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Early assessment of hearing aid benefit is crucial, as the extent to which hearing aids provide audible speech information predicts speech and language outcomes. A growing body of research has proposed neural envelope tracking as an objective measure of speech intelligibility, particularly for individuals unable to provide reliable behavioral feedback. However, its potential for evaluating speech intelligibility and hearing aid benefit in children with hearing loss remains unexplored. In this study, we investigated neural envelope tracking in children with permanent hearing loss through two separate experiments. EEG data were recorded while children listened to age-appropriate stories (Experiment 1) or an animated movie (Experiment 2) under aided and unaided conditions (using personal hearing aids) at multiple stimulus intensities. Neural envelope tracking was evaluated using a linear decoder reconstructing the speech envelope from the EEG in the delta band (0.5-4 Hz). Additionally, we calculated temporal response functions (TRFs) to investigate the spatio-temporal dynamics of the response. In both experiments, neural tracking increased with increasing stimulus intensity, but only in the unaided condition. In the aided condition, neural tracking remained stable across a wide range of intensities, as long as speech intelligibility was maintained. Similarly, TRF amplitudes increased with increasing stimulus intensity in the unaided condition, while in the aided condition significant differences were found in TRF latency rather than TRF amplitude. This suggests that decreasing stimulus intensity does not necessarily impact neural tracking. Furthermore, the use of personal hearing aids significantly enhanced neural envelope tracking, particularly in challenging speech conditions that would be inaudible when unaided. Finally, we found a strong correlation between neural envelope tracking and behaviorally measured speech intelligibility for both narrated stories (Experiment 1) and movie stimuli (Experiment 2). Altogether, these findings indicate that neural envelope tracking could be a valuable tool for predicting speech intelligibility benefits derived from personal hearing aids in hearing-impaired children. Incorporating narrated stories or engaging movies expands the accessibility of these methods even in clinical settings, offering new avenues for using objective speech measures to guide pediatric audiology decision-making., Competing Interests: Declaration of Competing Interest None., (Copyright © 2023 Elsevier B.V. All rights reserved.)

Hearing aids (HA) are a fundamental component in restoring auditory function; however, they cannot completely alleviate all problems encountered by adults with hearing impairment. The aim of this study is twofold. Firstly, we assess the health-related quality of life and coping strategies of experienced HA users. Secondly, we assess whether HA users can benefit from auditory training. To this end, 40 participants who had worn HAs for more than 6 months participated in this study. Half of the participants received auditory training, while the other half served as a passive control. The training consisted of a personalized training scheme, with outcome measures including speech in noise perception in free-field and via direct streaming to the HA, phoneme identification, cognitive control, and health-related quality of life. Results showed that experienced HA users reported a relatively good quality of life. Health-related quality of life was correlated with aided speech perception in noise, but not with aided pure tone audiometry. Coping strategies were adaptive, leading to improved communication. Participants showed improvements in trained tasks, consonant identification, and speech in noise perception. While both groups yielded improved speech in noise perception at the end, post hoc analysis following a three-way interaction showed a significantly larger pre-post difference for the trained group in the streaming condition. Although training showed some improvements, the study suggests that the training paradigm was not sufficiently challenging for HA users. To optimize daily life listening, we recommend that future training should incorporate more exercises in noise and focus on cognitive control.

Objectives: We investigated the impact of both intrinsic and extrinsic cognitive demands on auditory and secondary task performance in older adults with normal hearing (NH) and adults using hearing aids (HAs) in an ecologically relevant listening environment., Design: Fifteen adults with NH and 15 adults using HAs (60 to 72 years of age) were recruited to perform the "Audiovisual True-to-Life Assessment of Auditory Rehabilitation"-paradigm (AVATAR), which combines an auditory-visual speech-in-noise task with three secondary tasks on either auditory localization or visual short-term memory in a preload multitask paradigm. Intrinsic demands were altered by presenting speech either at equivalent speech accuracy levels of 50% correct (SPIN50%) or equivalent speech-to-noise ratios of +5dB SNR (SPIN+5dB). We manipulated the amount of extrinsic cognitive demands by including one (dual condition) or three secondary tasks (quadruple condition). Performance decrements on the secondary tasks were considered to reflect an increase in resource allocation to speech understanding and thus an increase in listening effort. In addition, we administered a cognitive test battery as well as a questionnaire on self-reported hearing difficulties and subjective measures of effort and motivation., Results: Speech understanding and secondary task performance on the visual short-term memory task were lower in the SPIN50% condition compared to SPIN+5dB. Whereas speech understanding at SPIN50% was equally high in the dual and quadruple conditions, the quadruple condition resulted in lower secondary task performance on the visual short-term memory task, higher levels of self-reported effort, and lower motivation compared to the dual-task condition. The same was true for experimental conditions at SPIN+5dB. Additionally, adults with NH outperformed adults using HAs on speech understanding, auditory localization, and self-reported hearing abilities, but no group differences were observed on secondary task costs or cognitive measures., Conclusions: This study showed that, irrespective of the hearing status of the listener, speech performance was not affected by the amount of extrinsic cognitive demands, but was worse and required more effort under conditions with a more negative SNR. Also, increasing the extrinsic cognitive demands resulted in lower performance on one of the secondary tasks, suggesting that more complex listening environments require more effort. Although adults with NH outperformed HA users on speech understanding and auditory localization, the two groups did not differ with respect to secondary task costs., Competing Interests: The authors have no conflicts of interest to disclose., (Copyright © 2021 Wolters Kluwer Health, Inc. All rights reserved.)

One of the key components of fully-implantable hearing devices is the implantable microphone. A crucial parameter when characterizing implantable microphones is the acoustic sensitivity, as it is one of the input parameters for the fitting algorithm and influences the achievable gain. The aim of our study was to investigate the sensitivity of an implanted subcutaneous microphone over time to answer two research questions: (1) How does the sensitivity change once the microphone is implanted under the skin (pre-op versus in-situ)? and (2) How does the sensitivity change from short-term to mid-term? We have measured the in-situ microphone sensitivity in three subjects implanted with a fully-implantable active middle ear implant from 2 weeks up to 20 weeks after implantation with a research software. The microphone sensitivity changed after implantation with clinically relevant changes around the resonance frequency of the microphone. Based on our results, it is essential to measure the in-situ microphone sensitivity at the time of initial fitting of the implant. Once implanted no clinically relevant changes in the microphone sensitivity were observed over time, with a clear decrease in variability over time., (Copyright © 2020 Elsevier B.V. All rights reserved.)

