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.     

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.     

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.     

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.     

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.     

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).     

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.     

Horizontal localization with bilateral hearing aids: without is better than with

This paper studies the effect of bilateral hearing aids on directional hearing in the frontal horizontal plane. Localization tests evaluated bilateral hearing aid users using different stimuli and different noise scenarios. Normal hearing subjects were used as a reference. The main research questions raised in this paper are: (i) How do bilateral hearing aid users perform on a localization task, relative to normal hearing subjects? (ii) Do bilateral hearing aids preserve localization cues, and (iii) Is there an influence of state of the art noise reduction algorithms, more in particular an adaptive directional microphone configuration, on localization performance? The hearing aid users were tested without and with their hearing aids, using both a standard omnidirectional microphone configuration and an adaptive directional microphone configuration. The following main conclusions are drawn. (i) Bilateral hearing aid users perform worse than normal hearing subjects in a localization task, although more than one-half of the subjects reach normal hearing performance when tested unaided. For both groups, localization performance drops significantly when acoustical scenarios become more complex. (ii) Bilateral, i.e., independently operating hearing aids do not preserve localization cues. (iii) Overall, adaptive directional noise reduction can have an additional and significant negative impact on localization performance.     

Methods of measuring the attenuation of hearing protection devices

The published literature describing three real-ear-attenuation-at-threshold (REAT), nine above-threshold, and four objective methods of measuring hearing protector attenuation is reviewed and analyzed with regard to the accuracy, practicality, and applicability of the various techniques. The analysis indicates that the REAT method is one of the most accurate available techniques since it assesses all of the sound paths to the occluded ear and, depending upon the experimenter's intention, can reflect actual in-use attenuation as well. An artifact in the REAT paradigm is that masking in the occluded ear due to physiological noise can spuriously increase low-frequency (less than or equal to 500 Hz) attenuation, although the error never exceeds approximately 5 dB, regardless of the device, except below 125 Hz. Since the preponderance of available data indicates that attenuation is independent of sound level for intentionally linear protectors, the use of above-threshold procedures to evaluate attenuation is not a necessity. An exception exists in the case of impulsive noises, for which the existing data are not unequivocal with regard to hearing protector response characteristics. Two of the objective methods (acoustical test fixture and microphone in real ear) are considerable time savers. All objective procedures are lacking in their ability to accurately determine the importance of the flanking bone-conduction paths, although some authors have incorporated this feature as a post-measurement correction. The microphone in real-ear approach is suggested to be one of the most promising for future standardization efforts and research purposes, and the acoustical test fixture technique is recommended (with certain reservations) for quality control and buyer acceptance testing.     

应用于声反馈抑制的变步长归一化子带自适应滤波算法*

针对自适应滤波算法中稳态失调量和收敛速度之间的矛盾,提出了一种新的变步长归一化子带自适应滤波算法。该算法在系统噪声抵消原理的基础上,用迭代收缩的方法估计得到无噪先验子带误差的功率,对每个子带步长进行更新。对所提出的算法进行数学分析,可以得出该算法是稳定的和收敛的。在长回声路径和短回声路径两种情况下,将该算法应用于助听器声反馈抑制系统中。相对于其他归一化子带自适应滤波算法,仿真实验表明,所提算法实现了更快的收敛速度,获得了更低的失调量。     

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.     

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.     

啸叫快速抑制的助听器回声抵消算法

针对助听器回声路径快速变化下易产生啸叫的问题,本文提出一种变步长标准最小均方差-陷波器(Variable Step Normalized least mean square-Notch Filter,VSN-NF)算法。在回声路径相对稳定时,提出一种基于状态分类的变步长标准最小均方差算法来估计回声信号。算法根据滤波器系数能量的长时平均值和短时平均值,将滤波器当前状态分为收敛态、过渡态与稳态,并根据不同状态选择不同的步长。在路径突然变化并产生啸叫时,算法通过关闭变步长NLMS算法来稳定啸叫频点,然后基于ZoomFFT算法动态生成陷波器来进行啸叫抑制;当啸叫抑制后,再开启变步长NLMS进行回声估计。针对易产生多频点啸叫的回声路径,VSN-NF算法还引入不同频带的两个陷波器来进行双频点啸叫抑制。同其它助听器回声抵消算法的对比实验显示,VSN-NF算法的回波抵消性能最好,尤其具有快速啸叫抑制能力。此外,算法生成的语音质量较高,实时性能好,适合于像助听器类的低功耗、小体积产品。      

Bearing splitting and near-surface source ranging in the direct zone of deep water

