Detecting high-frequency hearing loss with click-evoked otoacoustic emissions

In contrast to clinical click-evoked otoacoustic emission (CEOAE) tests that are inaccurate above 4-5 kHz, a research procedure measured CEOAEs up to 16 kHz in 446 ears and predicted the presence/absence of a sensorineural hearing loss. The behavioral threshold test that served as a reference to evaluate CEOAE test accuracy used a yes-no task in a maximum-likelihood adaptive procedure. This test was highly efficient between 0.5 and 12.7 kHz: Thresholds measured in 2 min per frequency had a median standard deviation (SD) of 1.2-1.5 dB across subjects. CEOAE test performance was assessed by the area under the receiver operating characteristic curve (AUC). The mean AUC from 1 to 10 kHz was 0.90 (SD=0.016). AUC decreased to 0.86 at 12.7 kHz and to 0.7 at 0.5 and 16 kHz, possibly due in part to insufficient stimulus levels. Between 1 and 12.7 kHz, the medians of the magnitude difference in CEOAEs and in behavioral thresholds were <4 dB. The improved CEOAE test performance above 4-5 kHz was due to retaining the portion of the CEOAE response with latencies as short as 0.3 ms. Results have potential clinical significance in predicting hearing status from at least 1 to 10 kHz using a single CEOAE 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.     

Speech-reception threshold in noise with one and two hearing aids

The binaural free-field speech-reception threshold (SRT) in 70-dBA noise was measured with conversational sentences for 24 hearing-impaired subjects without hearing aids, with a hearing aid left, right, and left plus right, respectively. The sentences were always presented in front of the listener and the interfering noise, with a spectrum equal to the long-term average spectrum of the sentences, was presented either frontally, from the right, or from the left side. For subjects with only moderate hearing loss, PTA (average air-conduction hearing level at 500, 1000, and 2000 Hz) less than 50 dB, the SRT in 70-dBA noise in both ears is determined by the signal-to-noise ratio even if only one hearing aid is used. For larger hearing losses the SRT appears to be partly determined by the absolute threshold. In conditions with a high noise level relative to the absolute threshold, in which case for both ears the SRT is determined by the signal-to-noise ratio, a second hearing aid, just as a monaural hearing aid, generally does not improve the SRT. However, in the case of a high hearing level, or a low noise level, in which a monaural hearing aid is profitable, the use of two hearing aids is even more profitable. In a separate experiment, acoustic head shadow was measured at the entrance of the ear canal and at the microphone location of a hearing aid. It appeared that, for a lateral noise source and speech frontal, the microphone position of behind-the-ear hearing aids has a negative effect on the signal-to-noise ratio of 2-3 dB.     

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.     

Effects of directional microphone and adaptive multichannel noise reduction algorithm on cochlear implant performance

Although cochlear implant (CI) users have enjoyed good speech recognition in quiet, they still have difficulties understanding speech in noise. We conducted three experiments to determine whether a directional microphone and an adaptive multichannel noise reduction algorithm could enhance CI performance in noise and whether Speech Transmission Index (STI) can be used to predict CI performance in various acoustic and signal processing conditions. In Experiment I, CI users listened to speech in noise processed by 4 hearing aid settings: omni-directional microphone, omni-directional microphone plus noise reduction, directional microphone, and directional microphone plus noise reduction. The directional microphone significantly improved speech recognition in noise. Both directional microphone and noise reduction algorithm improved overall preference. In Experiment II, normal hearing individuals listened to the recorded speech produced by 4- or 8-channel CI simulations. The 8-channel simulation yielded similar speech recognition results as in Experiment I, whereas the 4-channel simulation produced no significant difference among the 4 settings. In Experiment III, we examined the relationship between STIs and speech recognition. The results suggested that STI could predict actual and simulated CI speech intelligibility with acoustic degradation and the directional microphone, but not the noise reduction algorithm. Implications for intelligibility enhancement are discussed.     