Objectives: There is a high need among clinicians and researchers for an ecologically valid measure of auditory functioning and listening effort. Therefore, we developed AVATAR: an "Audiovisual True-to-life Assessment of Auditory Rehabilitation" which takes important characteristics of real-life listening situations into account, such as multimodal speech presentation, spatial separation of sound sources and multitasking. As such, AVATAR aims to assess both auditory functioning and the amount of allocated processing resources during listening in a realistic yet controllable way. In the present study, we evaluated AVATAR and investigated whether speech understanding in noise and multitask costs during realistic listening environments changed with increasing task complexity., Design: Thirty-five young normal-hearing participants performed different task combinations of an auditory-visual speech-in-noise task and three secondary tasks on both auditory localization and visual short-term memory in a simulated restaurant environment. Tasks were combined in increasing complexity and multitask costs on the secondary tasks were investigated as an estimate of the amount of cognitive resources allocated during listening and multitasking. In addition to behavioral measures of auditory functioning and effort, working memory capacity and self-reported hearing difficulties were established using a reading span test and a questionnaire on daily hearing abilities., Results: Whereas performance on the speech-in-noise task was not affected by task complexity, multitask costs on one of the secondary tasks became significantly larger with increasing task complexity. Working memory capacity correlated significantly with multitask costs, but no association was observed between behavioral outcome measures and self-reported hearing abilities or effort., Conclusions: AVATAR proved to be a promising model to assess speech intelligibility and auditory localization abilities and to gauge the amount of processing resources during effortful listening in ecologically relevant multitasking situations by means of multitask costs. In contrast with current clinical measures of auditory functioning, results showed that listening and multitasking in challenging listening environments can require a considerable amount of processing resources, even for young normal-hearing adults. Furthermore, the allocation of resources increased in more demanding listening situations. These findings open avenues for a more realistic assessment of auditory functioning and individually tuned auditory rehabilitation for individuals of different age and hearing profiles.

Objective: The recent integration of automated real-ear measurements (REM) in the fitting software facilitates the hearing aid fitting process. Such a fitting strategy, TargetMatch (TM), was evaluated. Test-retest reliability and matching accuracy were quantified, and compared to a REM-based fitting with manual adjustment. Also, it was investigated whether TM leads to better perceptual outcomes compared to a FirstFit (FF) approach, using software predictions only. Design and study sample: Ten hearing impaired participants were enrolled in a counterbalanced single-blinded cross-over study comparing TM and FF. Aided audibility, speech intelligibility and real-life benefits were assessed. Repeated measurements of both TM and REMs with manual adjustment were performed., Results: Compared to a REM-based fitting with manual adjustment, TM had higher test-retest reliability. Also, TM outperformed the other fitting strategies in terms of matching accuracy. Compared to a FF, improved aided audibility and real-life benefits were found. Speech intelligibility did not improve., Conclusions: Preliminary data suggest that automated REMs increase the likelihood of meeting amplification targets compared with a FF. REMs integrated in the fitting software provide additional reliability and accuracy compared to traditional REMs. Findings need to be verified in a larger and more varied sample.

State-of-the-art hearing aids (HAs) try to overcome the deficit of poor speech intelligibility (SI) in noisy listening environments using digital noise reduction (NR) techniques. The application of time-frequency masks to the noisy sound input is a common NR technique to increase SI. The binary mask with its binary weights and the Wiener filter with continuous weights are representatives of a hard- and a soft-decision approach for time-frequency masking. In normal-hearing listeners, the ideal Wiener filter (IWF) outperforms the ideal binary mask (IBM) in terms of SI and speech quality with perfect SI even at very low signal-to-noise ratios. In this paper, both approaches were investigated for hearing-impaired (HI) listeners. Perceptual and auditory model-based measures were used for the evaluation. The IWF outperformed the IBM in terms of SI. Quality-wise, there was no overall difference between the NR algorithms perceived. Additionally, the processed signals were evaluated based on an auditory nerve model using the neurogram similarity metric (NSIM). The mean NSIM values were significantly different for intelligible and unintelligible sentences. The results suggest that a soft-mask seems to be promising for application in HAs.

In Part I, we investigated 40-Hz auditory steady-state response (ASSR) amplitudes for the use of objective loudness balancing across the ears for normal-hearing participants and found median across-ear ratios in ASSR amplitudes close to 1. In this part, we further investigated whether the ASSR can be used to estimate binaural loudness balance for listeners with asymmetric hearing, for whom binaural loudness balancing is of particular interest. We tested participants with asymmetric hearing and participants with bimodal hearing, who hear with electrical stimulation through a cochlear implant (CI) in one ear and with acoustical stimulation in the other ear. Behavioral loudness balancing was performed at different percentages of the dynamic range. Acoustical carrier frequencies were 500, 1000, or 2000 Hz, and CI channels were stimulated in apical or middle regions in the cochlea. For both groups, the ASSR amplitudes at balanced loudness levels were similar for the two ears, with median ratios between left and right ear stimulation close to 1. However, individual variability was observed. For participants with asymmetric hearing loss, the difference between the behavioral balanced levels and the ASSR-predicted balanced levels was smaller than 10 dB in 50% and 56% of cases, for 500 Hz and 2000 Hz, respectively. For bimodal listeners, these percentages were 89% and 60%. Apical CI channels yielded significantly better results (median difference near 0 dB) than middle CI channels, which had a median difference of -7.25 dB.

