Development of a new method for deriving initial fittings for hearing aids with multi-channel compression: CAMEQ2-HF

Validity and reliability of integrated pressure level real-ear-to-coupler difference measurements

OBJECTIVES

(1) To validate the measurement of foam-tip real-ear-to-coupler differences (wRECD) using an integrated pressure level (IPL) method and (2) to compare the reliability of this method to SPL-based measurement of the wRECD.

DESIGN

SPL-based wRECD and the proposed IPL wRECD measurement were completed bilaterally. Test-retest reliability of IPL wRECD was determined with full re-insertion into the ear canal and compared to published SPL wRECD test-retest data.

STUDY SAMPLE

22 adults with normal hearing and middle ear status were recruited.

RESULTS

Differences between SPL-based wRECD and IPL wRECD measurements were within 1.51 dB on average below 5000 Hz. At and above 5000 Hz, IPL wRECD exceeded SPL wRECDs by 6.11 dB on average. The average test-retest difference for IPL wRECD across all assessed frequencies was 0.75 dB with the greatest improvements in reliability found below 750 Hz and above 3000 Hz.

CONCLUSIONS

IPL wRECD yielded improved estimates compared to SPL wRECD in high frequencies, where standing-wave interference is present. Independence from standing wave interference resulted in increased wRECD values above 4000 Hz using the IPL measurement paradigm. IPL wRECD is more reliable than SPL wRECD, does not require precise probe-microphone placement, and provides a wider valid wRECD bandwidth than SPL-based measurement.

The Cambridge aided loudness profile (CALP): item choices and data for unaided listening by young and older normal-hearing listeners and older hearing-impaired listeners

OBJECTIVE

To develop an improved version of the profile of aided loudness (PAL), intended for assessment of the appropriateness of the loudness of everyday sounds.

DESIGN

Initially, 16 participants with a range of ages and degrees of hearing loss indicated whether they encountered each situation described in the PAL and how specific they considered the description to be. Based on the responses, most situations from the PAL were eliminated and new situations were introduced, giving the Cambridge Aided Loudness Profile (CALP). The CALP was administered to 80 young and 22 older participants with normal hearing, who rated the loudness of each situation and satisfaction with this loudness (as for the original PAL). Satisfaction was strongly negatively correlated with loudness, suggesting that satisfaction was largely based on loudness. The CALP was then administered to 32 new young normal-hearing participants and 49 older participants with hearing loss, most of whom used hearing aids, who rated loudness and the appropriateness of loudness.

RESULTS

Some situations were rated as loud but appropriate in loudness, indicating that the CALP can distinguish these aspects.

CONCLUSIONS

The CALP questions were understood by all participants. The CALP may be useful for assessing the appropriateness of loudness.

Neural hyperactivity and altered envelope encoding in the central auditory system: Changes with advanced age and hearing loss

Temporal modulations are ubiquitous features of sound signals that are important for auditory perception. The perception of temporal modulations, or temporal processing, is known to decline with aging and hearing loss and negatively impact auditory perception in general and speech recognition specifically. However, neurophysiological literature also provides evidence of exaggerated or enhanced encoding of specifically temporal envelopes in aging and hearing loss, which may arise from changes in inhibitory neurotransmission and neuronal hyperactivity. This review paper describes the physiological changes to the neural encoding of temporal envelopes that have been shown to occur with age and hearing loss and discusses the role of disinhibition and neural hyperactivity in contributing to these changes. Studies in both humans and animal models suggest that aging and hearing loss are associated with stronger neural representations of both periodic amplitude modulation envelopes and of naturalistic speech envelopes, but primarily for low-frequency modulations (<80 Hz). Although the frequency dependence of these results is generally taken as evidence of amplified envelope encoding at the cortex and impoverished encoding at the midbrain and brainstem, there is additional evidence to suggest that exaggerated envelope encoding may also occur subcortically, though only for envelopes with low modulation rates. A better understanding of how temporal envelope encoding is altered in aging and hearing loss, and the contexts in which neural responses are exaggerated/diminished, may aid in the development of interventions, assistive devices, and treatment strategies that work to ameliorate age- and hearing-loss-related auditory perceptual deficits.

Validation of an integrated pressure level measured earmold wideband real-ear-to-coupler difference measurement

OBJECTIVE

To validate measurement of predicted earmold wideband real-ear-to-coupler difference (wRECD) using an integrated pressure level (IPL) calibrated transducer and the incorporation of an acoustically measured tubing length correction.

DESIGN

Unilateral earmold SPL wRECD using varied hearing aid tubing length and the proposed predicted earmold IPL wRECD measurement procedure were completed on all participants and compared.

STUDY SAMPLE

22 normal hearing adults with normal middle ear status were recruited.

RESULTS

There were no clinically significant differences between probe-microphone and predicted earmold IPL wRECD measurements between 500 and 2500 Hz. Above 5000 Hz, the predicted earmold IPL wRECD exceeded earmold SPL wRECDs due to lack of standing wave interference. Test-retest reliability of IPL wRECD measurement exceeded the reliability of earmold SPL wRECD measurement across all assessed frequencies, with the greatest improvements in the high frequencies. The acoustically measured tubing length correction largely accounted for acoustic effects of the participant's earmold.

CONCLUSIONS

IPL-based measurements provide a promising alternative to probe-microphone earmold wRECD procedures. Predicted earmold IPL wRECD is measured without probe-microphone placement, agrees well with earmold SPL wRECDs and is expected to extend the valid bandwidth of wRECD measurement.

Evaluation of deep marginal feedback cancellation for hearing aids using speech and music

Speech and music both play fundamental roles in daily life. Speech is important for communication while music is important for relaxation and social interaction. Both speech and music have a large dynamic range. This does not pose problems for listeners with normal hearing. However, for hearing-impaired listeners, elevated hearing thresholds may result in low-level portions of sound being inaudible. Hearing aids with frequency-dependent amplification and amplitude compression can partly compensate for this problem. However, the gain required for low-level portions of sound to compensate for the hearing loss can be larger than the maximum stable gain of a hearing aid, leading to acoustic feedback. Feedback control is used to avoid such instability, but this can lead to artifacts, especially when the gain is only just below the maximum stable gain. We previously proposed a deep-learning method called DeepMFC for controlling feedback and reducing artifacts and showed that when the sound source was speech DeepMFC performed much better than traditional approaches. However, its performance using music as the sound source was not assessed and the way in which it led to improved performance for speech was not determined. The present paper reveals how DeepMFC addresses feedback problems and evaluates DeepMFC using speech and music as sound sources with both objective and subjective measures. DeepMFC achieved good performance for both speech and music when it was trained with matched training materials. When combined with an adaptive feedback canceller it provided over 13 dB of additional stable gain for hearing-impaired listeners.

A deep learning solution to the marginal stability problems of acoustic feedback systems for hearing aids

For hearing aids, it is critical to reduce the acoustic coupling between the receiver and microphone to ensure that prescribed gains are below the maximum stable gain, thus preventing acoustic feedback. Methods for doing this include fixed and adaptive feedback cancellation, phase modulation, and gain reduction. However, the behavior of hearing aids in situations where the prescribed gain is only just below the maximum stable gain, called here "marginally stable gain," is not well understood. This paper analyzed marginally stable systems and identified three problems, including increased gain at frequencies with the smallest gain margin, short whistles caused by the limited rate of decay of the output when the input drops, and coloration effects. A deep learning framework, called deep marginal feedback cancellation (DeepMFC), was developed to suppress short whistles, and reduce coloration effects, as well as to limit excess amplification at certain frequencies. To implement DeepMFC, many receiver signals in closed-loop systems and corresponding open-loop systems were simulated, and the receiver signals of the closed-loop and open-loop systems were paired together to obtain parallel signals for training. DeepMFC achieved much better performance than existing feedback control methods using objective and subjective measures.