Effects of flanking noise bands on the rate of growth of loudness of tones in normal and recruiting ears

Five subjects with unilateral cochlear hearing impairments and three normally hearing subjects made loudness matches between tones presented alternately to two ears, as a function of the intensity of the tone in the impaired ear (or the left ear of the normal subjects). The impaired ears showed recruitment; the rate of growth of loudness with increasing intensity was more rapid in the impaired ear than the normal ear. Presenting the tone in the impaired ear with two noise bands on either side of the tone frequency, at a fixed signal-to-noise ratio, did not abolish the recruitment. This suggests that recruitment is not caused by an abnormally rapid spread of excitation in the peripheral auditory system. At low signal-to-noise ratios, a continuous background noise reduced the loudness of the tone more than a noise gated with the tone, suggesting that the continuous noise induces adaptation to the tone. The noise had a greater effect on the loudness of the tone in normal ears than in impaired ears. It is possible that the loudness reduction of the tone in noise is mediated by suppression; suppression is weak or absent in impaired ears, and so the loudness reduction is smaller.     

Experimental evaluation of leaky least-mean-square algorithms for active noise reduction in communication headsets

An adaptive leaky normalized least-mean-square (NLMS) algorithm has been developed to optimize stability and performance of active noise cancellation systems. The research addresses LMS filter performance issues related to insufficient excitation, nonstationary noise fields, and time-varying signal-to-noise ratio. The adaptive leaky NLMS algorithm is based on a Lyapunov tuning approach in which three candidate algorithms, each of which is a function of the instantaneous measured reference input, measurement noise variance, and filter length, are shown to provide varying degrees of tradeoff between stability and noise reduction performance. Each algorithm is evaluated experimentally for reduction of low frequency noise in communication headsets, and stability and noise reduction performance are compared with that of traditional NLMS and fixed-leakage NLMS algorithms. Acoustic measurements are made in a specially designed acoustic test cell which is based on the original work of Ryan et al. ["Enclosure for low frequency assessment of active noise reducing circumaural headsets and hearing protection," Can. Acoust. 21, 19-20 (1993)] and which provides a highly controlled and uniform acoustic environment. The stability and performance of the active noise reduction system, including a prototype communication headset, are investigated for a variety of noise sources ranging from stationary tonal noise to highly nonstationary measured F-16 aircraft noise over a 20 dB dynamic range. Results demonstrate significant improvements in stability of Lyapunov-tuned LMS algorithms over traditional leaky or nonleaky normalized algorithms, while providing noise reduction performance equivalent to that of the NLMS algorithm for idealized noise fields.     

Temporal suppression of the click-evoked otoacoustic emission level-curve

The click-evoked otoacoustic emission (CEOAE) level-curve grows linearly for clicks below 40-60 dB and saturates for higher inputs. This study investigates dynamic (i.e., time-dependent) features of the CEOAE level-curve by presenting a suppressor-click less than 8 ms before the test-click. An alteration of the CEOAE level-curve, designated here as temporal suppression, was observed within this time period, and was shown to depend on the levels and the temporal separation of the two clicks. Temporal suppression occurred for all four subjects tested, and resulted in a vertical offset from the unsuppressed level-curve for test-click levels greater than 50 dB peak-equivalent level (peSPL). Temporal suppression was greatest for suppressors presented 1-4 ms before the test click, and the magnitude and time scale of the effect were subject dependent. Temporal suppression was furthermore observed for the short- (i.e., 6-18 ms) and long-latency (i.e., 24-36 ms) regions of the CEOAE, indicating that temporal suppression similarly affects synchronized spontaneous otoacoustic emissions (SSOAEs) and purely evoked CEOAE components. Overall, this study demonstrates that temporal suppression of the CEOAE level-curve reflects a dynamic process in human cochlear processing that works on a time scale of 0-10 ms.     

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.     

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)].     

Overshoot in normal-hearing and hearing-impaired subjects.

Overshoot was measured in both ears of four subjects with normal hearing and in five subjects with permanent, sensorineural hearing loss (two with a unilateral loss). The masker was a 400-ms broadband noise presented at a spectrum level of 20, 30, or 40 dB SPL. The signal was a 10-ms sinusoid presented 1 or 195 ms after the onset of the masker. Signal frequency was 1.0 or 4.0 kHz, which placed the signal in a region of normal (1.0 kHz) or impaired (4.0 kHz) absolute sensitivity for the impaired ears. For the normal-hearing subjects, the effects of signal frequency and masker level were similar to those published previously. In particular, overshoot was larger at 4.0 than at 1.0 kHz, and overshoot at 4.0 kHz tended to decrease with increasing masker level. At 4.0 kHz, overshoot values were significantly larger in the normal ears: Maximum values ranged from about 7-26 dB in the normal ears, but were always less than 5 dB in the impaired ears. The smaller overshoot values resulted from the fact that thresholds in the short-delay condition were considerably better in the hearing-impaired subjects than in the normal-hearing subjects. At 1.0 kHz, overshoot values for the two groups of subjects more or less overlapped. The results suggest that permanent, sensorineural hearing loss disrupts the mechanisms responsible for a large overshoot effect.     