Background: Several studies have demonstrated negative effects of directional microphone configurations on left-right and front-back (FB) sound localization. New processing schemes, such as frequency-dependent directionality and front focus with wireless ear-to-ear communication in recent, commercial hearing aids may preserve the binaural cues necessary for left-right localization and may introduce useful spectral cues necessary for FB disambiguation., Purpose: In this study, two hearing aids with different processing schemes, which were both designed to preserve the ability to localize sounds in the horizontal plane (left-right and FB), were compared., Research Design: We compared horizontal (left-right and FB) sound localization performance of hearing aid users fitted with two types of behind-the-ear (BTE) devices. The first type of BTE device had four different programs that provided (1) no directionality, (2-3) symmetric frequency-dependent directionality, and (4) an asymmetric configuration. The second pair of BTE devices was evaluated in its omnidirectional setting. This setting automatically activates a soft forward-oriented directional scheme that mimics the pinna effect. Also, wireless communication between the hearing aids was present in this configuration (5). A broadband stimulus was used as a target signal. The directional hearing abilities of the listeners were also evaluated without hearing aids as a reference., Study Sample: A total of 12 listeners with moderate to severe hearing loss participated in this study. All were experienced hearing-aid users. As a reference, 11 listeners with normal hearing participated., Data Collection and Analysis: The participants were positioned in a 13-speaker array (left-right, -90°/+90°) or 7-speaker array (FB, 0-180°) and were asked to report the number of the loudspeaker located the closest to where the sound was perceived. The root mean square error was calculated for the left-right experiment, and the percentage of FB errors was used as a FB performance measure. RESULTS were analyzed with repeated-measures analysis of variance., Results: For the left-right localization task, no significant differences could be proven between the unaided condition and both partial directional schemes and the omnidirectional scheme. The soft forward-oriented system and the asymmetric system did show a detrimental effect compared with the unaided condition. On average, localization was worst when users used the asymmetric condition. Analysis of the results of the FB experiment showed good performance, similar to unaided, with both the partial directional systems and the asymmetric configuration. Significantly worse performance was found with the omnidirectional and the omnidirectional soft forward-oriented BTE systems compared with the other hearing-aid systems., Conclusions: Bilaterally fitted partial directional systems preserve (part of) the binaural cues necessary for left-right localization and introduce, preserve, or enhance useful spectral cues that allow FB disambiguation. Omnidirectional systems, although good for left-right localization, do not provide the user with enough spectral information for an optimal FB localization performance., (American Academy of Audiology.)

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.

Users of a cochlear implant together with a contralateral hearing aid-so-called bimodal listeners-have difficulties with localizing sound sources. This is mainly due to the distortion of interaural time and level difference cues (ITD and ILD), and limited ITD sensitivity. An algorithm is presented that enhances ILD cues. Horizontal plane sound-source localization performance of six bimodal listeners was evaluated in (1) a real sound field with their clinical devices, (2) in a virtual sound field, under direct computer control, and (3) in a virtual sound field with ILD enhancement. The results in the real sound field did not differ significantly from the results in the virtual field, and ILD enhancement improved localization performance by 4°-10° absolute error, relative to a mean absolute error of 28° in the condition without ILD enhancement.

Sensitivity to interaural time differences (ITDs) with unmodulated low-frequency stimuli was assessed in bimodal listeners who had previously shown to be good performers in ITD experiments. Two types of stimuli were used: (1) an acoustic sinusoid combined with an electric transposed signal and (2) an acoustic sinusoid combined with an electric clicktrain. No or very low sensitivity to ITD was found for these stimuli, even though subjects were highly trained on the task and were intensively tested in multiple test sessions. In previous studies with users of a cochlear implant (CI) and a contralateral hearing aid (HA) (bimodal listeners), sensitivity was shown to ITD with modulated stimuli with frequency content between 600 and 3600 Hz. The outcomes of the current study imply that in speech processing design for users of a CI in combination with a HA on the contralateral side, the emphasis should be more on providing salient envelope ITD cues than on preserving fine-timing ITD cues present in acoustic signals., (© 2011 Acoustical Society of America)

Objective: The effect of different commercial hearing aids on the ability to resolve front-back confusions and on sound localization in the frontal horizontal and vertical plane was studied., Design: Commercial hearing aids with a microphone placed in-the-ear-canal (ITC), behind-the-ear (BTE), and in-the-pinna (ITP) were evaluated in the frontal and full horizontal plane, and in the frontal vertical plane., Study Sample: A group of 13 hearing-impaired subjects evaluated the hearing aids. Nine normal-hearing listeners were used as a reference group., Results and Conclusions: Differences in sound localization in the front-back dimension were found for different hearing aids. A large inter-subject variability was found during the front-back and elevation experiments. With ITP or ITC microphones, almost all natural spectral information was preserved. One of the BTE hearing aids, which is equipped with a directional microphone configuration, generated a sufficient amount of spectral cues to allow front-back discrimination. No significant effect of hearing aids on elevation performance in the frontal vertical plane was observed. Hearing-impaired subjects reached the same performance with and without the different hearing aids. In the unaided condition, a frequency-specific audibility correction was applied. Some of the hearing-impaired listeners reached normal hearing performance with this correction.

In a previous study, it was shown that users of a cochlear implant and a contralateral hearing aid are sensitive to interaural time differences (ITDs). In the current study, we investigated (1) the influence on ITD sensitivity of bilaterally varying the place of excitation in the cochlea and of modulation frequency, and (2) the sensitivity to ITD with a 3-channel stimulus generated using continuous-interleaved-sampling (CIS)-like processing. The stimuli were (1) a high-frequency carrier (acoustic sinusoid and single-electrode electric pulse train), modulated with a half-wave-rectified low-frequency sinusoid (a so-called transposed stimulus), and (2) a 3-channel stimulus, generated by sending an acoustic click train through processing similar to the CIS strategy. Four bimodal listeners were sensitive to ITD for both stimulus types. For the first stimulus type, there was no significant influence on ITD sensitivity of the acoustic carrier frequency. Performance decreased with increasing modulation frequency with a limit of sensitivity at around 150-200 Hz. Sensitivity was similar for the single- and 3-channel stimulus. The results indicate the possibility of ITD perception with adapted clinical processors, which can lead to improved sound source localization and binaural unmasking., (Copyright © 2010 S. Karger AG, Basel.)