ASR-based speech intelligibility prediction: A review

Various types of methods and approaches are available to predict the intelligibility of speech signals, but many of these still suffer from two major problems: first, their required prior knowledge, which itself could limit the applicability and lower the objectivity of the method, and second, a low generalization capacity, e.g. across noise types, degradation conditions, and speech material. Automatic speech recognition (ASR) has been suggested as a machine-learning-based component of speech intelligibility prediction (SIP), aiming to ameliorate the shortcomings of other SIP methods. Since their first introduction, ASR-based SIP approaches have been developing at an increasingly rapid pace, were deployed in a range of contexts, and have shown promising performance in many scenarios. Our article provides an overview of this body of research. The main differences between competing methods are highlighted and their benefits are explained next to their limitations. We conclude with an outlook on future work and new related directions.

Influence of Audibility and Distortion on Recognition of Reverberant Speech for Children and Adults with Hearing Aid Amplification

BACKGROUND

Adults and children with sensorineural hearing loss (SNHL) have trouble understanding speech in rooms with reverberation when using hearing aid amplification. While the use of amplitude compression signal processing in hearing aids may contribute to this difficulty, there is conflicting evidence on the effects of amplitude compression settings on speech recognition. Less clear is the effect of a fast release time for adults and children with SNHL when using compression ratios derived from a prescriptive procedure.

PURPOSE

The aim of the study is to determine whether release time impacts speech recognition in reverberation for children and adults with SNHL and to determine if these effects of release time and reverberation can be predicted using indices of audibility or temporal and spectral distortion.

RESEARCH DESIGN

This is a quasi-experimental cohort study. Participants used a hearing aid simulator set to the Desired Sensation Level algorithm m[i/o] for three different amplitude compression release times. Reverberation was simulated using three different reverberation times.

PARTICIPANTS

Participants were 20 children and 16 adults with SNHL.

DATA COLLECTION AND ANALYSES

Participants were seated in a sound-attenuating booth and then nonsense syllable recognition was measured. Predictions of speech recognition were made using indices of audibility, temporal distortion, and spectral distortion and the effects of release time and reverberation were analyzed using linear mixed models.

RESULTS

While nonsense syllable recognition decreased in reverberation release time did not significantly affect nonsense syllable recognition. Participants with lower audibility were more susceptible to the negative effect of reverberation on nonsense syllable recognition.

CONCLUSION

We have extended previous work on the effects of reverberation on aided speech recognition to children with SNHL. Variations in release time did not impact the understanding of speech. An index of audibility best predicted nonsense syllable recognition in reverberation and, clinically, these results suggest that patients with less audibility are more susceptible to nonsense syllable recognition in reverberation.

Extended high-frequency audiometry in research and clinical practice

Audiometric testing in research and in clinical settings rarely considers frequencies above 8 kHz. However, the sensitivity of young healthy ears extends to 20 kHz, and there is increasing evidence that testing in the extended high-frequency (EHF) region, above 8 kHz, might provide valuable additional information. Basal (EHF) cochlear regions are especially sensitive to the effects of aging, disease, ototoxic drugs, and possibly noise exposure. Hence, EHF loss may be an early warning of damage, useful for diagnosis and for monitoring hearing health. In certain environments, speech perception may rely on EHF information, and there is evidence for an association between EHF loss and speech perception difficulties, although this may not be causal: EHF loss may instead be a marker for sub-clinical damage at lower frequencies. If there is a causal relation, then amplification in the EHF range may be beneficial if the technical difficulties can be overcome. EHF audiometry in the clinic presents with no particular difficulty, the biggest obstacle being lack of specialist equipment. Currently, EHF audiometry has limited but increasing clinical application. With the development of international guidelines and standards, it is likely that EHF testing will become widespread in future.

OPRA-RS: A Hearing-Aid Fitting Method Based on Automatic Speech Recognition and Random Search

Hearing-aid (HA) prescription rules (such as NAL-NL2, DSL-v5, and CAM2) are used by HA audiologists to define initial HA settings (e.g., insertion gains, IGs) for patients. This initial fitting is later individually adjusted for each patient to improve clinical outcomes in terms of speech intelligibility and listening comfort. During this fine-tuning stage, speech-intelligibility tests are often carried out with the patient to assess the benefits associated with different HA settings. As these tests tend to be time-consuming and performance on them depends on the patient's level of fatigue and familiarity with the test material, only a limited number of HA settings can be explored. Consequently, it is likely that a suboptimal fitting is used for the patient. Recent studies have shown that automatic speech recognition (ASR) can be used to predict the effects of IGs on speech intelligibility for patients with age-related hearing loss (ARHL). The aim of the present study was to extend this approach by optimizing, in addition to IGs, compression thresholds (CTs). However, increasing the number of parameters to be fitted increases exponentially the number of configurations to be assessed. To limit the number of HA settings to be tested, three random-search (RS) genetic algorithms were used. The resulting new HA fitting method, combining ASR and RS, is referred to as "objective prescription rule based on ASR and random search" (OPRA-RS). Optimal HA settings were computed for 12 audiograms, representing average and individual audiometric profiles typical for various levels of ARHL severity, and associated ASR performances were compared to those obtained with the settings recommended by CAM2. Each RS algorithm was run twice to assess its reliability. For all RS algorithms, ASR scores obtained with OPRA-RS were significantly higher than those associated with CAM2. Each RS algorithm converged on similar optimal HA settings across repetitions. However, significant differences were observed between RS algorithms in terms of maximum ASR performance and processing costs. These promising results open the way to the use of ASR and RS algorithms for the fine-tuning of HAs with potential speech-intelligibility benefits for the patient.

Listening to Music Through Hearing Aids: Potential Lessons for Cochlear Implants

Some of the problems experienced by users of hearing aids (HAs) when listening to music are relevant to cochlear implants (CIs). One problem is related to the high peak levels (up to 120 dB SPL) that occur in live music. Some HAs and CIs overload at such levels, because of the limited dynamic range of the microphones and analogue-to-digital converters (ADCs), leading to perceived distortion. Potential solutions are to use 24-bit ADCs or to include an adjustable gain between the microphones and the ADCs. A related problem is how to squeeze the wide dynamic range of music into the limited dynamic range of the user, which can be only 6-20 dB for CI users. In HAs, this is usually done via multi-channel amplitude compression (automatic gain control, AGC). In CIs, a single-channel front-end AGC is applied to the broadband input signal or a control signal derived from a running average of the broadband signal level is used to control the mapping of the channel envelope magnitude to an electrical signal. This introduces several problems: (1) an intense narrowband signal (e.g. a strong bass sound) reduces the level for all frequency components, making some parts of the music harder to hear; (2) the AGC introduces cross-modulation effects that can make a steady sound (e.g. sustained strings or a sung note) appear to fluctuate in level. Potential solutions are to use several frequency channels to create slowly varying gain-control signals and to use slow-acting (or dual time-constant) AGC rather than fast-acting AGC.