The effect of multimicrophone noise reduction systems on sound source localization by users of binaural hearing aids

This paper evaluates the influence of three multimicrophone noise reduction algorithms on the ability to localize sound sources. Two recently developed noise reduction techniques for binaural hearing aids were evaluated, namely, the binaural multichannel Wiener filter (MWF) and the binaural multichannel Wiener filter with partial noise estimate (MWF-N), together with a dual-monaural adaptive directional microphone (ADM), which is a widely used noise reduction approach in commercial hearing aids. The influence of the different algorithms on perceived sound source localization and their noise reduction performance was evaluated. It is shown that noise reduction algorithms can have a large influence on localization and that (a) the ADM only preserves localization in the forward direction over azimuths where limited or no noise reduction is obtained; (b) the MWF preserves localization of the target speech component but may distort localization of the noise component. The latter is dependent on signal-to-noise ratio and masking effects; (c) the MWF-N enables correct localization of both the speech and the noise components; (d) the statistical Wiener filter approach introduces a better combination of sound source localization and noise reduction performance than the ADM approach.     

一种基于虚拟传感的无需误差传声器的自适应有源降噪方法*

该文基于虚拟传感技术引入了一种用于耳机的无需误差传声器的自适应有源降噪方法。该算法仅使用一个参考传声器实现了一种前馈和反馈自适应算法结合的有源降噪算法,提高了有源降噪稳定性,简化了耳机硬件结构。利用DSP平台实现了该文提出的方案,并通过实验验证了其良好的降噪性能和实用价值。     

Hybrid active and passive control of fan noise

This paper deals with the global reduction of axial flow fan noise in ducts in a building using a hybrid passive-active noise control method. The effectiveness of using an infra-red device as a reference signal source is also investigated. It is shown that using such a hybrid noise control system over an axial-flow fan reduces the overall sound pressure level by 5 dB(A) in the surrounding environment and global control of the blade passing frequency can also be achieved. This paper also shows that using an infra-red device as a reference signal source produces marginally better control as compared with using a microphone reference sensor. Moreover, long term stability is guaranteed and the possibility of acoustic feedback is eliminated.     

Effects of nosocusis, and industrial and gun noise on hearing of U.S. adults.

Hearing losses estimated for exposure to industrial and gun noise and for "typical" nosocusis are applied to the distributions of the hearing levels of adult males and females of the general population of an industrialized society unscreened for exposure to noise or ear disease. Noise exposure and demographic data applicable to the United States, and procedures for predicting noise-induced permanent threshold shift (NIPTS) and nosocusis, were used to account for some 8.7 dB of the 13.4 dB average difference between the hearing levels at high frequencies for otologically and noise screened versus unscreened male ears; (this average difference is for the average of the hearing levels at 3000, 4000, and 6000 Hz, average for the 10th, 50th, and 90th percentiles, and ages 20-65 years). According to the present calculations, this difference is due, in order of importance, to (1) nosocusis, (2) exposure to gun noise, and (3) exposure of workers to industrial noise. For these same frequencies and overall average, adjustments for nosocusis accounts for 2 dB of the 5.9-dB difference between the hearing levels of screened and unscreened female ears. For the average at 500, 1000, and 2000 Hz, the overall differences between the screened and unscreened populations is but 3.4 dB for males and 2.9 dB for females. The adjustment procedures reduced these differences to -0.5 and 0.9 dB, respectively.     

Simplified hearing protector ratings—an international comparison

A computer was programmed to model the distributions of dB(A) levels reaching the ears of an imaginary workforce wearing hearing protectors selected on the basis of either octave band attenuation values or various simplified ratings in use in Australia, Germany, Poland, Spain or the U.S.A. Both multi-valued and single-valued versions of dB(A) reduction and sound level conversion ratings were considered. Ratings were compared in terms of precision and protection rate and the comparisons were replicated for different samples of noise spectra (N = 400) and hearing protectors (N = 70) to establish the generality of the conclusions. Different countries adopt different approaches to the measurement of octave band attenuation values and the consequences of these differences were investigated. All rating systems have built-in correction factors to account for hearing protector performance variability and the merits of these were determined in the light of their ultimate effects on the distribution of dB(A) levels reaching wearers' ears. It was concluded that the optimum rating is one that enables the dB(A) level reaching wearers to be estimated by subtracting a single rating value from the dB(C) level of the noise environment, the rating value to be determined for a pink noise spectrum from mean minus one standard deviation octave band attenuation values with further protection rate adjustments being achieved by the use of a constant correction factor.     

