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.     

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.     

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.     

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.     

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.     

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.     

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。     

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.     

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。     

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.