Detection, Speech Recognition, Loudness, and Preference Outcomes With a Direct Drive Hearing Aid: Effects of Bandwidth

Direct drive hearing devices, which deliver a signal directly to the middle ear by vibrating the tympanic membrane via a lens placed in contact with the umbo, are designed to provide an extension of audible bandwidth, but there are few studies of the effects of these devices on preference, speech intelligibility, and loudness. The current study is the first to compare aided speech understanding between narrow and extended bandwidth conditions for listeners with hearing loss while fitted with a direct drive hearing aid system. The study also explored the effect of bandwidth on loudness perception and investigated subjective preference for bandwidth. Fifteen adult hearing aid users with symmetrical sensorineural hearing loss participated in a prospective, within-subjects, randomized single-blind repeated-measures study. Participants wore the direct drive hearing aids for 4 to 15 weeks (average 6 weeks) prior to outcome measurement. Outcome measures were completed in various bandwidth conditions achieved by reducing the gain of the device above 5000 Hz or by filtering the stimuli. Aided detection thresholds provided evidence of amplification to 10000 Hz. A significant improvement was found in high-frequency consonant detection and recognition, as well as for speech in noise performance in the full versus narrow bandwidth conditions. Subjective loudness ratings increased with provision of the full bandwidth available; however, real-world trials showed most participants were able to wear the full bandwidth hearing aids with only small adjustments to the prescription method. The majority of participants had either no preference or a preference for the full bandwidth setting.

Compression and amplification algorithms in hearing aids impair the selectivity of neural responses to speech

In quiet environments, hearing aids improve the perception of low-intensity sounds. However, for high-intensity sounds in background noise, the aids often fail to provide a benefit to the wearer. Here, by using large-scale single-neuron recordings from hearing-impaired gerbils — an established animal model of human hearing — we show that hearing aids restore the sensitivity of neural responses to speech, but not their selectivity. Rather than reflecting a deficit in supra-threshold auditory processing, the low selectivity is a consequence of hearing-aid compression (which decreases the spectral and temporal contrasts of incoming sound) and of amplification (which distorts neural responses, regardless of whether hearing is impaired). Processing strategies that avoid the trade-off between neural sensitivity and selectivity should improve the performance of hearing aids.

Perceived Sound Quality Dimensions Influencing Frequency-Gain Shaping Preferences for Hearing Aid-Amplified Speech and Music

Hearing aids are typically fitted using speech-based prescriptive formulae to make speech more intelligible. Individual preferences may vary from these prescriptions and may also vary with signal type. It is important to consider what motivates listener preferences and how those preferences can inform hearing aid processing so that assistive listening devices can best be tailored for hearing aid users. Therefore, this study explored preferred frequency-gain shaping relative to prescribed gain for speech and music samples. Preferred gain was determined for 22 listeners with mild sloping to moderately severe hearing loss relative to individually prescribed amplification while listening to samples of male speech, female speech, pop music, and classical music across low-, mid-, and high-frequency bands. Samples were amplified using a fast-acting compression hearing aid simulator. Preferences were determined using an adaptive paired comparison procedure. Listeners then rated speech and music samples processed using prescribed and preferred shaping across different sound quality descriptors. On average, low-frequency gain was significantly increased relative to the prescription for all stimuli and most substantially for pop and classical music. High-frequency gain was decreased significantly for pop music and male speech. Gain adjustments, particularly in the mid- and high-frequency bands, varied considerably between listeners. Music preferences were driven by changes in perceived fullness and sharpness, whereas speech preferences were driven by changes in perceived intelligibility and loudness. The results generally support the use of prescribed amplification to optimize speech intelligibility and alternative amplification for music listening for most listeners.

DARF: A data-reduced FADE version for simulations of speech recognition thresholds with real hearing aids

Developing and selecting hearing aids is a time consuming process which is simplified by using objective models. Previously, the framework for auditory discrimination experiments (FADE) accurately simulated benefits of hearing aid algorithms with root mean squared prediction errors below 3 dB. One FADE simulation requires several hours of (un)processed signals, which is obstructive when the signals have to be recorded. We propose and evaluate a data-reduced FADE version (DARF) which facilitates simulations with signals that cannot be processed digitally, but that can only be recorded in real-time. DARF simulates one speech recognition threshold (SRT) with about 30 min of recorded and processed signals of the (German) matrix sentence test. Benchmark experiments were carried out to compare DARF and standard FADE exhibiting small differences for stationary maskers (1 dB), but larger differences with strongly fluctuating maskers (5 dB). Hearing impairment and hearing aid algorithms seemed to reduce the differences. Hearing aid benefits were simulated in terms of speech recognition with three pairs of real hearing aids in silence (≥8 dB), in stationary and fluctuating maskers in co-located (stat. 2 dB; fluct. 6 dB), and spatially separated speech and noise signals (stat. ≥8 dB; fluct. 8 dB). The simulations were plausible in comparison to data from literature, but a comparison with empirical data is still open. DARF facilitates objective SRT simulations with real devices with unknown signal processing in real environments. Yet, a validation of DARF for devices with unknown signal processing is still pending since it was only tested with three similar devices. Nonetheless, DARF could be used for improving as well as for developing or model-based fitting of hearing aids.

Sound Quality Ratings of Amplified Speech and Music Using a Direct Drive Hearing Aid: Effects of Bandwidth

OBJECTIVE

To determine sound quality for extended bandwidth amplification using a direct drive hearing device.

STUDY DESIGN

Prospective double-blind within-subjects repeated measures study.

SETTING

University hearing research laboratories.

PATIENTS

Fifteen experienced hearing aid users with symmetric mild-sloping-to-severe sensorineural hearing loss.

INTERVENTIONS

Sound quality ratings of speech and music passages were obtained using the Multiple Stimulus with Hidden References and Anchors (MUSHRA) protocol after wearing a direct drive hearing aid for at least 4 weeks. Passages were processed to filter out low-frequency (below 123 and 313 Hz) and high-frequency (above 4455, 5583, 6987, and 10,869 Hz) energy.

MAIN OUTCOME MEASURES

Comparison of sound quality ratings for speech and music between low and high-pass filter frequencies measured from 0 to 100, where 0 represents "bad" and 100 represents "excellent."

RESULTS

Wider bandwidth stimuli received higher sound quality ratings compared with narrower bandwidth stimuli. Conditions with more low-frequency energy (full-band and 123 Hz cut-off) were rated as having higher sound quality. More low-frequency energy in the 123 Hz condition was rated as having higher sound versus the 313 Hz condition (mean difference: 11.2%, p = 0.001). Full-band conditions with more low- and high-frequency energy were higher than the other high-frequency cutoff conditions (mean difference range: 12.9-15%, p < 0.001).

CONCLUSIONS

The direct drive system provides higher sound quality of both speech and music compared to narrowband conditions. Sound quality improvements were mainly attributable to low-frequency sound, but stimuli with specific high-frequency content were rated with higher sound quality when additional high-frequency energy was present.

Fit-to-Targets and Aided Speech Intelligibility Index Values for Hearing Aids Fitted to the DSL v5-Adult Prescription

BACKGROUND

 Matching hearing aid output levels to prescribed targets is a component of preferred practice, yet recent normative data on appropriateness of fittings are lacking. Verification measures that assess closeness of fit-to-target include raw deviations from target, root-mean-squared-error (RMSE) deviations from target, and aided Speech Intelligibility Index (SII) values. Establishing normative ranges for these measures may help hearing professionals determine whether a patient's fit-to-targets and/or aided speech audibility is typical for his or her degree of hearing loss.

PURPOSE

 This article aims to characterize the range of fit-to-target and the range of aided SII associated with hearing aid fittings using the Desired Sensation Level version 5.0 (DSL v5-adult) prescription with adults, considering also hearing aid style, venting, and audiometric characteristics.

RESEARCH DESIGN

 A descriptive and correlational study of data collected from a retrospective chart review.

RESULTS

 Hearing aid fittings to 281 ears were compiled. The four-frequency average deviation from target (RMSE) was within ± 5 dB of target in 77% of fittings for mid-level speech. Deviation from targets increased with hearing loss, particularly when the loss is greater than 85 dB hearing level or if the loss was steeply sloping. Venting increased the deviation from targets in the low frequencies. Aided SII values strongly correlated with the participants' hearing thresholds. Clinical ranges for RMSE and aided SII were developed for characterization of fitting outcomes.