This paper presents a physical evaluation of four feedback cancellation techniques in commercial hearing aids and two implementations of a recently developed feedback cancellation algorithm. Based on physical measures for detecting instability, oscillations and distortion, three performance aspects were measured: 1) the added stable gain compared to the hearing aid operating without feedback reduction for white noise as well as for spectrally colored input signals in two static acoustic conditions, 2) the amount of feedback, oscillations and distortion at gain values below the maximum stable gain, 3) the ability to track feedback path changes. Added stable gains between 3 dB and 26 dB were identified. Five of the six techniques achieve worse feedback reduction for a tonal opera input signal than for a speech input signal. Preventing the feedback canceller to drift away from an initial feedback path measurement results in improved performance for tonal signals at the expense of a worse feedback reduction in the acoustic conditions that differ from the condition for which the initialization was performed, as well as a worse tracking of feedback path changes. Repeated measures indicated that the reproducibility of the test set-up is crucial, in particular when the hearing aid operates close to instability.

In the framework of the European HearCom project, promising signal enhancement algorithms were developed and evaluated for future use in hearing instruments. To assess the algorithms' performance, five of the algorithms were selected and implemented on a common real-time hardware/software platform. Four test centers in Belgium, The Netherlands, Germany, and Switzerland perceptually evaluated the algorithms. Listening tests were performed with large numbers of normal-hearing and hearing-impaired subjects. Three perceptual measures were used: speech reception threshold (SRT), listening effort scaling, and preference rating. Tests were carried out in two types of rooms. Speech was presented in multitalker babble arriving from one or three loudspeakers. In a pseudo-diffuse noise scenario, only one algorithm, the spatially preprocessed speech-distortion-weighted multi-channel Wiener filtering, provided a SRT improvement relative to the unprocessed condition. Despite the general lack of improvement in SRT, some algorithms were preferred over the unprocessed condition at all tested signal-to-noise ratios (SNRs). These effects were found across different subject groups and test sites. The listening effort scores were less consistent over test sites. For the algorithms that did not affect speech intelligibility, a reduction in listening effort was observed at 0 dB SNR.

This paper evaluates speech enhancement in binaural multimicrophone hearing aids by noise reduction algorithms based on the multichannel Wiener filter (MWF) and the MWF with partial noise estimate (MWF-N). Both algorithms are specifically developed to combine noise reduction with the preservation of binaural cues. Objective and perceptual evaluations were performed with different speech-in-multitalker-babble configurations in two different acoustic environments. The main conclusions are as follows: (a) A bilateral MWF with perfect voice activity detection equals or outperforms a bilateral adaptive directional microphone in terms of speech enhancement while preserving the binaural cues of the speech component. (b) A significant gain in speech enhancement is found when transmitting one contralateral microphone signal to the MWF active at the ipsilateral hearing aid. Adding a second contralateral microphone showed a significant improvement during the objective evaluations but not in the subset of scenarios tested during the perceptual evaluations. (c) Adding the partial noise estimate to the MWF, done to improve the spatial awareness of the hearing aid user, reduces the amount of speech enhancement in a limited way. In some conditions the MWF-N even outperformed the MWF possibly due to an improved spatial release from masking.

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.

Binaural hearing provides advantages over monaural in several ways, particularly in difficult listening situations. For a person with bilateral hearing loss, the bilateral fitting of hearing aids thus seems like a natural choice. However, surprisingly few studies have been reported in which the additional benefit of bilateral versus unilateral hearing aid use has been investigated based on real-life experiences. Therefore, a project has been designed to address this issue and to find tools to identify people for whom the drawbacks would outweigh the advantages of bilateral fitting. A project following this design is likely to provide reliable evidence concerning differences in benefit between unilateral and bilateral fitting of hearing aids by evaluating correlations between entrance data and outcome measures and final preferences.

The interaural level difference (ILD) is an important cue for the localization of sound sources. The sensitivity to ILD was measured in 10 users of a cochlear implant (CI) in one ear and a hearing aid (HA) in the other severely impaired ear. For simultaneous presentation of a pulse train on the CI side and a sinusoid on the HA side the just noticeable difference (JND) in ILD and loudness growth functions were measured. The mean JND for pitch-matched electric and acoustic stimulation was 1.7 dB. A linear fit of the loudness growth functions on a decibel-versus-microampere scale shows that the slope depends on the subject's dynamic ranges., (Copyright 2008 S. Karger AG, Basel.)

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.

In this paper, the first real-time implementation and perceptual evaluation of a singular value decomposition (SVD)-based optimal filtering technique for noise reduction in a dual microphone behind-the-ear (BTE) hearing aid is presented. This evaluation was carried out for a speech weighted noise and multitalker babble, for single and multiple jammer sound source scenarios. Two basic microphone configurations in the hearing aid were used. The SVD-based optimal filtering technique was compared against an adaptive beamformer, which is known to give significant improvements in speech intelligibility in noisy environment. The optimal filtering technique works without assumptions about a speaker position, unlike the two-stage adaptive beamformer. However this strategy needs a robust voice activity detector (VAD). A method to improve the performance of the VAD was presented and evaluated physically. By connecting the VAD to the output of the noise reduction algorithms, a good discrimination between the speech-and-noise periods and the noise-only periods of the signals was obtained. The perceptual experiments demonstrated that the SVD-based optimal filtering technique could perform as well as the adaptive beamformer in a single noise source scenario, i.e., the ideal scenario for the latter technique, and could outperform the adaptive beamformer in multiple noise source scenarios.

