in situ hearing aid are violated. Solving the feedback problem will first require knowledge of the open-loop transfer function. Included in the open-loop transfer function is the acoustical path by which sound emanating from the earmold vent returns to the microphone (i.e., the feedback path). Reported herein are two different mathematical procedures for simulating transfer functions of the feedback path of an eyeglass-type hearing aid. In one procedure the vent exit was modeled as a point source of sound located on a flat plane, while it was treated as a point source on a sphere in the other. Results of laboratory experiments indicate that the mathematical models accurately predict those acoustic phenomena for which they were intended: point sources on plane and spherical baffles. Results of manikin experiments showed both models to be less accurate for simulating the feedback path around the human head. The maximum difference between experiment and theory was 6 dB at one frequency. Surprisingly, the flat-baffle model produced better agreement with experimental results than did the sphere model.     

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.     

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.     

Performance of an adaptive beamforming noise reduction scheme for hearing aid applications. II. Experimental verification of the predictions

A method to predict the amount of noise reduction which can be achieved using a two-microphone adaptive beamforming noise reduction system for hearing aids [J. Acoust. Soc. Am. 109, 1123 (2001)] is verified experimentally. 34 experiments are performed in real environments and 58 in simulated environments and the results are compared to the predictions. In all experiments, one noise source and one target signal source are present. Starting from a setting in a moderately reverberant room (reverberation time 0.42 s, volume 34 m3, distance between listener and either sound source 1 m, length of the adaptive filter 25 ms), eight different parameters of the acoustical environment and three different design parameters of the adaptive beamformer were systematically varied. For those experiments, in which the direct-to-reverberant ratios of the noise signal is +3 dB or less, the difference between the predicted and the measured improvement in signal-to-noise ratio (SNR) is -0.21+/-0.59 dB for real environments and -0.25+/-0.51 dB for simulated environments (average +/- standard deviation). At higher direct-to-reverberant ratios, SNR improvement is systematically underestimated by up to 5.34 dB. The parameters with the greatest influence on the performance of the adaptive beamformer have been found to be the direct-to-reverberant ratio of the noise source, the reverberation time of the acoustic environment, and the length of the adaptive filter.     

Similarity in loudness and distortion product otoacoustic emission input/output functions: implications for an objective hearing aid adjustment

The aim of the present study was to compare distortion product otoacoustic emissions (DPOAEs) to loudness with regard to the potentiality of DPOAEs to determine characteristic quantities of the cochlear-impaired ear and to derive objective hearing aid parameters. Recently, Neely et al. [J. Acoust. Soc. Am. 114, 1499-1507 (2003)] compared DPOAE input/output functions to the Fletcher and Munson [J. Acoust. Soc. Am. 5, 82-108 (1933)] loudness function finding a close resemblance in the slope characteristics of both measures. The present study extended their work by performing both loudness and DPOAE measurements in the same subject sample, and by developing a method for the estimation of gain needed to compensate for loss of cochlear sensitivity and compression. DPOAEs and loudness exhibited similar behavior when plotted on a logarithmic scale and slope increased with increasing hearing loss, confirming the findings of Neely et al. To compensate for undesired nonpathological impacts on the magnitude of DPOAE level, normalization of DPOAE data was implemented. A close resemblance between gain functions based on loudness and normalized DPOAE data was achieved. These findings suggest that DPOAEs are able to quantify the loss of cochlear sensitivity and compression and thus might provide parameters for a noncooperative hearing aid adjustment.     

Perceptual adaptation by normally hearing listeners to a simulated "hole" in hearing

Simulations of cochlear implants have demonstrated that the deleterious effects of a frequency misalignment between analysis bands and characteristic frequencies at basally shifted simulated electrode locations are significantly reduced with training. However, a distortion of frequency-to-place mapping may also arise due to a region of dysfunctional neurons that creates a "hole" in the tonotopic representation. This study simulated a 10 mm hole in the mid-frequency region. Noise-band processors were created with six output bands (three apical and three basal to the hole). The spectral information that would have been represented in the hole was either dropped or reassigned to bands on either side. Such reassignment preserves information but warps the place code, which may in itself impair performance. Normally hearing subjects received three hours of training in two reassignment conditions. Speech recognition improved considerably with training. Scores were much lower in a baseline (untrained) condition where information from the hole region was dropped. A second group of subjects trained in this dropped condition did show some improvement; however, scores after training were significantly lower than in the reassignment conditions. These results are consistent with the view that speech processors should present the most informative frequency range irrespective of frequency misalignment.     

Performance of an adaptive beamforming noise reduction scheme for hearing aid applications. I. Prediction of the signal-to-noise-ratio improvement

Adaptive beamformers have been proposed as noise reduction schemes for conventional hearing aids and cochlear implants. A method to predict the amount of noise reduction that can be achieved by a two-microphone adaptive beamformer is presented. The prediction is based on a model of the acoustic environment in which the presence of one acoustic target-signal source and one acoustic noise source in a reverberant enclosure is assumed. The acoustic field is sampled using two omnidirectional microphones mounted close to the ears of a user. The model takes eleven different parameters into account, including reverberation time and size of the room, directionality of the acoustic sources, and design parameters of the beamformer itself, including length of the adaptive filter and delay in the target signal path. An approximation to predict the achievable signal-to-noise improvement based on the model is presented. Potential applications as well as limitations of the proposed prediction method are discussed and a FORTRAN subroutine to predict the achievable signal-to-noise improvement is provided. Experimental verification of the predictions is provided in a companion paper [J. Acoust. Soc. Am. 109, 1134 (2001)].     

听觉障碍患者的空间听觉*

空间听觉是对声音空间属性或特性的主观感觉,包括对声源的定位、对环境反射声的主观感觉等。复杂声学环境下的语言获取也和空间听觉密切相关。听觉障碍通常会包括空间听觉能力的下降甚至缺失,影响语言的获取能力。人工听觉是治疗听觉障碍的手段,理想情况下应能恢复或改善患者的空间听觉能力。该文综述了听觉障碍患者的空间听觉及其人工恢复方面的研究、进展及存在的问题。