CONCLUSION

 Fitting to DSL v5-adult targets was observed within ± 5 dB absolute deviation, or within 5 dB RMSE, on average for typical adult hearing aid fittings. Confidence intervals for deviation from target and aided SII are proposed.

Application of Bayesian Active Learning to the Estimation of Auditory Filter Shapes Using the Notched-Noise Method

Time-efficient hearing tests are important in both clinical practice and research studies. This particularly applies to notched-noise tests, which are rarely done in clinical practice because of the time required. Auditory-filter shapes derived from notched-noise data may be useful for diagnosis of the cause of hearing loss and for fitting of hearing aids, especially if measured over a wide range of center frequencies. To reduce the testing time, we applied Bayesian active learning (BAL) to the notched-noise test, picking the most informative stimulus parameters for each trial based on nine Gaussian Processes. A total of 11 hearing-impaired subjects were tested. In 20 to 30 min, the test provided estimates of signal threshold as a continuous function of frequency from 500 to 4000 Hz for nine notch widths and for notches placed both symmetrically and asymmetrically around the signal frequency. The thresholds were found to be consistent with those obtained using a 2-up/1-down forced-choice procedure at a single center frequency. In particular, differences in threshold between the methods did not vary with notch width. An independent second run of the BAL test for one notch width showed that it is reliable. The data derived from the BAL test were used to estimate auditory-filter width and asymmetry and detection efficiency for center frequencies from 500 to 4000 Hz. The results agreed with expectations for cochlear hearing losses that were derived from the audiogram and a hearing model.

Improving hearing-aid gains based on automatic speech recognition

This study provides proof of concept that automatic speech recognition (ASR) can be used to improve hearing aid (HA) fitting. A signal-processing chain consisting of a HA simulator, a hearing-loss simulator, and an ASR system normalizing the intensity of input signals was used to find HA-gain functions yielding the highest ASR intelligibility scores for individual audiometric profiles of 24 listeners with age-related hearing loss. Significantly higher aided speech intelligibility scores and subjective ratings of speech pleasantness were observed when the participants were fitted with ASR-established gains than when fitted with the gains recommended by the CAM2 fitting rule.

Speech recognition, loudness, and preference with extended bandwidth hearing aids for adult hearing aid users

Objective: In contrast to the past, some current hearing aids can provide gain for frequencies above 4-5 kHz. This study assessed the effect of wider bandwidth on outcome measures using hearing aids fitted with the DSL v5.0 prescription.Design: There were two conditions: an extended bandwidth condition, for which the maximum available bandwidth was provided, and a restricted bandwidth condition, in which gain was reduced for frequencies above 4.5 kHz. Outcome measures were assessed in both conditions.Study sample: Twenty-four participants with mild-to-moderately-severe sensorineural high-frequency sloping hearing loss.Results: Providing extended bandwidth resulted in maximum audible output frequency values of 7.5 kHz on average for an input level of 65 dB SPL. An improvement in consonant discrimination scores (4.1%), attributable to better perception of /s/, /z/, and /t/ phonemes, was found in the extended bandwidth condition, but no significant change in loudness perception or preferred listening levels was found. Most listeners (79%) had either no preference (33%) or some preference for the extended bandwidth condition (46%).Conclusions: The results suggest that providing the maximum bandwidth available with modern hearing aids fitted with DSL v5.0, using targets from 0.25 to 8 kHz, can be beneficial for the tested population.

Effect of the number of amplitude-compression channels and compression speed on speech recognition by listeners with mild to moderate sensorineural hearing loss

The use of a large number of amplitude-compression channels in hearing aids has potential advantages, such as the ability to compensate for variations in loudness recruitment across frequency and provide appropriate frequency-response shaping. However, sound quality and speech intelligibility could be adversely affected due to reduction of spectro-temporal contrast and distortion, especially when fast-acting compression is used. This study assessed the effect of the number of channels and compression speed on speech recognition when the multichannel processing was used solely to implement amplitude compression, and not for frequency-response shaping. Computer-simulated hearing aids were used. The frequency-dependent insertion gains for speech with a level of 65 dB sound pressure level were applied using a single filter before the signal was filtered into compression channels. Fast-acting (attack, 10 ms; release, 100 ms) or slow-acting (attack, 50 ms; release, 3000 ms) compression using 3, 6, 12, and 22 channels was applied subsequently. Using a sentence recognition task with speech in two- and eight-talker babble at three different signal-to-babble ratios (SBRs), 20 adults with sensorineural hearing loss were tested. The number of channels and compression speed had no significant effect on speech recognition, regardless of babble type or SBR.

Achieved Gain and Subjective Outcomes for a Wide-Bandwidth Contact Hearing Aid Fitted Using CAM2

Supplemental Digital Content is available in the text.

Objectives:

The objective of this study was to test the ability to achieve, maintain, and subjectively benefit from extended high-frequency amplification in a real-world use scenario, with a device that restores audibility for frequencies up to 10 kHz.

Design:

A total of 78 participants (149 ears) with mild to moderately-severe sensorineural hearing loss completed one of two studies conducted across eight clinical sites. Participants were fitted with a light-driven contact hearing aid (the Earlens system) that directly drives the tympanic membrane, allowing extended high-frequency output and amplification with minimal acoustic feedback. Cambridge Method for Loudness Equalization 2 - High Frequency (CAM2)-prescribed gains for experienced users were used for initial fitting, and adjustments were made when required according to participant preferences for loudness and comfort or when measures of functional gain (FG) indicated that more or less gain was needed. Participants wore the devices for an extended period. Prescribed versus adjusted output and gain, frequency-specific FG, and self-perceived benefit assessed with the Abbreviated Profile of Hearing Aid Benefit, and a custom questionnaire were documented. Self-perceived benefit results were compared with those for unaided listening and to ratings with participants’ own acoustic hearing aids.

Results:

The prescribed low-level insertion gain from 6 to 10 kHz averaged 53 dB across all ears, with a range from 26 to 86 dB. After adjustment, the gain from 6 to 10 kHz decreased to an average of 45 dB with a range from 16 to 86 dB. Measured FG averaged 39 dB from 6 to 10 kHz with a range from 11 to 62 dB. Abbreviated Profile of Hearing Aid Benefit results revealed a significant improvement in communication relative to unaided listening, averaging 28 to 32 percentage points for the background noise, reverberation, and ease of communication subscales. Relative to participants’ own hearing aids, the subscales ease of communication and aversiveness showed small but significant improvements for Earlens ranging from 6 to 7 percentage points. For the custom satisfaction questionnaire, most participants rated the Earlens system as better than their own hearing aids in most situations.

Conclusions:

Participants used and reported subjective benefit from the Earlens system. Most participants preferred slightly less gain at 6 to 10 kHz than prescribed for experienced users by CAM2, preferring similar gains to those prescribed for inexperienced users, but gains over the extended high frequencies were high relative to those that are currently available with acoustic hearing aids.

Listener Factors Explain Little Variability in Self-Adjusted Hearing Aid Gain

Self-adjustment of hearing aid gain can provide valuable information about the gain preferences of individual listeners, but these preferences are not well understood. Listeners with mild-to-moderate hearing loss used self-adjustment to select amplification gain and compression parameters in real time on a portable touch screen device while listening in quiet and noisy backgrounds. Adjustments to gain prescribed by the National Acoustics Laboratories' non-linear fitting procedure (NAL-NL2) showed large between-subject variability. Known listener characteristics (age, gender, hearing thresholds, hearing aid experience, acceptable noise level, and external ear characteristics) and listener engagement with the self-adjustment software were examined as potential predictors of this variability. Neither listener characteristics nor time spent adjusting gain were robust predictors of gain change from NAL-NL2. Listeners with less than 2 years of hearing aid experience and who also had better hearing thresholds tended to select less gain, relative to NAL-NL2, than experienced hearing aid users who had poorer thresholds. Listener factors explained no more than 10% of the between-subject variance in deviation from NAL-NL2, suggesting that modifying prescriptive fitting formulae based on the factors examined here would be unlikely to result in amplification parameters that are similar to user-customized settings. Self-adjustment typically took less than 3 min, indicating that listeners could use comparable technology without a substantial time commitment.