Objective: To evaluate the improvement in speech intelligibility in noise obtained with an assistive real-time fixed endfire array of bidirectional microphones in comparison with an omnidirectional hearing aid microphone in a realistic environment., Design: The microphone array was evaluated physically in anechoic and reverberant conditions. Perceptual tests of speech intelligibility in noise were carried out in a reverberant room, with two types of noise and six different noise scenarios with single and multiple noise sources. Ten normal-hearing subjects and 10 hearing aid users participated. The speech reception threshold for sentences was measured in each test setting for the omnidirectional microphone of the hearing aid and for the hearing aid in combination with the array with one and three active microphones. In addition, the extra improvement of five active array microphones, relative to three, was determined in another group of 10 normal-hearing listeners., Results: Improvements in speech intelligibility in noise obtained with the array relative to an omnidirectional microphone depend on noise scenario and subject group. Improvements up to 12 dB for normal-hearing and 9 dB for hearing-impaired listeners were obtained with three active array microphones relative to an omnidirectional microphone for one noise source at 90 degrees . For three uncorrelated noise sources at 90 degrees, 180 degrees, and 270 degrees, improvements of approximately 9 dB and 6 dB were obtained for normal-hearing and hearing-impaired listeners, respectively. Even with a single noise source at 45 degrees, benefits of 4 dB were achieved in both subject groups. Five active microphones in the array can provide an additional improvement at 45 degrees of approximately 1 dB, relative to the three-microphone configuration for normal-hearing listeners., Conclusions: These improvements in signal-to-noise ratio can be of great benefit for hearing aid users, who have difficulties with speech understanding in noisy environments.

Objective: The performance of an adaptive beam-former in a 2-microphone, behind-the-ear hearing aid for speech understanding in noisy environments was evaluated. Physical and perceptual evaluations were carried out. This was the first large-scale test of a wearable real-time implementation of this algorithm. The main perceptual research questions of this study were related to the influence on the noise reduction performance of (1) the spectro-temporal character of the jammer sound, (2) the jammer sound scene, (3) hearing impairment, and (4) the basic microphone configuration in the hearing aid. Four different speech materials were used for the perceptual evaluations. All tests were carried out in an acoustical environment comparable to living room reverberation., Design: The adaptive beamformer was implemented in Audallion, a small, body-worn processor, linked to a Danasound 2-microphone behind-the-ear aid. The strategy was evaluated physically in different acoustical environments. Using speech reception threshold (SRT) measurements, the processing was evaluated perceptually and the different research questions addressed with three groups of subjects. Groups I, II, and III consisted of 10 normal-hearing, 5 hearing-impaired, and 7 normal-hearing persons, respectively. The tests were carried out in three spectro-temporally different jammer sounds (unmodulated and modulated speech weighted noise, multitalker babble) and in three different noise scenarios (single noise source at 90 degrees, noise sources at 90 degrees and 270 degrees relative to speaker position, diffuse noise scene). Two microphone configurations were compared: a device equipped with two omnidirectional microphones and a device equipped with one hardware directional and one omnidirectional microphone. In each of these conditions, the adaptive beamformer and the directional and omnidirectional microphone configurations were tested., Results: The improvement in signal-to-noise ratio from the use of the adaptive beamformer did not depend on the spectro-temporal character of the jammer sounds and the speech materials used, although the absolute levels of the SRTs varied appreciably for different speech-noise combinations. The performance of the adaptive noise reduction depended on the jammer sound scene., Conclusions: No difference in signal-to-noise ratio improvement was observed between hearing-impaired and normal-hearing listeners, although individual SRT levels may differ. On average, an SRT improvement of 7.7 and 3.9 dB for a single noise source at 90 degrees and 5.9 and 3.4 dB for two noise sources at 90 degrees and 270 degrees was obtained for both normal-hearing and hearing-impaired listeners, using the adaptive beamformer and the directional microphone, respectively, relative to the omnidirectional microphone signal. In diffuse noise, only small improvements were obtained., (Copyright 2004 Lippincott Williams and Wilkins)

An adaptive beamformer for behind-the-ear dual-microphone hearing aids has been optimized for speech intelligibility enhancement in the presence of disturbing sounds or noise. The noise reduction approach is based on the scheme presented by Vanden Berghe and Wouters (1998). A real-time implementation of the signal processing is realized in Audallion, a wearable, small digital signal processing (DSP) platform. After physical evaluation, speech-in-noise intelligibility tests have been carried out on three normally-hearing and two hearing-impaired subjects. A significant speech reception threshold improvement of 11.3 dB was obtained in a moderately reverberant environment for one jammer sound source (steady speech-weighted noise or multi-talker babble) in a direction of 90 degrees relative to the direction of the speech.

We introduce a new SVD-based (Singular value decomposition) strategy for noise reduction in hearing aids. This technique is evaluated for noise reduction in a behind-the-ear (BTE) hearing aid where two omnidirectional microphones are mounted in an endfire configuration. The behaviour of the SVD-based technique is compared to a two-stage adaptive beamformer for hearing aids developed by Vanden Berghe and Wouters (1998). The evaluation and comparison is done with a performance metric based on the speech intelligibility index (SII). The speech and noise signals are recorded in reverberant conditions with a signal-to-noise ratio of and the spectrum of the noise signals is similar to the spectrum of the speech signal. The SVD-based technique works without initialization nor assumptions about a look direction, unlike the two-stage adaptive beamformer. Still, for different noise scenarios, the SVD-based technique performs as well as the two-stage adaptive beamformer, for a similar filter length and adaptation time for the filter coefficients. In a diffuse noise scenario, the SVD-based technique performs better than the two-stage adaptive beamformer and hence provides a more flexible and robust solution under speaker position variations and reverberant conditions.

Despite many considerable technical advances in the field of hearing aids and cochlear implants, people using auditory prostheses still have major problems with speech understanding in the presence of interfering sounds and with directional hearing. Both abilities are dependent on sound stream segregation in real-world listening environments. In this paper, two timely and important issues related to sound stream segregation in auditory prostheses are addressed, namely, the coding of monaural and binaural cues. Several state-of-the-art signal processing algorithms used in cochlear implants (CIs) and in hearing aids (HAs) are introduced. A review is given of some recent proposals to improve temporal coding in monaural CIs, and of recent work to improve the transmission of binaural cues in both HAs, CIs, and combined acoustic and electric hearing (bimodal hearing). The ultimate aim is to improve speech and music perception, and, additionally, the preservation of binaural cues to preserve directional hearing. [ABSTRACT FROM PUBLISHER]

