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.     

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.     

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.     

应用于助听器反馈抑制的信噪比自适应滤波算法

针对基于自适应滤波器的助听器反馈抑制系统,该文提出了一种基于信噪比的归一化最小均方误差算法,采用最小值统计法估计误差信号的噪声分量,从而计算出误差信号的信噪比来计算自适应滤波系数的更新步长。当误差信号信噪比越高,语声占主要成分,信号的相关性越强,此时将滤波器的更新步长控制在较小值,减小滤波器的失调量;当信噪比越低时,噪声占主要成分,信号的相关性相对较弱,更新步长取较大值,加快滤波器的收敛速度。在仿真实验中,该文提出的基于信噪比的归一化最小均方误差算法相较于传统算法在平均稳态失调量和稳态失调范围上分别低1 dB和2 dB,其最大稳态增益提高了4 dB,同时具有更快的稳态收敛速度,验证了该文提出算法的有效性。     

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.     

Simulating the open-loop transfer function as a means for understanding acoustic feedback in hearing aids

Suppressing unstable acoustic feedback in hearing aids will first require knowledge of the open-loop transfer functions of such systems. Reported herein is a mathematical technique for simulating the open-loop transfer function of an in situ eyeglass-type hearing aid. In particular, a computer program was developed that characterized the hearing aid as a serial connection of two-port blocks, each representing one individual component of a hearing aid. Included, for example, were two-port blocks representing the microphone, amplifier, receiver, sound tubes leading to the eardrum (including the ear canal itself), earmold vent, and external pathway from the vent outlet back to the microphone. The computer program was validated by replicating laboratory data derived from an experiment involving a nonstandard manikin fitted with a nonstandard artificial ear. Next, the open-loop transfer function of an eyeglass-type hearing aid in situ on the manikin was simulated via the computer program. Unfortunately, those computer-generated data were not replicated in the laboratory due to the difficulty encountered in actually measuring the open-loop transfer function. Nevertheless, investigators were able to utilize those data to predict, within +/- 25 Hz, the "squeal" frequency of unstable acoustic feedback.     

汉语数字助听器高频听损增强方法的实验研究

提出采用正弦模型改善患者高频听觉的非线性降频方法。正弦模型语音分解得到的幅度、频率和相位是算法三个主要的处理参数。为了避免谱失真,将语音频谱按倍频程划分为6个部分。最接近并低于患者门限频率的部分,只做幅度放大处理。按照不同频段对于语音理解度的贡献程度,将患者门限频率以上的频率段压缩并转移到患者的可听频段,并将对应相位信息变为最接近的对应低频相位。在本研究中,10个受试者进行了语音理解度测试。测试结果显示,经过训练后,患者的平均理解率至少提高45%。下一步的研究应增加受试者数量,并增加对患者的听损情况的详细分析,从而设计出更合理,更细致的降频助听算法。     

Dynamic and static position control of optical feedback solitons

We report on the experimental implementation of an external control for optical feedback solitons using incoherent spatial intensity distributions in a liquid crystal light valve (LCLV) optical single feedback system. The external control provides excellent experimental possibilities for static and dynamic control of the lateral positions of the optical feedback solitons which will be demonstrated. Particularly, the influence of different gradients onto the drift motion of spatial solitons is experimentally investigated in detail. In agreement with theoretical predictions, the drift velocity of the soliton increases according to the steepness of the gradient. Additionally, a completely incoherent addressing scheme including creation and erasure of feedback solitons is demonstrated for the LCLV setup.     

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.     

Auditory handicap of hearing impairment and the limited benefit of hearing aids

The aim of this article is to promote a better understanding of hearing impairment as a communicative handicap, primarily in noisy environments, and to explain by means of a quantitative model the essentially limited applicability of hearing aids. After data on the prevalence of hearing impairment and of auditory handicap have been reviewed, it is explained that every hearing loss for speech can be interpreted as the sum of a loss class A (attenuation), characterized by a reduction of the levels of both speech signal and noise, and a loss D (distortion), comparable with a decrease in speech-to-noise ratio. On the average, the hearing loss of class D (hearing loss in noise) appears to be about one-third (in decibels) of the total hearing loss (A + D, hearing loss in quiet). A hearing aid can compensate for class-A-hearing losses, giving difficulties primarily in quiet, but not for class-D hearing losses, giving difficulties primarily in noise. The latter class represents the first stage of auditory handicap, beginning at an average hearing loss of about 24 dB.     

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.     

Evaluation of an adaptive beamforming method for hearing aids.

In this paper evaluations of a two-microphone adaptive beamforming system for hearing aids are presented. The system, based on the constrained adaptive beamformer described by Griffiths and Jim [IEEE Trans. Antennas Propag. AP-30, 27-34 (1982)], adapts to preserve target signals from straight ahead and to minimize jammer signals arriving from other directions. Modifications of the basic Griffiths-Jim algorithm are proposed to alleviate problems of target cancellation and misadjustment that arise in the presence of strong target signals. The evaluations employ both computer simulations and a real-time hardware implementation and are restricted to the case of a single jammer. Performance is measured by the spectrally weighted gain in the target-to-jammer ratio in the steady state. Results show that in environments with relatively little reverberation: (1) the modifications allow good performance even with misaligned arrays and high input target-to-jammer ratios; and (2) performance is better with a broadside array with 7-cm spacing between microphones than with a 26-cm broadside or a 7-cm endfire configuration. Performance degrades in reverberant environments; at the critical distance of a room, improvement with a practical system is limited to a few dB.     

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.     

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。     

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.     

混沌神经网络的延时反馈控制研究

针对混沌神经网络，提出了一种改进的延时反馈控制方法. 利用该方法，当延时参数τ为奇数时，被控神经网络收敛于记忆模式以及它的反相模式的2周期上. 若选取不同的延时参数，被控网络则收敛于不同的周期态上. 关键词： 控制混沌 延时反馈控制 混沌神经网络     

一种用于数字助听器的WOLA滤波器组的设计准则

针对WOLA(weighted overlap-add)滤波器组在数字助听器多通道处理中的应用,本文提出了一个满足该领域要求的具体的设计准则。根据这一准则,可用窗函数法进行滤波器设计,并把问题简化为对窗长度的选取。对线性扫频信号和语音信号多通道放大两种情况进行了仿真,结果表明,适当选取窗长度,滤波器组输出信号将符合要求并且无混迭。     

Mathematical predictions of electroacoustic frequency response of in situ hearing aids

The amplitude spectrum of an acoustic signal presented to the microphone of a hearing aid is altered drastically before it finally reaches the user's eardrum. A major part of this alteration is due to the interaction of various mechanical and acoustic resonances which are characteristic of the hearing-aid receiver and the sound transmission system linking the receiver with the eardrum. Because of the complexity of this phenomenon, there is yet no means for predicting, a priori, the true shape of the sound spectrum that will occur at the user's eardrum. This paper reports on the development and testing of just such a scheme. The accuracy of this scheme--a computer-aided mathematical technique--is measured in the laboratory on real and artificial ears. The results of those measurements show good agreement between experimental and computer-generated data below 5000 Hz.