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Auditory enhancement is a spectral contrast aftereffect that can facilitate the detection of novel events in an ongoing background. A single-interval paradigm combined with roved frequency content between trials can yield as much as 20 dB enhancement in young normal-hearing listeners. This study compared such enhancement in 15 listeners with sensorineural hearing loss with that in 15 age-matched adults and 15 young adults with normal audiograms. All groups were presented with stimulus levels of 70 dB sound pressure level (SPL) per component. The two groups with normal hearing were also tested at 45 dB SPL per component. The hearing-impaired listeners showed very little enhancement overall. However, when tested at the same high (70-dB) level, both young and age-matched normal-hearing listeners also showed substantially reduced enhancement, relative to that found at 45 dB SPL. Some differences in enhancement emerged between young and older normal-hearing listeners at the lower sound level. The results suggest that enhancement is highly level-dependent and may also decrease somewhat with age or slight hearing loss. Implications for hearing-impaired listeners may include a poorer ability to adapt to real-world acoustic variability, due in part to the higher levels at which sound must be presented to be audible., (© 2023 Acoustical Society of America.)

Recent studies have found that envelope following responses (EFRs) are a marker of age-related and noise- or ototoxic-induced cochlear synaptopathy (CS) in research animals. Whereas the cochlear injury can be well controlled in animal research studies, humans may have an unknown mixture of sensorineural hearing loss [SNHL; e.g., inner- or outer-hair-cell (OHC) damage or CS] that cannot be teased apart in a standard hearing evaluation. Hence, a direct translation of EFR markers of CS to a differential CS diagnosis in humans might be compromised by the influence of SNHL subtypes and differences in recording modalities between research animals and humans. To quantify the robustness of EFR markers for use in human studies, this study investigates the impact of methodological considerations related to electrode montage, stimulus characteristics, and presentation, as well as analysis method on human-recorded EFR markers. The main focus is on rectangularly modulated pure-tone stimuli to evoke the EFR based on a recent auditory modelling study that showed that the EFR was least affected by OHC damage and most sensitive to CS in this stimulus configuration. The outcomes of this study can help guide future clinical implementations of electroencephalography-based SNHL diagnostic tests.

Previous studies of level discrimination reported that listeners with high-frequency sensorineural hearing loss (SNHL) place greater weight on high frequencies than normal-hearing (NH) listeners. It is not clear whether these results are influenced by stimulus factors (e.g., group differences in presentation levels, cross-frequency discriminability of level differences used to measure weights) and whether such weights generalize to other tasks. Here, NH and SNHL weights were measured for level, duration, and frequency discrimination of two-tone complexes after measuring discriminability just-noticeable differences for each frequency and stimulus dimension. Stimuli were presented at equal sensation level (SL) or equal sound pressure level (SPL). Results showed that weights could change depending on which frequency contained the more discriminable level difference with uncontrolled cross-frequency discriminability. When cross-frequency discriminability was controlled, weights were consistent for level and duration discrimination, but not for frequency discrimination. Comparing equal SL and equal SPL weights indicated greater weight on the higher-level tone for level and duration discrimination. Weights were unrelated to improvements in recognition of low-pass-filtered speech with increasing cutoff frequency. These results suggest that cross-frequency weights and NH and SNHL weighting differences are influenced by stimulus factors and may not generalize to the use of speech cues in specific frequency regions.

This work presents a single-channel speech enhancement (SE) framework based on the super-Gaussian extension of the joint maximum a posteriori (SGJMAP) estimation rule. The developed SE algorithm is an open-source research smartphone-based application for hearing improvement studies. In this algorithm, the SGJMAP-based estimation for noisy speech mixture is smoothed along the frequency axis by a Mel filter-bank, resulting in a Mel-warped frequency-domain SGJMAP estimation. The impulse response of this Mel-warped estimation is obtained by applying a Mel-warped inverse discrete cosine transform (Mel-IDCT). This helps in filtering out the background noise and enhancing the speech signal. The proposed application is implemented on an iPhone (Apple, Cupertino, CA) to operate in real time and tested with normal-hearing (NH) and hearing-impaired (HI) listeners with different types of hearing aids through wireless connectivity. The objective speech quality and intelligibility test results are used to compare the performance of the proposed algorithm to existing conventional single-channel SE methods. Additionally, test results from NH and HI listeners show substantial improvement in speech recognition with the developed method in simulated real-world noisy conditions at different signal-to-noise ratio levels.