Comparison of Frequency Transposition and Frequency Compression for People With Extensive Dead Regions in the Cochlea

The objective was to determine the effects of two frequency-lowering algorithms (frequency transposition, FT, and frequency compression, FC) on audibility, speech identification, and subjective benefit, for people with high-frequency hearing loss and extensive dead regions (DRs) in the cochlea. A single-blind randomized crossover design was used. FT and FC were compared with each other and with a control condition (denoted ‘Control’) without frequency lowering, using hearing aids that were otherwise identical. Data were collected after at least 6 weeks of experience with a condition. Outcome measures were audibility, scores for consonant identification, scores for word-final /s, z/ detection (S test), sentence-in-noise intelligibility, and a questionnaire assessing self-perceived benefit (Spatial and Qualities of Hearing Scale). Ten adults with steeply sloping high-frequency hearing loss and extensive DRs were tested. FT and FC improved the audibility of some high-frequency sounds for 7 and 9 participants out of 10, respectively. At the group level, performance for FT and FC did not differ significantly from that for Control for any of the outcome measures. However, the pattern of consonant confusions varied across conditions. Bayesian analysis of the confusion matrices revealed a trend for FT to lead to more consistent error patterns than FC and Control. Thus, FT may have the potential to give greater benefit than Control or FC following extended experience or training.

Perceptual Effects of Adjusting Hearing-Aid Gain by Means of a Machine-Learning Approach Based on Individual User Preference

This study investigated a method to adjust hearing-aid gain by use of a machine-learning algorithm that estimates the optimal setting of gain parameters based on user preference indicated in an iterative paired-comparison procedure. Twenty hearing-impaired participants completed this procedure for 12 different sound scenarios. During the adjustment procedure, their task was to indicate a preference based on one of three sound attributes: Basic Audio Quality, Listening Comfort, or Speech Clarity. In a double-blind comparison of recordings of the processed scenarios, and using the same attributes as criteria, the adjusted gain settings were subsequently compared with two prescribed settings of the same hearing aid (with and without activation of an automatic sound-classification system). The results showed that the adjustment method provided a general improvement of Basic Audio Quality, an improvement of Listening Comfort in a traffic-noise scenario but not in three scenarios with speech babble, and no significant improvement of Speech Clarity. A large variation in gain adjustments was observed across participants, both among those who did benefit and among those who did not benefit from the adjustment. There was no clear connection between the gain adjustments and the perceived benefit, which indicates that the preferred gain settings for a given sound scenario and a given listening intention are highly individual and difficult to predict.

The Influence of Hearing Aid Gain on Gap-Detection Thresholds for Children and Adults With Hearing Loss

OBJECTIVES

The objective of this experiment was to examine the contributions of audibility to the ability to perceive a gap in noise for children and adults. Sensorineural hearing loss (SNHL) in adulthood is associated with a deficit in gap detection. It is well known that reduced audibility in adult listeners with SNHL contributes to this deficit; however, it is unclear the extent to which hearing aid amplification can restore gap-detection thresholds, and the effect of childhood SNHL on gap-detection thresholds have not been described. For adults, it was hypothesized that restoring the dynamic range of hearing for listeners with SNHL would lead to approximately normal gap-detection thresholds. Children with normal hearing (NH) exhibit poorer gap-detection thresholds than adults. Because of their hearing loss, children with SNHL have less auditory experience than their peers with NH. Yet, it is unknown the extent to which auditory experience impacts their ability to perceive gaps in noise. Even with the provision of amplification, it was hypothesized that children with SNHL would show a deficit in gap detection, relative to their peers with normal hearing, because of reduced auditory experience.

DESIGN

The ability to detect a silent interval in noise was tested by adapting the stimulus level required for detection of gap durations between 3 and 20 ms for adults and children with and without SNHL. Stimulus-level thresholds were measured for participants with SNHL without amplification and with two prescriptive procedures-the adult and child versions of the desired sensation level i/o program-using a hearing aid simulator. The child version better restored the normal dynamic range than the adult version. Adults and children with NH were tested without amplification.

RESULTS

When fitted using the procedure that best restored the dynamic range, adults with SNHL had stimulus-level thresholds similar to those of adults with normal hearing. Compared to the children with NH, the children with SNHL required a higher stimulus level to detect a 5-ms gap, despite having used the procedure that better restored the normal dynamic range of hearing. Otherwise, the two groups of children had similar stimulus-level thresholds.

CONCLUSION

These findings suggest that apparent deficits in temporal resolution, as measured using stimulus-level thresholds for the detection of gaps, are dependent on age and audibility. These novel results indicate that childhood SNHL may impair temporal resolution as measured by stimulus-level thresholds for the detection of a gap in noise. This work has implications for understanding the effects of amplification on the ability to perceive temporal cues in speech.

Predictors of Hearing-Aid Outcomes

Over 360 million people worldwide suffer from disabling hearing loss. Most of them can be treated with hearing aids. Unfortunately, performance with hearing aids and the benefit obtained from using them vary widely across users. Here, we investigate the reasons for such variability. Sixty-eight hearing-aid users or candidates were fitted bilaterally with nonlinear hearing aids using standard procedures. Treatment outcome was assessed by measuring aided speech intelligibility in a time-reversed two-talker background and self-reported improvement in hearing ability. Statistical predictive models of these outcomes were obtained using linear combinations of 19 predictors, including demographic and audiological data, indicators of cochlear mechanical dysfunction and auditory temporal processing skills, hearing-aid settings, working memory capacity, and pretreatment self-perceived hearing ability. Aided intelligibility tended to be better for younger hearing-aid users with good unaided intelligibility in quiet and with good temporal processing abilities. Intelligibility tended to improve by increasing amplification for low-intensity sounds and by using more linear amplification for high-intensity sounds. Self-reported improvement in hearing ability was hard to predict but tended to be smaller for users with better working memory capacity. Indicators of cochlear mechanical dysfunction, alone or in combination with hearing settings, did not affect outcome predictions. The results may be useful for improving hearing aids and setting patients’ expectations.

A Set of Time-and-Frequency-Localized Short-Duration Speech-Like Stimuli for Assessing Hearing-Aid Performance via Cortical Auditory-Evoked Potentials

Short-duration speech-like stimuli, for example, excised from running speech, can be used in the clinical setting to assess the integrity of the human auditory pathway at the level of the cortex. Modeling of the cochlear response to these stimuli demonstrated an imprecision in the location of the spectrotemporal energy, giving rise to uncertainty as to what and when of a stimulus caused any evoked electrophysiological response. This article reports the development and assessment of four short-duration, limited-bandwidth stimuli centered at low, mid, mid-high, and high frequencies, suitable for free-field delivery and, in addition, reproduction via hearing aids. The durations were determined by the British Society of Audiology recommended procedure for measuring Cortical Auditory-Evoked Potentials. The levels and bandwidths were chosen via a computational model to produce uniform cochlear excitation over a width exceeding that likely in a worst-case hearing-impaired listener. These parameters produce robustness against errors in insertion gains, and variation in frequency responses, due to transducer imperfections, room modes, and age-related variation in meatal resonances. The parameter choice predicts large spectral separation between adjacent stimuli on the cochlea. Analysis of the signals processed by examples of recent digital hearing aids mostly show similar levels of gain applied to each stimulus, independent of whether the stimulus was presented in isolation, bursts, continuous, or embedded in continuous speech. These stimuli seem to be suitable for measuring hearing-aided Cortical Auditory-Evoked Potentials and have the potential to be of benefit in the clinical setting.

