Hybrid feedforward-feedback active noise reduction for hearing protection and communication

A hybrid active noise reduction (ANR) architecture is presented and validated for a circumaural earcup and a communication earplug. The hybrid system combines source-independent feedback ANR with a Lyapunov-tuned leaky LMS filter (LyLMS) improving gain stability margins over feedforward ANR alone. In flat plate testing, the earcup demonstrates an overall C-weighted total noise reduction of 40 dB and 30-32 dB, respectively, for 50-800 Hz sum-of-tones noise and for aircraft or helicopter cockpit noise, improving low frequency (<100 Hz) performance by up to 15 dB over either control component acting individually. For the earplug, a filtered-X implementation of the LyLMS accommodates its nonconstant cancellation path gain. A fast time-domain identification method provides a high-fidelity, computationally efficient, infinite impulse response cancellation path model, which is used for both the filtered-X implementation and communication feedthrough. Insertion loss measurements made with a manikin show overall C-weighted total noise reduction provided by the ANR earplug of 46-48 dB for sum-of-tones 80-2000 Hz and 40-41 dB from 63 to 3000 Hz for UH-60 helicopter noise, with negligible degradation in attenuation during speech communication. For both hearing protectors, a stability metric improves by a factor of 2 to several orders of magnitude through hybrid ANR.     

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.     

Speech intelligibility improvements with hearing aids using bilateral and binaural adaptive multichannel Wiener filtering based noise reduction

This paper evaluates noise reduction techniques in bilateral and binaural hearing aids. Adaptive implementations (on a real-time test platform) of the bilateral and binaural speech distortion weighted multichannel Wiener filter (SDW-MWF) and a competing bilateral fixed beamformer are evaluated. As the SDW-MWF relies on a voice activity detector (VAD), a realistic binaural VAD is also included. The test subjects (both normal hearing subjects and hearing aid users) are tested by an adaptive speech reception threshold (SRT) test in different spatial scenarios, including a realistic cafeteria scenario with nonstationary noise. The main conclusions are: (a) The binaural SDW-MWF can further improve the SRT (up to 2 dB) over the improvements achieved by bilateral algorithms, although a significant difference is only achievable if the binaural SDW-MWF uses a perfect VAD. However, in the cafeteria scenario only the binaural SDW-MWF achieves a significant SRT improvement (2.6 dB with perfect VAD, 2.2 dB with real VAD), for the group of hearing aid users. (b) There is no significant degradation when using a real VAD at the input signal-to-noise ratio (SNR) levels where the hearing aid users reach their SRT. (c) The bilateral SDW-MWF achieves no SRT improvements compared to the bilateral fixed beamformer.     

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.     

Effective compression and noise reduction configurations for hearing protectors

The author proposed to adopt wide dynamic range compression and adaptive multichannel modulation-based noise reduction algorithms to enhance hearing protector performance. Three experiments were conducted to investigate the effects of compression and noise reduction configurations on the amount of noise reduction, speech intelligibility, and overall preferences using existing digital hearing aids. In Experiment 1, sentence materials were recorded in speech spectrum noise and white noise after being processed by eight digital hearing aids. When the hearing aids were set to 3:1 compression, the amount of noise reduction achieved was enhanced or maintained for hearing aids with parallel configurations, but reduced for hearing aids with serial configurations. In Experiments 2 and 3, 16 normal-hearing listeners' speech intelligibility and perceived sound quality were tested when they listened to speech recorded through hearing aids with parallel and serial configurations. Regardless of the configuration, the noise reduction algorithms reduced the noise level and maintained speech intelligibility in white noise. Additionally, the listeners preferred the parallel rather than the serial configuration in 3:1 conditions and the serial configuration in 1:1 rather than 3:1 compression when the noise reduction algorithms were activated. Implications for hearing protector and hearing aid design are discussed.     

Speech production in noise with and without hearing protection

People working in noisy environments often complain of difficulty communicating when they wear hearing protection. It was hypothesized that part of the workers' communication difficulties stem from changes in speech production that occur when hearing protectors are worn. To address this possibility, overall and one-third-octave-band SPL measurements were obtained for 16 men and 16 women as they produced connected speech while wearing foam, flange, or no earplugs (open ears) in quiet and in pink noise at 60, 70, 80, 90, and 100 dB SPL. The attenuation and the occlusion effect produced by the earplugs were measured. The Speech Intelligibility Index (SII) was also calculated for each condition. The talkers produced lower overall speech levels, speech-to-noise ratios, and SII values, and less high-frequency speech energy, when they wore earplugs compared with the open-ear condition. Small differences in the speech measures between the talkers wearing foam and flange earplugs were observed. Overall, the results of the study indicate that talkers wearing earplugs (and consequently their listeners) are at a disadvantage when communicating in noise.     

Effects of fluctuating noise and interfering speech on the speech-reception threshold for impaired and normal hearing

The speech-reception threshold (SRT) for sentences presented in a fluctuating interfering background sound of 80 dBA SPL is measured for 20 normal-hearing listeners and 20 listeners with sensorineural hearing impairment. The interfering sounds range from steady-state noise, via modulated noise, to a single competing voice. Two voices are used, one male and one female, and the spectrum of the masker is shaped according to these voices. For both voices, the SRT is measured as well in noise spectrally shaped according to the target voice as shaped according to the other voice. The results show that, for normal-hearing listeners, the SRT for sentences in modulated noise is 4-6 dB lower than for steady-state noise; for sentences masked by a competing voice, this difference is 6-8 dB. For listeners with moderate sensorineural hearing loss, elevated thresholds are obtained without an appreciable effect of masker fluctuations. The implications of these results for estimating a hearing handicap in everyday conditions are discussed. By using the articulation index (AI), it is shown that hearing-impaired individuals perform poorer than suggested by the loss of audibility for some parts of the speech signal. Finally, three mechanisms are discussed that contribute to the absence of unmasking by masker fluctuations in hearing-impaired listeners. The low sensation level at which the impaired listeners receive the masker seems a major determinant. The second and third factors are: reduced temporal resolution and a reduction in comodulation masking release, respectively.     