Detection of temporal gaps in sinusoids by normally hearing and hearing-impaired subjects

A two-alternative forced-choice task was used to measure psychometric functions for the detection of temporal gaps in a 1-kHz, 400-ms sinusoidal signal. The signal always started and finished at a positive-going zero crossing, and the gap duration was varied from 0.5 to 6.0 ms in 0.5-ms steps. The signal level was 80 dB SPL, and a spectrally shaped noise was used to mask splatter associated with the abrupt onset and offset of the signal. Two subjects with normal hearing, two subjects with unilateral cochlear hearing loss, and two subjects with bilateral cochlear hearing loss were tested. The impaired ears had confirmed reductions in frequency selectivity at 1 kHz. For the normal ears, the psychometric functions were nonmonotonic, showing minima for gap durations corresponding to integer multiples of the signal period (n ms, where n is a positive integer) and maxima for durations corresponding to (n - 0.5) ms. For the impaired ears, the psychometric functions showed only small (nonsignificant) nonmonotonicities. Performance overall was slightly worse for the impaired than for the normal ears. The main features of the results could be accounted for using a model consisting of a bandpass filter (the auditory filter), a square-law device, and a sliding temporal integrator. Consistent with the data, the model demonstrates that, although a broader auditory filter has a faster transient response, this does not necessarily lead to improved performance in a gap detection task. The model also indicates that gap thresholds do not provide a direct measure of temporal resolution, since they depend at least partly on intensity resolution.     

Use of the sound levels of noise for assessing the adequacy of hearing protectors

The dB(A) sound level of a noise is accepted as a measure of the damage risk to unprotected ears but often it is not a reliable guide to the risk to ears fitted with hearing protectors. For any dB(A) level inside a protector, normally there will be substantially higher sound levels outside that protector. This paper shows how, from sequential frequency attenuation bands of the protector, and sound level weightings, external sound levels can be calculated, below which the noise inside the protector does not exceed a chosen dB(A) level. Further valuable information may be obtained by mapping external dB(A) and dB(C) levels to cover all possible noise spectra that give the chosen dB(A) level inside the protector. Thus, from a pair of measured sound levels, use of the method indicates whether the protector is sufficient or not, or whether more detailed measurment of the noise is required. This knowledge enhances the scope of the sound level meter and reduces the need for frequency analysis of industrial noise. Its application should be a helpful addition to the data provided by suppliers of hearing protectors.     

Noise levels and hearing thresholds in the drop forging industry

A-weighted equivalent continuous noise levels for hammer and press operations in a drop-forging industry were determined using both tape recordings of the noise and personal noise dosimeters. The results indicated average A-weighted Leq values of 108 dB for hammer operators and 99 dB for press operators. Comparison of hearing level statistics for 716 hammer and press operators and 293 control subjects indicated the severe hazard to hearing of impact noise exposures. For mean exposure times of less than 10 years, hearing levels for the press (99 dB) and hammer (108 dB) operator age groups are nearly identical, and in the latter case are less than those predicted for exposure to equivalent continuous noise. For long-term exposures of 10 years or more, the results of this study indicate that hearing losses resulting from impact noise in the drop-forging industry are as great or greater than those resulting from continuous noise.     

基于脉冲声分离的刚性球散射声控制

提出了一种基于脉冲声的三维空间中刚性球散射声分离方法,并利用前馈、固定系数控制方式对分离出的散射声进行有源控制,抑制散射声强度,实现了刚性球散射体在观测点处“声学不可见”。该方法利用脉冲信号作为初级噪声,通过有无刚性球时传声器采集脉冲信号的差值确定散射声大小,实现散射声与声源直达声的分离。对分离出的散射声进行多通道有源控制以验证该文所提分离方法及控制系统的有效性。实验结果表明,700~1000 Hz范围内,有源控制开启后,双通道散射声的平均降噪量大于5 dB,多通道散射声的平均降噪量大于8 dB,且误差传声器处采集的残余声场与无刚性球时采集的初级声场信号波形基本一致,实现了刚性球散射体在误差传声器处“声学不可见”。此外,参考传声器布放位置的选取问题也在该文做了详细讨论。