Users of bilateral cochlear implants and a cochlear implant combined with a contralateral hearing aid are sensitive to interaural time differences (ITDs). The way cochlear implant speech processors work and differences between modalities may result in interaural differences in shape of the temporal envelope presented to the binaural system. The effect of interaural differences in envelope shape on ITD sensitivity was investigated with normal-hearing listeners using a 4 kHz pure tone modulated with a periodic envelope with a trapezoid shape in each cycle. In one ear the onset segment of the trapezoid was transformed by a power function. No effect on the just noticeable difference in ITD was found with an interaural difference in envelope shape, but the ITD for a centered percept was significantly different across envelope shape conditions. [ABSTRACT FROM AUTHOR]

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This paper presents combined active noise control and noise reduction schemes for hearing aids to tackle secondary path effects and effects of noise leakage through an open fitting. While such leakage contributions and the secondary acoustic path from the loudspeaker to the tympanic membrane are usually not taken into account in standard noise reduction systems, they appear to have a non-negligible impact on the final signal-to-noise ratio. Using a noise-reduction algorithm and an active noise control system in cascade may be efficient as long as the causality margin of the system is large enough. Putting the two functional blocks in parallel and then integrating them is found to lead to a more robust algorithm. A Filtered-x Multichannel Wiener Filter is presented and applied to integrate noise reduction and active noise control. The cascaded scheme and the integrated scheme are compared experimentally with a Multichannel Wiener Filter in a classic noise reduction framework without active noise control, where the integrated scheme is found to provide the best performance. [ABSTRACT FROM PUBLISHER]

Binaural hearing aids use microphone signals from both left and right hearing aid to generate an output signal for each ear. The microphone signals can be processed by a procedure based on speech distortion weighted multichannel wiener filtering (SDW-MWF) to achieve significant noise reduction in a speech + noise scenario. In binaural procedures, it is also desirable to preserve binaural cues, in particular the interaural time difference (ITD) and interaural level difference (ILD), which are used to localize sounds. It has been shown in previous work that the binaural SDW-MWF procedure only preserves these binaural cues for the desired speech source, but distorts the noise binaural cues. Two extensions of the binaural SDW-MWF have therefore been proposed to improve the binaural cue preservation, namely the MWF with partial noise estimation (MWF-η) and MWF with interaural transfer function extension (MWF-ITF). In this paper, the binaural cue preservation of these extensions is analyzed theoretically and tested based on objective performance measures. Both extensions are able to preserve binaural cues for the speech and noise sources, while still achieving significant noise reduction performance. [ABSTRACT FROM AUTHOR]

Two forward-masking experiments were conducted with six cochlear implant listeners to test whether asymmetric pulse shapes would improve the place-specificity of stimulation compared to symmetric ones. The maskers were either cathodic-first symmetric biphasic, pseudomonophasic (i.e., with a second anodic phase longer and lower in amplitude than the first phase), or “delayed pseudomonophasic” (identical to pseudomonophasic but with an inter-phase gap) stimuli. In experiment 1, forward-masking patterns for monopolar maskers were obtained by keeping each masker fixed on a middle electrode of the array and measuring the masked thresholds of a monopolar signal presented on several other electrodes. The results were very variable, and no difference between pulse shapes was found. In experiment 2, six maskers were used in a wide bipolar (bipolar+9) configuration: the same three pulse shapes as in experiment 1, either cathodic-first relative to the most apical or relative to the most basal electrode of the bipolar channel. The pseudomonophasic masker showed a stronger excitation proximal to the electrode of the bipolar pair for which the short, high-amplitude phase was anodic. However, no difference was obtained with the symmetric and, more surprisingly, with the delayed pseudomonophasic maskers. Implications for cochlear implant design are discussed. [ABSTRACT FROM AUTHOR]

We present sound localization results from 30 children with bilateral cochlear implants. All children received their implants sequentially, at ages from 6 months to 9 years for the first implant and 1.5–12 years for the second implant, with delays of 10 months to 9 years. Localization was measured in the sound field, with a broadband bell-ring presented from 1 of 9 loudspeakers positioned in the frontal horizontal plane. The majority of the children (63%) were able to localize this signal significantly better than chance level. Mean absolute error scores varied from 9 to 51° (root mean square error scores from 13 to 63°). The best scores were obtained by children who received their first implant before the age of 2 years and by children who used hearing aids prior to implantation for a period of 18 months or longer. Age at second implantation was important in the group of children who did not use a contralateral hearing aid during the unilateral implant period. Additionally, children who attended a mainstream school had significantly better localization scores than children who attended a school for the deaf. No other child or implantation variables were related to localization performance. Data of parent questionnaires derived from the Speech, Spatial and Qualities of Hearing Scale were significantly correlated with localization performance. This study shows that the sound localization ability of children with bilateral cochlear implants varies across subjects, from near-normal to chance performance, and that stimulation early in life, acoustically or electrically, is important for the development of this capacity. Copyright © 2009 S. Karger AG, Basel [ABSTRACT FROM AUTHOR]

Users of a cochlear implant and contralateral hearing aid are sensitive to interaural level differences (ILDs). However, when using their clinical devices, most of these subjects cannot use ILD cues for localization in the horizontal plane. This is partly due to a lack of high-frequency residual hearing in the acoustically stimulated ear. Using acoustic simulations of a cochlear implant and hearing loss, it is shown that localization performance can be improved by up to 14° rms error relative to 48° rms error for broadband noise by artificially introducing ILD cues in the low frequencies. The algorithm that was used for ILD introduction is described. [ABSTRACT FROM AUTHOR]

On the basis of a statistical evaluation of the binaural sound source localization performance during listening tests by human subjects, it is shown that the convolution of a measured head-related transfer function (HRTF) with the room impulse response generated by a hybrid image source model with a stochastic scattering process using secondary sources provides an adequate model for the prediction of binaural room impulse responses (BRIR) for directional sound localization in the frontal horizontal plane. The source localization performance for sound stimuli presented to test subjects in a natural way was compared to that presented via headphones. Listening via headphones tends to decrease the localization performance only, and only slightly when localizing narrow-band high-frequency stimuli. Binaural sound presented to test subjects via headphones was generated on the basis of measurements and simulations. Two different headphone conditions were evaluated. The overall localization performance for simulated headphone sound obtained using a simulated BRIR was found to be equivalent to the one using a measured BRIR. The study also confirms expectations that the deterioration of sound source localization performance in reverberant rooms is closely linked to the direct-to-reverberant ratio for given loudspeaker and listener positions, rather than to the reverberation time of the room as such. [ABSTRACT FROM AUTHOR]