Real-time operation is critical for noise reduction in hearing technology. The essential requirement of real-time operation is causality-that an algorithm does not use future time-frame information and, instead, completes its operation by the end of the current time frame. This requirement is extended currently through the concept of "effectively causal," in which future time-frame information within the brief delay tolerance of the human speech-perception mechanism is used. Effectively causal deep learning was used to separate speech from background noise and improve intelligibility for hearing-impaired listeners. A single-microphone, gated convolutional recurrent network was used to perform complex spectral mapping. By estimating both the real and imaginary parts of the noise-free speech, both the magnitude and phase of the estimated noise-free speech were obtained. The deep neural network was trained using a large set of noises and tested using complex noises not employed during training. Significant algorithm benefit was observed in every condition, which was largest for those with the greatest hearing loss. Allowable delays across different communication settings are reviewed and assessed. The current work demonstrates that effectively causal deep learning can significantly improve intelligibility for one of the largest populations of need in challenging conditions involving untrained background noises.

The modified rhyme test [MRT; House, Williams, Hecker, and Kryter. (1965). J. Acoust. Soc. Am. 37, 158-166] is a widely used test for measuring the intelligibility of communication systems [ANSI (2009). S3.2 (American National Standards Institute, New York)] but has never gained widespread acceptance as a clinical test of speech intelligibility for listeners who are hearing impaired (HI). In this study, a clinical version of the MRT consisting of two 80-word lists was developed and tested on 2394 service members with varying levels of hearing loss. The test used a factorial design incorporating two speech levels [70 and 78 dB sound pressure level (SPL)], two signal-to-noise ratios (+4 and -4 dB), and two binaural conditions (diotic and binaural). High-frequency emphasis reduced the impact of audibility for HI listeners, focusing the test on the distortion component of hearing loss. The results show that listeners with normal hearing (NH) obtained an average score of 80% correct on the MRT 80 test. Listeners with a moderate hearing impairment scored an average of 70% correct. The overall level had little impact on performance for either NH or HI listeners. The results demonstrate that the MRT 80 test could be a useful test to assess the distortion effects of hearing loss on speech intelligibility, particularly in cases where it is desirable to use a closed-set test for automatic administration.

Children with sensorineural hearing loss show considerable variability in spoken language outcomes. The present study tested whether specific deficits in supra-threshold auditory perception might contribute to this variability. In a previous study by Halliday, Rosen, Tuomainen, and Calcus [(2019). J. Acoust. Soc. Am. 146, 4299], children with mild-to-moderate sensorineural hearing loss (MMHL) were shown to perform more poorly than those with normal hearing (NH) on measures designed to assess sensitivity to the temporal fine structure (TFS; the rapid oscillations in the amplitude of narrowband signals over short time intervals). However, they performed within normal limits on measures assessing sensitivity to the envelope (E; the slow fluctuations in the overall amplitude). Here, individual differences in unaided sensitivity to the TFS accounted for significant variance in the spoken language abilities of children with MMHL after controlling for nonverbal intelligence quotient, family history of language difficulties, and hearing loss severity. Aided sensitivity to the TFS and E cues was equally important for children with MMHL, whereas for children with NH, E cues were more important. These findings suggest that deficits in TFS perception may contribute to the variability in spoken language outcomes in children with sensorineural hearing loss.

Acoustic beamforming has been shown to improve identification of target speech in noisy listening environments for individuals with sensorineural hearing loss. This study examined whether beamforming would provide a similar benefit for individuals with aphasia (acquired neurological language impairment). The benefit of beamforming was examined for persons with aphasia (PWA) and age- and hearing-matched controls in both a speech masking condition and a speech-shaped, speech-modulated noise masking condition. Performance was measured when natural spatial cues were provided, as well as when the target speech level was enhanced via a single-channel beamformer. Because typical psychoacoustic methods may present substantial experimental confounds for PWA, clinically guided modifications of experimental procedures were determined individually for each PWA participant. Results indicated that the beamformer provided a significant overall benefit to listeners. On an individual level, both PWA and controls who exhibited poorer performance on the speech masking condition with spatial cues benefited from the beamformer, while those who achieved better performance with spatial cues did not. All participants benefited from the beamformer in the noise masking condition. The findings suggest that a spatially tuned hearing aid may be beneficial for older listeners with relatively mild hearing loss who have difficulty taking advantage of spatial cues.

Multiple auditory steady-state response (MASSR) is recommended to estimate hearing thresholds in difficult-to-test individuals. The multiple stimuli that evoke MASSR may present an interstimulus interaction (ISI) that is able to distort the generation of responses. No consensus exists on the effects of the ISI in MASSR when dealing with high sound level stimuli or cases of sensorineural hearing loss. This study investigated the effects of ISI on the amplitude and detectability of auditory steady-state responses, with a focus at and above 65 dB sound pressure level (SPL). Normal hearing (NH) and sensorineural hearing impaired (SNHI) adults were tested with different stimulus types [amplitude modulation (AM) One octave chirp (OC), and a weighted OC (WOC)], stimulus levels, and modalities (single or multiple stimuli). ISI typically attenuated response amplitude of a control stimulus caused by an interference stimulus one octave above the control stimulus. At and above 80 dB SPL, attenuations of around 50% decreased the number of detectable responses near SNHI thresholds, especially for OC and WOC. AM stimuli obtained a higher detection rate than OC and WOC when presented 10 dB above the behavioral hearing threshold of SNHI participants. Using OC in MASSR when assessing elevated thresholds might diminish accuracy on threshold estimation, and extend test duration.