Design of Compensated Multi-Channel Dynamic-Range Compressor for Hearing Aid Devices using Polyphase Implementation

Dynamic-range compression (DRC) is widely used in hearing aid devices (HADs) to reduce the wide dynamic range of input speech signal to match the residual dynamic range of people with hearing loss. Most compression systems use multi-channel compression to provide more effective and accurate solutions to match input signal with hearing-impaired people's audiogram. However, multi-channel compression introduces distortion to the system, and increases computational complexity. It limits the sampling rate and results in systems latency, hence, introduces difficulty realizing it in real-time. In this paper, a compensation filter is proposed to reduce the distortion, and polyphase implementation is applied to reduce the computational complexity. Objective and subjective tests are conducted to evaluate the quality and intelligibility of the output audio (speech) signal under different noise types and signal to noise ratios (SNRs).

Evaluation of a multi-channel algorithm for reducing transient sounds

OBJECTIVE

The objective was to evaluate and select appropriate parameters for a multi-channel transient reduction (MCTR) algorithm for detecting and attenuating transient sounds in speech.

DESIGN

In each trial, the same sentence was played twice. A transient sound was presented in both sentences, but its level varied across the two depending on whether or not it had been processed by the MCTR and on the "strength" of the processing. The participant indicated their preference for which one was better and by how much in terms of the balance between the annoyance produced by the transient and the audibility of the transient (they were told that the transient should still be audible).

STUDY SAMPLE

Twenty English-speaking participants were tested, 10 with normal hearing and 10 with mild-to-moderate hearing-impairment. Frequency-dependent linear amplification was provided for the latter.

RESULTS

The results for both participant groups indicated that sounds processed using the MCTR were preferred over the unprocessed sounds. For the hearing-impaired participants, the medium and strong settings of the MCTR were preferred over the weak setting.

CONCLUSIONS

The medium and strong settings of the MCTR reduced the annoyance produced by the transients while maintaining their audibility.

Survey of Current Practice in the Fitting and Fine-Tuning of Common Signal-Processing Features in Hearing Aids for Adults

BACKGROUND

Current guidelines for adult hearing aid fittings recommend the use of a prescriptive fitting rationale with real-ear verification that considers the audiogram for the determination of frequency-specific gain and ratios for wide dynamic range compression. However, the guidelines lack recommendations for how other common signal-processing features (e.g., noise reduction, frequency lowering, directional microphones) should be considered during the provision of hearing aid fittings and fine-tunings for adult patients.

PURPOSE

The purpose of this survey was to identify how audiologists make clinical decisions regarding common signal-processing features for hearing aid provision in adults.

RESEARCH DESIGN

An online survey was sent to audiologists across the United States. The 22 survey questions addressed four primary topics including demographics of the responding audiologists, factors affecting selection of hearing aid devices, the approaches used in the fitting of signal-processing features, and the strategies used in the fine-tuning of these features.

STUDY SAMPLE

A total of 251 audiologists who provide hearing aid fittings to adults completed the electronically distributed survey. The respondents worked in a variety of settings including private practice, physician offices, university clinics, and hospitals/medical centers.

DATA COLLECTION AND ANALYSIS

Data analysis was based on a qualitative analysis of the question responses. The survey results for each of the four topic areas (demographics, device selection, hearing aid fitting, and hearing aid fine-tuning) are summarized descriptively.

RESULTS

Survey responses indicate that audiologists vary in the procedures they use in fitting and fine-tuning based on the specific feature, such that the approaches used for the fitting of frequency-specific gain differ from other types of features (i.e., compression time constants, frequency lowering parameters, noise reduction strength, directional microphones, feedback management). Audiologists commonly rely on prescriptive fitting formulas and probe microphone measures for the fitting of frequency-specific gain and rely on manufacturers' default settings and recommendations for both the initial fitting and the fine-tuning of signal-processing features other than frequency-specific gain.

CONCLUSIONS

The survey results are consistent with a lack of published protocols and guidelines for fitting and adjusting signal-processing features beyond frequency-specific gain. To streamline current practice, a transparent evidence-based tool that enables clinicians to prescribe the setting of other features from individual patient characteristics would be desirable.

Effects of Modified Hearing Aid Fittings on Loudness and Tone Quality for Different Acoustic Scenes

OBJECTIVE

To compare loudness and tone-quality ratings for sounds processed via a simulated five-channel compression hearing aid fitted using NAL-NL2 or using a modification of the fitting designed to be appropriate for the type of listening situation: speech in quiet, speech in noise, music, and noise alone.

DESIGN

Ratings of loudness and tone quality were obtained for stimuli presented via a loudspeaker in front of the participant. For normal-hearing participants, levels of 50, 65, and 80 dB SPL were used. For hearing-impaired participants, the stimuli were processed via a simulated hearing aid with five-channel fast-acting compression fitted using NAL-NL2 or using a modified fitting. Input levels to the simulated hearing aid were 50, 65, and 80 dB SPL. All participants listened with one ear plugged. For speech in quiet, the modified fitting was based on the CAM2B method. For speech in noise, the modified fitting used slightly (0 to 2 dB) decreased gains at low frequencies. For music, the modified fitting used increased gains (by 5 to 14 dB) at low frequencies. For noise alone, the modified fitting used decreased gains at all frequencies (by a mean of 1 dB at low frequencies increasing to 8 dB at high frequencies).

RESULTS

For speech in quiet, ratings of loudness with the NAL-NL2 fitting were slightly lower than the mean ratings for normal-hearing participants for all levels, while ratings with CAM2B were close to normal for the two lower levels, and slightly greater than normal for the highest level. Ratings of tone quality were close to the optimum value ("just right") for both fittings, except that the CAM2B fitting was rated as very slightly boomy for the 80-dB SPL level. For speech in noise, the ratings of loudness were very close to the normal values and the ratings of tone quality were close to the optimal value for both fittings and for all levels. For music, the ratings of loudness were close to the normal values for NAL-NL2 and slightly above normal for the modified fitting. The tone quality was rated as very slightly tinny for NAL-NL2 and very slightly boomy for the modified fitting. For noise alone, the NAL-NL2 fitting was rated as slightly louder than normal for all levels, while the modified fitting was rated as close to normal. Tone quality was rated as slightly sharper for the NAL-NL2 fitting than for the modified fitting.

CONCLUSIONS

Loudness and tone quality can sometimes be made slightly closer to "normal" by modifying gains for different listening situations. The modification for music required to achieve "normal" tone quality appears to be less than used in this study.

Light-Driven Contact Hearing Aid for Broad-Spectrum Amplification: Safety and Effectiveness Pivotal Study

Objective:

Demonstrate safety and effectiveness of the light-driven contact hearing aid to support FDA clearance.

Study Design:

A single-arm, open-label investigational-device clinical trial.

Setting:

Two private-practice and one hospital-based ENT clinics.

Patients:

Forty-three subjects (86 ears) with mild-to-severe bilateral sensorineural hearing impairment.

Intervention:

Bilateral amplification delivered via a light-driven contact hearing aid comprising a Tympanic Lens (Lens) with a customized platform to directly drive the umbo and a behind-the-ear sound processor (Processor) that encodes sound into light pulses to wirelessly deliver signal and power to the Lens.