Auditory steady-state responses (ASSRs) in EEG measurements are currently used for reliable hearing threshold estimation at audiometric frequencies. Especially, newborns with hearing problems benefit from this technique, as with this information, diagnosis can be better specified and hearing aids can be better fitted at an early age. Unfortunately, measurement duration is still very long for clinical widespread use due to the lack of efficient signal detection techniques with sufficient robustness against artifacts. In this paper, a simplified procedural framework for ASSR detection is worked out that allows the development of a multichannel processing strategy, starting from a detection theory approach. It is shown that a sufficient statistic can be calculated that best captures the amount of ASSR in the recorded data. The evaluation is conducted using data from ten normal-hearing adults. It is concluded that most single- and multichannel approaches are similar in performance when applied to uncontaminated EEG. When artifact-rich EEG is used, the proposed detection-theory-based approach significantly improves the number of ASSR detections compared with a noise-weighted common EEG channel derivation (vertex-occiput). [ABSTRACT FROM AUTHOR]

A multi-channel noise reduction technique is presented based on a Speech Distortion-Weighted Multi-channel Wiener Filter (SDW-MWF) approach that incorporates the conditional Speech Presence Probability (SPP). A traditional SDW-MWF uses a fixed parameter to a trade-off between noise reduction and speech distortion without taking speech presence into account. Consequently, the improvement in noise reduction comes at the cost of a higher speech distortion since the speech dominant segments and the noise dominant segments are weighted equally. Incorporating the conditional SPP in SDW-MWF allows to exploit the fact that speech may not be present at all frequencies and at all times, while the noise can indeed be continuously present. In speech dominant segments it is then desirable to have less noise reduction to avoid speech distortion, while in noise dominant segments it is desirable to have as much noise reduction as possible. Experimental results with hearing aid scenarios demonstrate that the proposed SDW-MWF incorporating the conditional SPP improves the signal-to-noise ratio compared to a traditional SDW-MWF. [ABSTRACT FROM AUTHOR]

In a binaural hearing aid system, output signals need to be generated for the left and the right ear. Using the binaural multichannel Wiener filter (MWF), which exploits all microphone signals from both hearing aids, a significant reduction of background noise can be achieved. However, due to power and bandwidth limitations of the binaural link, it is typically not possible to transmit all microphone signals between the hearing aids. To limit the amount of transmitted information, this paper presents reduced-bandwidth MWF-based noise reduction algorithms, where a filtered combination of the contralateral microphone signals is transmitted. A first scheme uses a signal-independent beamformer, whereas a second scheme uses the output of a monaural MWF on the contralateral microphone signals and a third scheme involves an iterative distributed MWF (DB-MWF) procedure. It is shown that in the case of a rank-1 speech correlation matrix, corresponding to a single speech source, the DB-MWF procedure converges to the binaural MWF solution. Experimental results compare the noise reduction performance of the reduced-bandwidth algorithms with respect to the benchmark binaural MWF. It is shown that the best performance of the reduced-bandwidth algorithms is obtained by the DB-MWF procedure and that the performance of the DB-MWF procedure approaches quite well the optimal performance of the binaural MWF. [ABSTRACT FROM AUTHOR]

The interaural level difference (ILD) is an important cue for the localization of sound sources. Just noticeable differences (JND) in ILD were measured in 12 normal hearing subjects for uncorrelated noise bands with a bandwidth of 1/3 octave and a different center frequency in both ears. In one ear the center frequency was either 250, 500, 1000, or 4000 Hz. In the other ear, a frequency shift of 0, 1/6, 1/3, or 1 octave was introduced. JNDs in ILD for unshifted, uncorrelated noise bands of 1/3 octave width were 2.6, 2.6, 2.5, and 1.4 dB for 250, 500, 1000, and 4000 Hz, respectively. Averaged over all shifts, JNDs decreased significantly with increasing frequency. For the shifted conditions, JNDs increased significantly with increasing shift. Performance on average worsened by 0.5, 0.9, and 1.5 dB for shifts of 1/6, 1/3, and 1 octave. Though performance decreases, the just noticeable ILDs for the shifted conditions were still in a range usable for lateralization. This has implications for signal processing algorithms for bilateral bimodal hearing instruments and the fitting of bilateral cochlear implants. [ABSTRACT FROM AUTHOR]

In this paper, solutions for combined feedback and noise suppression in hearing aids are developed. The techniques presented are based on the generalized sidelobe canceller (GSC) and adaptive feedback canceller (AFC), with a prediction error method (PEM) adaptation to avoid speech distortion. Two possible cascades of GSC-based noise reduction and AFC, namely an ldquoAFC firstrdquo and a ldquoGSC first,rdquo as well as a truly integrated solution that jointly suppresses feedback and noise are discussed. The integrated solution (called PEM-GFIC) achieves optimum synergies between noise and feedback suppression at the lowest computational cost. In addition, it cancels more feedback than the generalized echo and interference canceller, a joint solution for echo and noise suppression. In the cascaded solutions, the feedback and noise suppression filters are not always optimally exploited. For high input SNRs, ldquoAFC firstrdquo scheme generally may achieve better feedback cancellation because of its larger number of degrees of freedom. However, noise reduction by the GSC-stage seriously affects the feedback cancellation performance. At low SNRs, ldquoGSC firstrdquo generally achieves more feedback cancellation than PEM-GFIC at the expense of worse noise reduction. At high hearing aid gains and/or large SNRs, the noise reduction stage however negatively affects the performance of the feedback cancellation filter, resulting in a worse feedback and noise suppression compared to PEM-GFIC. [ABSTRACT FROM PUBLISHER]