This study investigated the effect of presentation level on spectral-ripple detection for listeners with and without sensorineural hearing loss (SNHL). Participants were 25 listeners with normal hearing and 25 listeners with SNHL. Spectral-ripple detection thresholds (SRDTs) were estimated at three spectral densities (0.5, 2, and 4 ripples per octave, RPO) and three to four sensation levels (SLs) (10, 20, 40, and, when possible, 60 dB SL). Each participant was also tested at 90 dB sound pressure level (SPL). Results indicate that level affected SRDTs. However, the effect of level depended on ripple density and hearing status. For all listeners and all RPO conditions, SRDTs improved from 10 to 40 dB SL. In the 2- and 4-RPO conditions, SRDTs became poorer from the 40 dB SL to the 90 dB SPL condition. The results suggest that audibility likely controls spectral-ripple detection at low SLs for all ripple densities, whereas spectral resolution likely controls spectral-ripple detection at high SLs and ripple densities. For optimal ripple detection across all listeners, clinicians and researchers should use a SL of 40 dB SL. To avoid absolute-level confounds, a presentation level of 80 dB SPL can also be used.

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.

Spatial perception is an important part of a listener's experience and ability to function in everyday environments. However, the current understanding of how well listeners can locate sounds is based on measurements made using relatively simple stimuli and tasks. Here the authors investigated sound localization in a complex and realistic environment for listeners with normal and impaired hearing. A reverberant room containing a background of multiple talkers was simulated and presented to listeners in a loudspeaker-based virtual sound environment. The target was a short speech stimulus presented at various azimuths and distances relative to the listener. To ensure that the target stimulus was detectable to the listeners with hearing loss, masked thresholds were first measured on an individual basis and used to set the target level. Despite this compensation, listeners with hearing loss were less accurate at locating the target, showing increased front-back confusion rates and higher root-mean-square errors. Poorer localization was associated with poorer masked thresholds and with more severe low-frequency hearing loss. Localization accuracy in the multitalker background was lower than in quiet and also declined for more distant targets. However, individual accuracy in noise and quiet was strongly correlated.

Listeners with sensorineural hearing loss routinely experience less spatial release from masking (SRM) in speech mixtures than listeners with normal hearing. Hearing-impaired listeners have also been shown to have degraded temporal fine structure (TFS) sensitivity, a consequence of which is degraded access to interaural time differences (ITDs) contained in the TFS. Since these "binaural TFS" cues are critical for spatial hearing, it has been hypothesized that degraded binaural TFS sensitivity accounts for the limited SRM experienced by hearing-impaired listeners. In this study, speech stimuli were noise-vocoded using carriers that were systematically decorrelated across the left and right ears, thus simulating degraded binaural TFS sensitivity. Both (1) ITD sensitivity in quiet and (2) SRM in speech mixtures spatialized using ITDs (or binaural release from masking; BRM) were measured as a function of TFS interaural decorrelation in young normal-hearing and hearing-impaired listeners. This allowed for the examination of the relationship between ITD sensitivity and BRM over a wide range of ITD thresholds. This paper found that, for a given ITD sensitivity, hearing-impaired listeners experienced less BRM than normal-hearing listeners, suggesting that binaural TFS sensitivity can account for only a modest portion of the BRM deficit in hearing-impaired listeners. However, substantial individual variability was observed.

Sensitivity to interaural time differences (ITDs) was measured in two groups of listeners, one with normal hearing and one with sensorineural hearing loss. ITD detection thresholds were measured for pure tones and for speech (a single word), in quiet and in the presence of noise. It was predicted that effects of hearing loss would be reduced for speech as compared to tones due to the redundancy of information across frequency. Thresholds were better overall, and the effects of hearing loss less pronounced, for speech than for tones. There was no evidence that effects of hearing loss were exacerbated in noise.

Auditory enhancement, where a target sound within a masker is rendered more audible by the prior presentation of the masker alone, may play an important role in auditory perception under variable everyday acoustic conditions. Cochlear hearing loss may reduce enhancement effects, potentially contributing to the difficulties experienced by hearing-impaired (HI) individuals in noisy and reverberant environments. However, it remains unknown whether, and by how much, enhancement under simultaneous masking is reduced in HI listeners. Enhancement of a pure tone under simultaneous masking with a multi-tone masker was measured in HI listeners and age-matched normal-hearing (NH) listeners as function of the spectral notch width of the masker, using stimuli at equal sensation levels as well as at equal sound pressure levels, but with the stimuli presented in noise to the NH listeners to maintain the equal sensation level between listener groups. The results showed that HI listeners exhibited some enhancement in all conditions. However, even when conditions were made as comparable as possible, in terms of effective spectral notch width and presentation level, the enhancement effect in HI listeners under simultaneous masking was reduced relative to that observed in NH listeners.