Main Outcome Measures:

The primary safety endpoint was a determination of “no change” (PTA4 < 10 dB) in residual unaided hearing at the 120-day measurement interval. The primary efficacy endpoint was improvement in word recognition using NU-6 at the 30-day measurement interval over the baseline unaided case. Secondary efficacy endpoints included functional gain from 2 to 10 kHz and speech-in-noise improvement over the baseline unaided case using both omnidirectional and directional microphones.

Results:

The results for the 86 ears in the study determined a mean change of −0.40 dB in PTA4, indicating no change in residual hearing (p < 0.0001). There were no serious device- or procedure-related adverse events, or unanticipated adverse events. Word recognition aided with the Earlens improved significantly (p < 0.0001) over the unaided performance, by 35% rationalized arcsine units on average. Mean functional gain was 31 dB across 2 to 10 kHz. The average speech-recognition threshold improvement over the unaided case for the Hearing in Noise Test was 0.75 dB (p = 0.028) and 3.14 dB (p < 0.0001) for the omnidirectional and directional microphone modes, respectively.

Conclusion:

The safety and effectiveness data supported a de novo 510(k) submission that received clearance from the FDA.

Quality ratings of frequency-compressed speech by participants with extensive high-frequency dead regions in the cochlea

OBJECTIVE

The objective was to assess the degradation of speech sound quality produced by frequency compression for listeners with extensive high-frequency dead regions (DRs).

DESIGN

Quality ratings were obtained using values of the starting frequency (Sf) of the frequency compression both below and above the estimated edge frequency, fe, of each DR. Thus, the value of Sf often fell below the lowest value currently used in clinical practice. Several compression ratios were used for each value of Sf. Stimuli were sentences processed via a prototype hearing aid based on Phonak Exélia Art P.

STUDY SAMPLE

Five participants (eight ears) with extensive high-frequency DRs were tested.

RESULTS

Reductions of sound-quality produced by frequency compression were small to moderate. Ratings decreased significantly with decreasing Sf and increasing CR. The mean ratings were lowest for the lowest Sf and highest CR. Ratings varied across participants, with one participant rating frequency compression lower than no frequency compression even when Sf was above fe.

CONCLUSIONS

Frequency compression degraded sound quality somewhat for this small group of participants with extensive high-frequency DRs. The degradation was greater for lower values of Sf relative to fe, and for greater values of CR. Results varied across participants.

Evaluation of a Frequency-Lowering Algorithm for Adults With High-Frequency Hearing Loss

The objective was to determine the effects of a frequency-lowering algorithm (frequency composition, Fcomp) on consonant identification, word-final /s, z/ detection, the intelligibility of sentences in noise, and subjective benefit, for people with high-frequency hearing loss, including people with dead regions (DRs) in the cochlea. A single-blind randomized crossover design was used. Performance with Bernafon Acriva 9 hearing aids was compared with Fcomp off and Fcomp on. Participants wore the hearing aids in each condition in a counterbalanced order. Data were collected after at least 8 weeks of experience with a condition. Outcome measures were audibility, scores from the speech perception tests, and scores from a questionnaire comparing self-perceived hearing ability with Fcomp off and Fcomp on. Ten adults with mild to severe high-frequency hearing loss (seven with extensive DRs, one with patchy or restricted DRs, and two with no DR) were tested. Fcomp improved the audibility of high-frequency sounds for 6 out of 10 participants. There was no overall effect of Fcomp on consonant identification, but the pattern of consonant confusions varied across conditions and participants. For word-final /s, z/ detection, performance was significantly better with Fcomp on than with Fcomp off. Questionnaire scores showed no differences between conditions. In summary, Fcomp improved word-final /s, z/ detection. No benefit was found for the other measures.

Evaluation of a method for enhancing interaural level differences at low frequencies

A method (called binaural enhancement) for enhancing interaural level differences at low frequencies, based on estimates of interaural time differences, was developed and evaluated. Five conditions were compared, all using simulated hearing-aid processing: (1) Linear amplification with frequency-response shaping; (2) binaural enhancement combined with linear amplification and frequency-response shaping; (3) slow-acting four-channel amplitude compression with independent compression at the two ears (AGC4CH); (4) binaural enhancement combined with four-channel compression (BE-AGC4CH); and (5) four-channel compression but with the compression gains synchronized across ears. Ten hearing-impaired listeners were tested, and gains and compression ratios for each listener were set to match targets prescribed by the CAM2 fitting method. Stimuli were presented via headphones, using virtualization methods to simulate listening in a moderately reverberant room. The intelligibility of speech at ±60° azimuth in the presence of competing speech on the opposite side of the head at ±60° azimuth was not affected by the binaural enhancement processing. Sound localization was significantly better for condition BE-AGC4CH than for condition AGC4CH for a sentence, but not for broadband noise, lowpass noise, or lowpass amplitude-modulated noise. The results suggest that the binaural enhancement processing can improve localization for sounds with distinct envelope fluctuations.

Preferred Compression Speed for Speech and Music and Its Relationship to Sensitivity to Temporal Fine Structure

Multichannel amplitude compression is widely used in hearing aids. The preferred compression speed varies across individuals. Moore (2008) suggested that reduced sensitivity to temporal fine structure (TFS) may be associated with preference for slow compression. This idea was tested using a simulated hearing aid. It was also assessed whether preferences for compression speed depend on the type of stimulus: speech or music. Twenty-two hearing-impaired subjects were tested, and the stimulated hearing aid was fitted individually using the CAM2A method. On each trial, a given segment of speech or music was presented twice. One segment was processed with fast compression and the other with slow compression, and the order was balanced across trials. The subject indicated which segment was preferred and by how much. On average, slow compression was preferred over fast compression, more so for music, but there were distinct individual differences, which were highly correlated for speech and music. Sensitivity to TFS was assessed using the difference limen for frequency at 2000 Hz and by two measures of sensitivity to interaural phase at low frequencies. The results for the difference limens for frequency, but not the measures of sensitivity to interaural phase, supported the suggestion that preference for compression speed is affected by sensitivity to TFS.

Better-ear glimpsing at low frequencies in normal-hearing and hearing-impaired listeners

Better-ear glimpsing is an auditory process that takes advantage of short-term interaural level differences (ILDs) to improve the understanding of speech in spatial fluctuating noise. Since ILDs are mainly present at high frequencies, where most hearing-impaired (HI) listeners have the strongest hearing loss, HI individuals cannot fully utilize ILDs for better-ear glimpsing, which may lead to poorer understanding of speech in noise. This problem may be alleviated by hearing aids that artificially generate ILDs at low frequencies where hearing is typically less impaired. The present study therefore investigated the spatial benefit in speech intelligibility that is provided by better-ear glimpsing with low-frequency extended ILDs in a symmetric two-distractor speech background. Speech reception thresholds were measured in a spatially co-located and separated condition as a function of frequency region in ten normal-hearing (NH) and ten mild-to-moderate sensorineural HI subjects. In both groups the extended ILDs provided a substantial spatial advantage on top of the advantage already provided by natural ILDs. Moreover, the spatial advantage was largely independent of frequency region, suggesting that both NH and HI subjects can utilize low-frequency ILDs for improving speech understanding in noise. Overall performance as well as spatial advantage was reduced in the HI group.

Extended High-Frequency Bandwidth Improves Speech Reception in the Presence of Spatially Separated Masking Speech

OBJECTIVES

The hypothesis that extending the audible frequency bandwidth beyond the range currently implemented in most hearing aids can improve speech understanding was tested for normal-hearing and hearing-impaired participants using target sentences and spatially separated masking speech.