The standard continuous adaptation feedback cancellation algorithm for feedback suppression in hearing aids suffers from a large model error or bias if the received sound signal is spectrally colored. To reduce the bias in the feedback path estimate, we propose adaptive feedback cancellation techniques that are based on a closed-loop identification of the feedback path as well as the (auto-regressive) modeling of the desired signal. In general, both models are not simultaneously identifiable in the closed-loop system at hand. We show that—under certain conditions, e.g., if a delay is inserted in the forward path—identification of both models is indeed possible. Two classes of adaptive procedures for identifying the desired signal model and the feedback path are derived: a two-channel identification method as well as a prediction error method. In contrast to the two-channel identification method, the prediction error method allows use of different adaptation schemes for the feedback path and for the desired signal model and, hence, is found to be preferable for highly non-stationary sound signals. Simulation results demonstrate that the proposed techniques outperform the standard continuous adaptation algorithm if the conditions for identifiability are satisfied. [ABSTRACT FROM AUTHOR]

For small-sized arrays such as hearing aids, noise reduction is obtained at the expense of an increased sensitivity to errors in the assumed signal model, i.e., microphone mismatch, variations in speaker and microphone positions, reverberation. However, the noise reduction algorithm should still be robust, i.e., insensitive to small signal model errors. In this paper, we evaluate the robustness of the Generalized Sidelobe Canceller (GSC) and a recently developed Multichannel Wiener Filtering (MWF) technique for hearing aid applications both analytically and experimentally. The analysis reveals that robustness of the GSC is especially crucial in complicated noise scenarios and that microphone mismatch is particularly harmful to the GSC, even when the adaptive noise canceller is adapted during noise only. Hence, a constraint on the noise sensitivity of the GSC is essential, at the expense of less noise reduction. The MWF on the other hand, is not affected by microphone mismatch and has a potential benefit over the robust GSC with noise sensitivity constraint. However, the MWF is sensitive to the estimation accuracy of the second order statistics of speech and noise so that its benefit may be lost in nonstationary noise scenarios. [ABSTRACT FROM AUTHOR]

Recently, a generalized noise reduction scheme has been proposed, called the Spatially Preprocessed, Speech Distortion Weighted, Multichannel Wiener Filter (SP-SDW-MWF). It encompasses the Generalized Sidelobe Canceller (GSC) and a multichannel Wiener filtering technique as extreme cases. Compared with the widely studied GSC with Quadratic Inequality Constraint (QICuGSC), the SP-SDW-MWF achieves a better noise reduction performance for a given maximum speech distortion level. In this paper, we develop a low-cost, stochastic gradient implementation of the SP-SDW-MWE To speed up convergence and reduce computational complexity, the algorithm is implemented in the frequency domain. Experimental results with a behind-the-ear hearing aid show that the proposed frequency-domain stochastic gradient algorithm preserves the benefit of the exact SP-SDW-MWF over the QIC-GSC, while its computational cost is comparable to the least mean square-based scaled projection algorithm for QIC-GSC. [ABSTRACT FROM AUTHOR]

Cochlear implant recipients perceive a rise in pitch when the site of stimulation is moved from the apex toward the base. The place pitch sensitivity is typically measured using the stimulation of single channels. However, all current cochlear implant devices stimulate multiple channels simultaneously or with pulses temporally interleaved. The primary goal of the present study is to test whether the sensitivity of a cochlear implant recipient to changes in perceived pitch associated with changes of place of excitation improves or deteriorates when the number of active channels is increased, compared with stimulation with only one active channel. Place pitch sensitivity was recorded in four Nucleus CI24 subjects as a function of number of active channels (from 1 to 8). Just noticeable differences were estimated from a constant stimuli 2AFC pitch-ranking experiment with roving loudness. Reference and comparison stimuli contained the same number of active channels but were shifted one or two electrodes toward the base or toward the apex. The place pitch sensitivity was measured using monopolar stimulation at two locations along the electrode array. To minimize cues related to loudness, the multichannel stimuli were loudness balanced relative to the single-channel stimuli presented at C-level. The number of active channels did not affect place pitch sensitivity. This is consistent with a model that compares the edges of the excitation pattern irrespective of the overlap between excitation patterns. There was a significant difference in sensitivity to place pitch among subjects. The average just noticeable differences of place pitch, extrapolated from a fitting procedure, for the subjects ranged from 0.25 mm to 0.46 mm. [ABSTRACT FROM AUTHOR]

Objectives: In contrast to asymmetric hearing (AH), normal- hearing (NH) listeners perceive balanced loudness at both ears, leading to good sound source localization skills. We aimed to objectively measure binaural balanced loudness in case of AH. Material and methods: We tested NH and AH listeners in two tasks: 1) EEC was recorded to the stimuli presented monaurally at different intensities to obtain 40-Hz ASSR amplitude growth of both ears, and 2) behavioral binaural loudness balancing was performed. Results: For NH listeners, who have equal loudness growth across the ears, the difference between the left and right ear 40-Hz ASSR amplitude growth was small. For listeners with an AHL, similar ASSR amplitudes were found corresponding to similar loudness. Conclusions: This demonstrates the potential of using 40-Hz ASSRs to objectively find the binaural balanced loudness for fitting in case of AH. Such an objective measure can contribute to more automatic and objective fitting of hearing aids. [ABSTRACT FROM AUTHOR]

Abstract: This paper presents weighted approaches for integrated active noise control and noise reduction in hearing aids. The unweighted integrated active noise control and noise reduction scheme introduced in the previous work does not allow to trade-off between the active noise control and the noise reduction. In some circumstances it will, however, be useful to emphasize one of the functional blocks. Changing the original optimisation problem to a constrained optimisation problem leads to a scheme based on a weighted mean squared error criterion that allows to focus either on the active noise control or on the noise reduction. It is similarly possible to derive a scheme that allows to focus either on reducing the speech distortion or on reducing the residual noise at the eardrum. In a single speech source scenario and when the number of sound sources (speech plus noise sources) is less than or equal to the number of microphones, it is possible to derive a simple formula for the output signal-to-noise ratio of the latter scheme. It can then be shown that this scheme delivers a constant signal-to-noise ratio at the eardrum for any weighting factor. [Copyright &y& Elsevier]