The goal of this study was to reconcile the differences between measures of loudness obtained with continuous, unbounded scaling procedures, such as magnitude estimation and production, and those obtained using a limited number of discrete categories, such as categorical loudness scaling (CLS). The former procedures yield data with ratio properties, but some listeners find it difficult to generate numbers proportional to loudness and the numbers cannot be compared across listeners to explore individual differences. CLS, where listeners rate loudness on a verbal scale, is an easier task, but the numerical values or categorical units (CUs) assigned to the points on the scale are not proportional to loudness. Sufficient CLS data are now available to assign values in sones, a scale proportional to loudness, to the loudness categories. As a demonstration of this approach, data from Heeren, Hohmann, Appell, and Verhey [J. Acoust. Soc. Am. 133, EL314-EL319 (2013)] were used to develop a CU sone metric, whose values were then substituted for the original CU values in reanalysis of a large set of CLS data obtained by Rasetshwane, Trevino, Gombert, Liebig-Trehearn, Kopun, Jesteadt, Neely, and Gorga [J. Acoust. Soc. Am. 137, 1899-1913 (2015)]. The resulting data are well fitted by power functions and are in general agreement with previously published results obtained with magnitude estimation, magnitude production, and cross modality matching.

Noise exposure and aging can damage cochlear synapses required for suprathreshold listening, even when cochlear structures needed for hearing at threshold remain unaffected. To control for effects of aging, behavioral amplitude modulation (AM) detection and subcortical envelope following responses (EFRs) to AM tones in 25 age-restricted (18-19 years) participants with normal thresholds, but different self-reported noise exposure histories were studied. Participants with more noise exposure had smaller EFRs and tended to have poorer AM detection than less-exposed individuals. Simulations of the EFR using a well-established cochlear model were consistent with more synaptopathy in participants reporting greater noise exposure.

A hearing-aid strategy that combines a beamforming microphone array in the high frequencies with natural binaural signals in the low frequencies was examined. This strategy attempts to balance the benefits of beamforming (improved signal-to-noise ratio) with the benefits of binaural listening (spatial awareness and location-based segregation). The crossover frequency was varied from 200 to 1200 Hz, and performance was compared to full-spectrum binaural and beamformer conditions. Speech intelligibility in the presence of noise or competing speech was measured in listeners with and without hearing loss. Results showed that the optimal crossover frequency depended on the listener and the nature of the interference.

This study investigated how six different amplification methods influence acoustic properties, and subsequently perception, of high-frequency cues in fricatives that have been processed with conventional full bandwidth amplification or nonlinear frequency compression (NFC)-12 conditions total. Amplification methods included linear gain, fast/slow-acting wide dynamic range compression crossed with fixed/individualized compression parameters, and a method with adaptive time constants. Twenty-one hearing-impaired listeners identified seven fricatives in nonsense syllables produced by female talkers. For NFC stimuli, frequency-compressed filters that precisely aligned 1/3-octave bands between input and output were used to quantify effective compression ratio, audibility, and temporal envelope modulation relative to the input. Results indicated significant relationships between these acoustic properties, each of which contributed significantly to fricative recognition across the entire corpus of stimuli. Recognition was significantly better for NFC stimuli compared with full bandwidth stimuli, regardless of the amplification method, which had complementary effects on audibility and envelope modulation. Furthermore, while there were significant differences in recognition across the amplification methods, they were not consistent across phonemes. Therefore, neither recognition nor acoustic data overwhelmingly suggest that one amplification method should be used over another for transmission of high-frequency cues in isolated syllables. Longer duration stimuli and more realistic listening conditions should be examined.

Otoacoustic emissions (OAEs) provide salient information about cochlear function and dysfunction. Two broad classes of emissions, linear reflection and nonlinear distortion, arise via distinct cochlear processes and hence, appear to provide independent information about cochlear health and hearing. Considered in combination, these two OAE types may characterize sensory hearing loss most effectively. In this study, the level-dependent growth of stimulus-frequency OAEs (a reflection-type emission) and distortion-product OAEs (a distortion-type emission) were measured in ten normal-hearing ears and eight ears with slight-to-moderate sensorineural hearing loss. Metrics of OAE strength and compression were derived from OAE input/output functions and then considered in a combined fashion. Results indicate that SFOAEs and DPOAEs differ significantly in their strength and compression features. When SFOAE and DPOAE metrics are displayed together on a two-dimensional plot, relatively well-defined data clusters describe their normative relationship. In hearing-impaired ears, this relationship is disrupted but not in a uniform way across ears; ears with similar audiograms showed differently altered joint-OAE profiles. Hearing loss sometimes affected only one OAE or one more than the other. Results suggest a joint-OAE profile is promising and warrants study in a large group of subjects with sensory hearing loss of varied etiologies.