Sound multipath propagation is very important for target localization and identification in different acoustical zones of deep water. In order to distinguish the multipath characteristics in deep water, the Northwest Pacific Acoustic Experiment was conducted in 2015. A low-frequency horizontal line array towed at the depth of around 150 m on a receiving ship was used to receive the noise radiated by the source ship. During this experiment, a bearing-splitting phenomenon in the direct zone was observed through conventional beamforming of the horizontal line array within the frequency band 160 Hz–360 Hz. In this paper, this phenomenon is explained based on ray theory. In principle, the received signal in the direct zone of deep water arrives from two general paths including a direct one and bottom bounced one, which vary considerably in arrival angles. The split bearings correspond to the contributions of these two paths. The bearing-splitting phenomenon is demonstrated by numerical simulations of the bearing-time records and experimental results, and they are well consistent with each other. Then a near-surface source ranging approach based on the arrival angles of direct path and bottom bounced path in the direct zone is presented as an application of bearing splitting and is verified by experimental results. Finally,the applicability of the proposed ranging approach for an underwater source within several hundred meters in depth in the direct zone is also analyzed and demonstrated by simulations.     

The use of acoustical test fixtures for the measurement of hearing protector attenuation. Part II: Modeling the external ear, simulating bone conduction, and comparing test fixture and real-ear data

This paper investigates two main features of the human head which influence the measured attenuation of circumaural and intraaural hearing protection devices (HPDs): the external ear and the different pathways of bone conduction. A theoretical model for the external ear shows that its influence on the insertion loss of HPDs, on the sensitivity level of headphones or earphones, and on the insertion gain of hearing aids, all can be described by one equation. While it is not necessary to simulate the eardrum impedance in order to measure the insertion loss of earmuffs and the sensitivity level of headphones with acoustical test fixtures (ATFs), the required accuracy of an ear simulator is more stringent when the same measurements are performed on intraaural devices. For the evaluation of HPDs, bone conduction plays an important role. We have developed a model to estimate HPD-dependent bone conduction effects. The model includes two bone conduction sources: one in the external ear and one in the middle ear. The model explains, for example, the occlusion effect of HPDs and the masking error at low frequencies due to physiological noise that arises when real-ear attenuation at threshold (REAT) measurements are made. Consequently, objectively measured insertion loss can now be used to predict REAT with improved accuracy. ATF and REAT data are compared using nine earmuffs and nine earplugs. In the majority of cases, the two sets of data agree well. Discrepancies are discussed.     

Modelling and characterization of airborne noise sources

A comprehensive model of an arbitrary airborne noise source is developed in this paper. It can be used for noise prediction in complex environment conditions. The model, described by a limited number of parameters—the source impedance and blocked pressure—is invariant with respect to the surrounding acoustical environment. Procedures for experimental identification of model parameters in an arbitrary enclosure are presented. These procedures require only conventional equipment and facilities, and thus can be applied widely. Two analytical examples confirm the validity of the approach.     

Active cancellation of occlusion: an electronic vent for hearing aids and hearing protectors

The occlusion effect is commonly described as an unnatural and mostly annoying quality of the voice of a person wearing hearing aids or hearing protectors. As a result, it is often reported by hearing aid users as a deterrent to wearing hearing aids. This paper presents an investigation into active occlusion cancellation. Measured transducer responses combined with models of an active feedback scheme are first examined in order to predict the effectiveness of occlusion reduction. The simulations predict 18 dB of occlusion reduction in completely blocked ear canals. Simulations incorporating a 1 mm vent (providing passive occlusion reduction) predict a combined active and passive occlusion reduction of 20 dB. A prototype occlusion canceling system was constructed. Averaged across 12 listeners with normal hearing, it provided 15 dB of occlusion reduction. Ten of the subjects reported a more natural own voice quality and an appreciable increase in comfort with the cancellation active, and 11 out of the 12 preferred the active system over the passive system.     

Boundary element modeling of the external human auditory system

In this paper the response of the external auditory system to acoustical waves of varying frequencies and angles of incidence is computed using a boundary element method. The resonance patterns of both the ear canal and the concha are computed and compared with experimental data. Specialized numerical algorithms are developed that allow for the efficient computation of the eardrum pressures. In contrast to previous results in the literature that consider only the "blocked meatus" configuration, in this work the simulations are conducted on a boundary element mesh that includes both the external head/ear geometry, as well as the ear canal and eardrum. The simulation technology developed in this work is intended to demonstrate the utility of numerical analysis in studying physical phenomena related to the external auditory system. Later work could extend this towards simulating in situ hearing aids, and possibly using the simulations as a tool for optimizing hearing aid technologies for particular individuals.     

A hybrid active noise controller for finite ducts

A hybrid active noise controller (ANC) is proposed to solve some existing problems, which are related to the non-minimum phase (NMP) path models between uncollocated sensors and actuators in many ANC systems. For hybrid ANC schemes, the NMP path causes design difficulties to both feedforward and feedback control. These problems can be solved effectively by adding an extra actuator in the ANC system. A new design procedure is presented to take the greatest advantage of the extra actuator. Theoretical analysis and experimental results are presented to show the improved performance of the proposed ANC.