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.     

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.     

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.     

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.     

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.     

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.     

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.     

仿人耳听觉的助听器双耳声源定位算法

为提高复杂场景下的听障患者的语言理解度,本文提出一种仿人耳听觉的助听器双耳声源定位算法。算法首先借鉴耳蜗分频特性和听觉掩蔽特性,将声音信号进行多通道分解,并提取人耳敏感频带的信号进行双耳时间差(Interaural Time Difference,ITD)估计;然后基于人耳哈斯效应,提取有效的ITD信息;最后采用头相关模型,将ITD转化为声源方向信息。同时,为了改善混响和多干扰声场景下的声源定位能力,本文提出一种多通道的加权联合策略。仿真和场景测试实验表明,算法的抗干扰性强,定位精度高。而且,在7名受试者的理解度测试中,同现有的助听器增强算法相比,结合定位算法的语音增强算法达到3~5dB的性能改善。      

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.     

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.     

Comparison of different forms of compression using wearable digital hearing aids

Four different compression algorithms were implemented in wearable digital hearing aids: (1) The slow-acting dual-front-end automatic gain control (AGC) system [B. C. J. Moore, B. R. Glasberg, and M. A. Stone, Br. J. Audiol. 25, 171-182 (1991)], combined with appropriate frequency response equalization, with a compression threshold of 63 dB sound pressure level (SPL) and with a compression ratio of 30 (DUAL-HI); (2) The dual-front-end AGC system combined with appropriate frequency response equalization, with a compression threshold of 55 dB SPL and with a compression ratio of 3 (DUAL-LO). This was intended to give some impression of the levels of sounds in the environment; (3) Fast-acting full dynamic range compression in four channels (FULL-4). The compression was designed to minimize envelope distortion due to overshoots and undershoots; (4) A combination of (2) and (3) above, where each applied less compression than when used alone (DUAL-4). Initial fitting was partly based on the concept of giving a flat specific-loudness pattern for a 65-dB SPL speech-shaped noise input, and this was followed by fine tuning using an adaptive procedure with speech stimuli. Eight subjects with moderate to severe cochlear hearing loss were tested in a counter-balanced design. Subjects had at least 2 weeks experience with each system in everyday life before evaluation using the Abbreviated Profile of Hearing Aid Benefit (APHAB) test and measures of speech intelligibility in quiet (AB word lists at 50 and 80 dB SPL) and noise (adoptive sentence lists in speech-shaped noise, or that same noise amplitude modulated with the envelope of speech from a single talker). The APHAB scores did not indicate clear differences between the four systems. Scores for the AB words in quiet were high for all four systems at both 50 and 80 dB SPL. The speech-to-noise ratios required for 50% intelligibility were low (indicating good performance) and similar for all the systems, but there was a slight trend for better performance in modulated noise with the DUAL-4 system than with the other systems. A subsequent trial where three subjects directly compared each of the four systems in their everyday lives indicated a slight preference for the DUAL-LO system. Overall, the results suggest that it is not necessary to compress fast modulations of the input signal.     

Effects of signal-to-noise ratio, signal periodicity, and degree of hearing impairment on the performance of voice-separation algorithms

Procedures for enhancing the intelligibility of a target talker in the presence of a co-channel competing talker were evaluated in tests involving (i) continuously voiced sentences spoken on a monotone, (ii) continuously voiced sentences with time-varying intonation, and (iii) noncontinuously voiced sentences produced with natural intonation. The procedures were based on the methods of harmonic selection and cepstral filtering [R.J. Stubbs and Q. Summerfield, J. Acoust. Soc. Am. 87, 359-372 (1990)]. Target and competing voices were combined at signal-to-noise ratios (SNRs) between -10 dB and +10 dB. Subjects were a group with normal hearing and a heterogeneous group with mild-moderate cochlear hearing impairments. Processing enhanced the target voice over a range of SNRs for each type of sentence and for most listeners. Enhancement was greatest at negative SNRs. Among the impaired listeners, benefit was generally greater for those with milder losses. These results consolidate and extend previous demonstrations that voice-separation algorithms that exploit the harmonic structure of the voiced portions of speech can enhance intelligibility. However, practical application of such algorithms depends on a solution to the problem of tracking the fundamental-frequency contour of one voice in the presence of a competing voice.     

Spatial release from masking in normal-hearing children and children who use hearing aids

Listening to speech in competing sounds poses a major difficulty for children with impaired hearing. This study aimed to determine the ability of children (3-12 yr of age) to use spatial separation between target speech and competing babble to improve speech intelligibility. Fifty-eight children (31 with normal hearing and 27 with impaired hearing who use bilateral hearing aids) were assessed by word and sentence material. Speech reception thresholds (SRTs) were measured with speech presented from 0° azimuth, and competing babble from either 0° or ±90° azimuth. Spatial release from masking (SRM) was defined as the difference between SRTs measured with co-located speech and babble and SRTs measured with spatially separated speech and babble. On average, hearing-impaired children attained near-normal performance when speech and babble originated from the frontal source, but performed poorer than their normal-hearing peers when babble was spatially separated from target speech. On average, normal-hearing children obtained an SRM of 3 dB whereas children with hearing loss did not demonstrate SRM. Results suggest that hearing-impaired children may need enhancement in signal-to-noise ratio to hear speech in difficult listening conditions as well as normal-hearing children.