This study examined the effects of age and hearing loss on recognition of speech presented when the auditory and visual speech information was misaligned in time (i.e., asynchronous). Prior research suggests that older listeners are less sensitive than younger listeners in detecting the presence of asynchronous speech for auditory-lead conditions, but recognition of speech in auditory-lead conditions has not yet been examined. Recognition performance was assessed for sentences and words presented in the auditory-visual modalities with varying degrees of auditory lead and lag. Detection of auditory-visual asynchrony for sentences was assessed to verify that listeners detected these asynchronies. The listeners were younger and older normal-hearing adults and older hearing-impaired adults. Older listeners (regardless of hearing status) exhibited a significant decline in performance in auditory-lead conditions relative to visual lead, unlike younger listeners whose recognition performance was relatively stable across asynchronies. Recognition performance was not correlated with asynchrony detection. However, one of the two cognitive measures assessed, processing speed, was identified in multiple regression analyses as contributing significantly to the variance in auditory-visual speech recognition scores. The findings indicate that, particularly in auditory-lead conditions, listener age has an impact on the ability to recognize asynchronous auditory-visual speech signals.

The objectives of this study were to determine if music perception by pediatric cochlear implant users can be improved by (1) providing access to bilateral hearing through two cochlear implants or a cochlear implant and a contralateral hearing aid (bimodal users) and (2) any history of music training. The Montreal Battery of Evaluation of Musical Ability test was presented via soundfield to 26 bilateral cochlear implant users, 8 bimodal users and 16 children with normal hearing. Response accuracy and reaction time were recorded via an iPad application. Bilateral cochlear implant and bimodal users perceived musical characteristics less accurately and more slowly than children with normal hearing. Children who had music training were faster and more accurate, regardless of their hearing status. Reaction time on specific subtests decreased with age, years of musical training and, for implant users, better residual hearing. Despite effects of these factors on reaction time, bimodal and bilateral cochlear implant users' responses were less accurate than those of their normal hearing peers. This means children using bilateral cochlear implants and bimodal devices continue to experience challenges perceiving music that are related to hearing impairment and/or device limitations during development.

Individuals with hearing impairment have particular difficulty perceptually segregating concurrent voices and understanding a talker in the presence of a competing voice. In contrast, individuals with normal hearing perform this task quite well. This listening situation represents a very different problem for both the human and machine listener, when compared to perceiving speech in other types of background noise. A machine learning algorithm is introduced here to address this listening situation. A deep neural network was trained to estimate the ideal ratio mask for a male target talker in the presence of a female competing talker. The monaural algorithm was found to produce sentence-intelligibility increases for hearing-impaired (HI) and normal-hearing (NH) listeners at various signal-to-noise ratios (SNRs). This benefit was largest for the HI listeners and averaged 59%-points at the least-favorable SNR, with a maximum of 87%-points. The mean intelligibility achieved by the HI listeners using the algorithm was equivalent to that of young NH listeners without processing, under conditions of identical interference. Possible reasons for the limited ability of HI listeners to perceptually segregate concurrent voices are reviewed as are possible implementation considerations for algorithms like the current one.

The masking release (i.e., better speech recognition in fluctuating compared to continuous noise backgrounds) observed for normal-hearing (NH) listeners is generally reduced or absent in hearing-impaired (HI) listeners. One explanation for this lies in the effects of reduced audibility: elevated thresholds may prevent HI listeners from taking advantage of signals available to NH listeners during the dips of temporally fluctuating noise where the interference is relatively weak. This hypothesis was addressed through the development of a signal-processing technique designed to increase the audibility of speech during dips in interrupted noise. This technique acts to (i) compare short-term and long-term estimates of energy, (ii) increase the level of short-term segments whose energy is below the average energy, and (iii) normalize the overall energy of the processed signal to be equivalent to that of the original long-term estimate. Evaluations of this energy-equalizing (EEQ) technique included consonant identification and sentence reception in backgrounds of continuous and regularly interrupted noise. For HI listeners, performance was generally similar for processed and unprocessed signals in continuous noise; however, superior performance for EEQ processing was observed in certain regularly interrupted noise backgrounds.