DESIGN

The Hearing In Speech Test (HIST) speech corpus was re-recorded, and four masking talkers were recorded at a sample rate of 44.1 kHz. All talkers were male native speakers of American English. For each subject, the reception threshold for sentences (RTS) was measured in two spatial configurations. In the asymmetric configuration, the target was presented from -45° azimuth and two colocated masking talkers were presented from +45° azimuth. In the diffuse configuration, the target was presented from 0° azimuth and four masking talkers were each presented from a different azimuth: +45°, +135°, -135°, and -45°. The new speech sentences, masking materials, and configurations were presented using low-pass filter cutoff frequencies of 4, 6, 8, and 10 kHz. For the normal-hearing participants, stimuli were presented in the sound field using loudspeakers. For the hearing-impaired participants, the spatial configurations were simulated using earphones, and a multiband wide-dynamic-range compressor with a modified CAM2 fitting algorithm was used to compensate for each participant's hearing loss.

RESULTS

For the normal-hearing participants (N = 24, mean age 40 years), the RTS improved significantly by 3.0 dB when the bandwidth was increased from 4 to 10 kHz, and a significant improvement of 1.3 dB was obtained from extending the bandwidth from 6 to 10 kHz, in both spatial configurations. Hearing-impaired participants (N = 25, mean age 71 years) also showed a significant improvement in RTS with extended bandwidth, but the effect was smaller than for the normal-hearing participants. The mean decrease in RTS when the bandwidth was increased from 4 to 10 kHz was 1.3 dB for the asymmetric condition and 0.5 dB for the diffuse condition.

CONCLUSIONS

Extending bandwidth from 4 to 10 kHz can improve the ability of normal-hearing and hearing-impaired participants to understand target speech in the presence of spatially separated masking speech. Future studies of the benefits of extended high-frequency amplification should investigate other realistic listening situations, masker types, spatial configurations, and room reverberation conditions, to determine added value in overcoming the technical challenges associated with implementing a device capable of providing extended high-frequency amplification.

Physiological motivated transmission-lines as front end for loudness models

The perception of loudness is strongly influenced by peripheral auditory processing, which calls for a physiologically correct peripheral auditory processing stage when constructing advanced loudness models. Most loudness models, however, rather follow a functional approach: a parallel auditory filter bank combined with a compression stage, followed by spectral and temporal integration. Such classical loudness models do not allow to directly link physiological measurements like otoacoustic emissions to properties of their auditory filterbank. However, this can be achieved with physiologically motivated transmission-line models (TLMs) of the cochlea. Here two active and nonlinear TLMs were tested as the peripheral front end of a loudness model. The TLMs are followed by a simple generic back end which performs integration of basilar-membrane "excitation" across place and time to yield a loudness estimate. The proposed model approach reaches similar performance as other state-of-the-art loudness models regarding the prediction of loudness in sones, equal-loudness contours (including spectral fine structure), and loudness as a function of bandwidth. The suggested model provides a powerful tool to directly connect objective measures of basilar membrane compression, such as distortion product otoacoustic emissions, and loudness in future studies.

Dynamic Range Across Music Genres and the Perception of Dynamic Compression in Hearing-Impaired Listeners

Dynamic range compression serves different purposes in the music and hearing-aid industries. In the music industry, it is used to make music louder and more attractive to normal-hearing listeners. In the hearing-aid industry, it is used to map the variable dynamic range of acoustic signals to the reduced dynamic range of hearing-impaired listeners. Hence, hearing-aided listeners will typically receive a dual dose of compression when listening to recorded music. The present study involved an acoustic analysis of dynamic range across a cross section of recorded music as well as a perceptual study comparing the efficacy of different compression schemes. The acoustic analysis revealed that the dynamic range of samples from popular genres, such as rock or rap, was generally smaller than the dynamic range of samples from classical genres, such as opera and orchestra. By comparison, the dynamic range of speech, based on recordings of monologues in quiet, was larger than the dynamic range of all music genres tested. The perceptual study compared the effect of the prescription rule NAL-NL2 with a semicompressive and a linear scheme. Music subjected to linear processing had the highest ratings for dynamics and quality, followed by the semicompressive and the NAL-NL2 setting. These findings advise against NAL-NL2 as a prescription rule for recorded music and recommend linear settings.

Effect of audibility on spatial release from speech-on-speech masking

This study investigated to what extent spatial release from masking (SRM) deficits in hearing-impaired adults may be related to reduced audibility of the test stimuli. Sixteen adults with sensorineural hearing loss and 28 adults with normal hearing were assessed on the Listening in Spatialized Noise-Sentences test, which measures SRM using a symmetric speech-on-speech masking task. Stimuli for the hearing-impaired listeners were delivered using three amplification levels (National Acoustic Laboratories - Revised Profound prescription (NAL-RP) +25%, and NAL-RP +50%), while stimuli for the normal-hearing group were filtered to achieve matched audibility. SRM increased as audibility increased for all participants. Thus, it is concluded that reduced audibility of stimuli may be a significant factor in hearing-impaired adults' reduced SRM even when hearing loss is compensated for with linear gain. However, the SRM achieved by the normal hearers with simulated audibility loss was still significantly greater than that achieved by hearing-impaired listeners, suggesting other factors besides audibility may still play a role.

Comparison of the CAM2A and NAL-NL2 hearing-aid fitting methods for participants with a wide range of hearing losses

OBJECTIVE

To compare preferences for sounds processed via a simulated five-channel compression hearing aid fitted using CAM2A and NAL-NL2.

DESIGN

Within a trial, the same segment of sound was presented twice, once with CAM2A settings and once with NAL-NL2 settings, in random order. The participant indicated which one was preferred and by how much. Stimuli included female and male speech in quiet and four types of music. The compression speed was slow or fast and the input sound level was 50, 65, or 80 dB SPL.

STUDY SAMPLE

Sixteen experienced hearing-aid users with a wide range of sensorineural hearing losses.

RESULTS

For both compression speeds, CAM2A was slightly preferred over NAL-NL2 for input levels of 65 and 80 dB, but NAL-NL2 was slightly preferred at 50 dB SPL.

CONCLUSIONS

Preferences for CAM2A relative to NAL-NL2 vary with input level. The results suggest that preferences for CAM2A might be increased by using lower gains for high frequencies and low input levels.

The influence of hearing-aid compression on forward-masked thresholds for adults with hearing loss

This paper describes forward-masked thresholds for adults with hearing loss. Previous research has demonstrated that the loss of cochlear compression contributes to deficits in this measure of temporal resolution. Cochlear compression can be mimicked with fast-acting compression where the normal dynamic range is mapped to the impaired dynamic range. To test the hypothesis that fast-acting compression will most-closely approximate the normal ability to perceive forward-masked pure-tones, forward-masked thresholds were measured for two groups of adults (normal hearing, hearing loss). Adults with normal hearing were tested without amplification. Adults with hearing loss were tested with three different compression speeds and two different prescriptive procedures using a hearing-aid simulator. The two prescriptive procedures differed in the extent to which the normal dynamic range was mapped onto the impaired dynamic range. When using a faster compression speed with the prescriptive procedure that best restored the lost dynamic range, forward-masked thresholds for the listeners with hearing loss approximated those observed for the listeners with normal hearing.

Parameter-based binaural hearing aid algorithms to improve speech intelligibility and localization in complex environments

This paper presents new binaural enhancement and noise suppression algorithms for binaural hearing aids. To enhance interaural level difference (ILD) cues at low frequencies, which are usually small, interaural time difference (ITD) cues are estimated and transformed to ILDs. The binaural noise suppression algorithm consists of adaptive beamforming and a coherence-based suppression filter. The estimated phase and signal-to-noise ratio (SNR) at each hearing aid are used to perform the processing. The performance of the proposed methods was assessed using perceptual evaluation with hearing-impaired listeners and objective evaluation.