Distortion-product otoacoustic emissions (DPOAEs) arise in the cochlea in response to two tones with frequencies f 1 and f 2 and mainly consist of two components, a nonlinear-distortion and a coherent-reflection component. Wave interference between these components limits the accuracy of DPOAEs when evaluating the function of the cochlea with conventional continuous stimulus tones. Here, DPOAE components are separated in the time domain from DPOAE signals elicited with short stimulus pulses. The extracted nonlinear-distortion components are used to derive estimated distortion-product thresholds (EDPTs) from semi-logarithmic input-output (I/O) functions for 20 normal-hearing and 21 hearing-impaired subjects. I/O functions were measured with frequency-specific stimulus levels at eight frequencies f 2  = 1,…, 8 kHz (f 2 /f 1  = 1.2). For comparison, DPOAEs were also elicited with continuous primary tones. Both acquisition paradigms yielded EDPTs, which significantly correlated with behavioral thresholds (p < 0.001) and enabled derivation of estimated hearing thresholds (EHTs) from EDPTs using a linear regression relationship. DPOAE-component separation in the time domain significantly reduced the standard deviation of EHTs compared to that derived from continuous DPOAEs (p < 0.01). In conclusion, using frequency-specific stimulus levels and DPOAE-component separation increases the reliability of DPOAE I/O functions for assessing cochlear function and estimating behavioral thresholds.

Old, hearing-impaired listeners generally benefit little from lateral separation of multiple talkers when listening to one of them. This study aimed to determine how spatial release from masking (SRM) in such listeners is affected when the interaural time differences (ITDs) in the temporal fine structure (TFS) are manipulated by tone-vocoding (TVC) at the ears by a master hearing aid system. Word recall was compared, with and without TVC, when target and masker sentences from a closed set were played simultaneously from the front loudspeaker (co-located) and when the maskers were played 45° to the left and right of the listener (separated). For 20 hearing-impaired listeners aged 64 to 86, SRM was 3.7 dB smaller with TVC than without TVC. This difference in SRM correlated with mean audiometric thresholds below 1.5 kHz, even when monaural TFS sensitivity (discrimination of frequency-shifts in identically filtered complexes) was partialed out, suggesting that low-frequency audiometric thresholds may be a good indicator of candidacy for hearing aids that preserve ITDs. The TVC difference in SRM was not correlated with age, pure-tone ITD thresholds, nor fundamental frequency difference limens, and only with monaural TFS sensitivity before control for low-frequency audiometric thresholds.

Adaptation to speech with a foreign accent is possible through prior exposure to talkers with that same accent. For young listeners with normal hearing, short term, accent-independent adaptation to a novel foreign accent is also facilitated through exposure training with multiple foreign accents. In the present study, accent-independent adaptation is examined in younger and older listeners with normal hearing and older listeners with hearing loss. Retention of training benefit is additionally explored. Stimuli for testing and training were HINT sentences recorded by talkers with nine distinctly different accents. Following two training sessions, all listener groups showed a similar increase in speech perception for a novel foreign accent. While no group retained this benefit at one week post-training, results of a secondary reaction time task revealed a decrease in reaction time following training, suggesting reduced listening effort. Examination of listeners' cognitive skills reveals a positive relationship between working memory and speech recognition ability. The present findings indicate that, while this no-feedback training paradigm for foreign-accented English is successful in promoting short term adaptation for listeners, this paradigm is not sufficient in facilitation of perceptual learning with lasting benefits for younger or older listeners.

Although many studies have evaluated the performance of virtual audio displays with normal hearing listeners, very little information is available on the effect that hearing loss has on the localization of virtual sounds. In this study, normal hearing (NH) and hearing impaired (HI) listeners were asked to localize noise stimuli with short (250 ms), medium (1000 ms), and long (4000 ms) durations both in the free field and with a non-individualized head-tracked virtual audio display. The results show that the HI listeners localized sounds less accurately than the NH listeners, and that both groups consistently localized virtual sounds less accurately than free-field sounds. These results indicate that HI listeners are sensitive to individual differences in head related transfer functions (HRTFs), which means that they might have difficulty using auditory display systems that rely on generic HRTFs to control the apparent locations of virtual sounds. However, the results also reveal a high correlation between free-field and virtual localization performance in the HI listeners. This suggests that it may be feasible to use non-individualized virtual audio display systems to predict the auditory localization performance of HI listeners in clinical environments where free-field speaker arrays are not available.

This study investigated the effects of fast-acting hearing-aid compression on normal-hearing and hearing-impaired listeners' spatial perception in a reverberant environment. Three compression schemes-independent compression at each ear, linked compression between the two ears, and "spatially ideal" compression operating solely on the dry source signal-were considered using virtualized speech and noise bursts. Listeners indicated the location and extent of their perceived sound images on the horizontal plane. Linear processing was considered as the reference condition. The results showed that both independent and linked compression resulted in more diffuse and broader sound images as well as internalization and image splits, whereby more image splits were reported for the noise bursts than for speech. Only the spatially ideal compression provided the listeners with a spatial percept similar to that obtained with linear processing. The same general pattern was observed for both listener groups. An analysis of the interaural coherence and direct-to-reverberant ratio suggested that the spatial distortions associated with independent and linked compression resulted from enhanced reverberant energy. Thus, modifications of the relation between the direct and the reverberant sound should be avoided in amplification strategies that attempt to preserve the natural sound scene while restoring loudness cues.

The abilities of 59 adult hearing-aid users to hear phonetic details were assessed by measuring their abilities to identify syllable constituents in quiet and in differing levels of noise (12-talker babble) while wearing their aids. The set of sounds consisted of 109 frequently occurring syllable constituents (45 onsets, 28 nuclei, and 36 codas) spoken in varied phonetic contexts by eight talkers. In nominal quiet, a speech-to-noise ratio (SNR) of 40 dB, scores of individual listeners ranged from about 23% to 85% correct. Averaged over the range of SNRs commonly encountered in noisy situations, scores of individual listeners ranged from about 10% to 71% correct. The scores in quiet and in noise were very strongly correlated, R = 0.96. This high correlation implies that common factors play primary roles in the perception of phonetic details in quiet and in noise. Otherwise said, hearing-aid users' problems perceiving phonetic details in noise appear to be tied to their problems perceiving phonetic details in quiet and vice versa.

Consonant-vowel (CV) perception experiments provide valuable insights into how humans process speech. Here, two CV identification experiments were conducted in a group of hearing-impaired (HI) listeners, using 14 consonants followed by the vowel /ɑ/. The CVs were presented in quiet and with added speech-shaped noise at signal-to-noise ratios of 0, 6, and 12 dB. The HI listeners were provided with two different amplification schemes for the CVs. In the first experiment, a frequency-independent amplification (flat-gain) was provided and the CVs were presented at the most-comfortable loudness level. In the second experiment, a frequency-dependent prescriptive gain was provided. The CV identification results showed that, while the average recognition error score obtained with the frequency-dependent amplification was lower than that obtained with the flat-gain, the main confusions made by the listeners on a token basis remained the same in a majority of the cases. An entropy measure and an angular distance measure were proposed to assess the highly individual effects of the frequency-dependent gain on the consonant confusions in the HI listeners. The results suggest that the proposed measures, in combination with a well-controlled phoneme speech test, may be used to assess the impact of hearing-aid signal processing on speech intelligibility.

The extent to which informational masking (IM) is involved in real-world listening is not well understood. In the literature, IM effects of more than 8 dB are reported, but these experiments typically used simplified spatial configurations and speech materials with exaggerated confusions. Westermann and Buchholz [(2015b). J. Acoust. Soc. Am. 138, 584-593] considered a simulated cafeteria and found only substantial IM effects when the target and maskers were colocated and the same talker. The present study further investigates the relevance of IM in real-world environments, specifically distractions by nearby maskers and the effect of hearing impairment. Speech reception thresholds (SRTs) were measured with normal hearing (NH) and sensorineural hearing impaired (HI) listeners in a simulated cafeteria environment. Three different masker configurations were considered: (1) seven dialogues distributed in the cafeteria, (2) two monologues presented close to the listener with varying angular separation, and (3) a combination of (1) and (2). The contribution of IM was measured as the difference in SRTs between speech maskers and unintelligible vocoded maskers. No significant IM was found with the seven dialogues alone. Including nearby maskers resulted in substantial IM for both NH and HI listeners, suggesting that such maskers might result in IM in real-world environments.

Interaural differences in time (ITDs) and interaural differences in level (ILDs) contribute to a listener's ability to achieve spatial release from masking (SRM), and help to improve speech intelligibility in noisy environments. In this study, the extent to which ITDs and ILDs contribute to SRM and the relationships with aging and hearing loss were examined. SRM was greatest when stimuli were presented with consistent ITD and ILD, relative to ITD or ILD alone, all of which produced greater SRM than when ITD and ILD cues were in conflict with each other. This pattern was independent of age and hearing loss.

Some Japanese monosyllables contain consonants that are not easily discernible for individuals with sensorineural hearing loss. However, the acoustic features that make these monosyllables difficult to discern have not been clearly identified. Here, this study used the autocorrelation function (ACF), which can capture temporal features of signals, to clarify the factors influencing speech intelligibility. For each monosyllable, five factors extracted from the ACF [Φ(0): total energy; τ 1 and ϕ 1 : delay time and amplitude of the maximum peak; τ e : effective duration; W ϕ (0) : spectral centroid], voice onset time, speech intelligibility index, and loudness level were compared with the percentage of correctly perceived articulations (144 ears) obtained by 50 Japanese vowel and consonant-vowel monosyllables produced by one female speaker. Results showed that median effective duration [(τ e ) med ] was strongly correlated with the percentage of correctly perceived articulations of the consonants (r = 0.87, p < 0.01). (τ e ) med values were computed by running ACFs with the time lag at which the magnitude of the logarithmic-ACF envelope had decayed to -10 dB. Effective duration is a measure of temporal pattern persistence, i.e., the duration over which the waveform maintains a stable pattern. The authors postulate that low recognition ability is related to degraded perception of temporal fluctuation patterns.

In many situations, listeners with sensorineural hearing loss demonstrate reduced spatial release from masking compared to listeners with normal hearing. This deficit is particularly evident in the "symmetric masker" paradigm in which competing talkers are located to either side of a central target talker. However, there is some evidence that reduced target audibility (rather than a spatial deficit per se) under conditions of spatial separation may contribute to the observed deficit. In this study a simple "glimpsing" model (applied separately to each ear) was used to isolate the target information that is potentially available in binaural speech mixtures. Intelligibility of these glimpsed stimuli was then measured directly. Differences between normally hearing and hearing-impaired listeners observed in the natural binaural condition persisted for the glimpsed condition, despite the fact that the task no longer required segregation or spatial processing. This result is consistent with the idea that the performance of listeners with hearing loss in the spatialized mixture was limited by their ability to identify the target speech based on sparse glimpses, possibly as a result of some of those glimpses being inaudible.

Supervised speech segregation has been recently shown to improve human speech intelligibility in noise, when trained and tested on similar noises. However, a major challenge involves the ability to generalize to entirely novel noises. Such generalization would enable hearing aid and cochlear implant users to improve speech intelligibility in unknown noisy environments. This challenge is addressed in the current study through large-scale training. Specifically, a deep neural network (DNN) was trained on 10 000 noises to estimate the ideal ratio mask, and then employed to separate sentences from completely new noises (cafeteria and babble) at several signal-to-noise ratios (SNRs). Although the DNN was trained at the fixed SNR of - 2 dB, testing using hearing-impaired listeners demonstrated that speech intelligibility increased substantially following speech segregation using the novel noises and unmatched SNR conditions of 0 dB and 5 dB. Sentence intelligibility benefit was also observed for normal-hearing listeners in most noisy conditions. The results indicate that DNN-based supervised speech segregation with large-scale training is a very promising approach for generalization to new acoustic environments.

The effects of age and hearing loss on recognition of unaccented and accented words of varying syllable length were investigated. It was hypothesized that with increments in length of syllables, there would be atypical alterations in syllable stress in accented compared to native English, and that these altered stress patterns would be sensitive to auditory temporal processing deficits with aging. Sets of one-, two-, three-, and four-syllable words with the same initial syllable were recorded by one native English and two Spanish-accented talkers. Lists of these words were presented in isolation and in sentence contexts to younger and older normal-hearing listeners and to older hearing-impaired listeners. Hearing loss effects were apparent for unaccented and accented monosyllabic words, whereas age effects were observed for recognition of accented multisyllabic words, consistent with the notion that altered syllable stress patterns with accent are sensitive for revealing effects of age. Older listeners also exhibited lower recognition scores for moderately accented words in sentence contexts than in isolation, suggesting that the added demands on working memory for words in sentence contexts impact recognition of accented speech. The general pattern of results suggests that hearing loss, age, and cognitive factors limit the ability to recognize Spanish-accented speech.

When competing speech sounds are spatially separated, listeners can make use of the ear with the better target-to-masker ratio. Recent studies showed that listeners with normal hearing are able to efficiently make use of this "better-ear," even when it alternates between left and right ears at different times in different frequency bands, which may contribute to the ability to listen in spatialized speech mixtures. In the present study, better-ear glimpsing in listeners with bilateral sensorineural hearing impairment, who perform poorly in spatialized speech mixtures, was investigated. The results suggest that this deficit is not related to better-ear glimpsing.

How age and hearing loss affect the perception of interrupted speech may vary based on both the physical properties of preserved or obliterated speech fragments and individual listener characteristics. To investigate perceptual processes and interruption parameters influencing intelligibility across interruption rates, participants of different age and hearing status heard sentences interrupted by silence at either a single primary rate (0.5-8 Hz; 25%, 50%, 75% duty cycle) or at an additional concurrent secondary rate (24 Hz; 50% duty cycle). Although age and hearing loss significantly affected intelligibility, the ability to integrate sub-phonemic speech fragments produced by the fast secondary rate was similar in all listener groups. Age and hearing loss interacted with rate with smallest group differences observed at the lowest and highest interruption rates of 0.5 and 24 Hz. Furthermore, intelligibility of dual-rate gated sentences was higher than single-rate gated sentences with the same proportion of retained speech. Correlations of intelligibility of interrupted speech to pure-tone thresholds, age, or measures of working memory and auditory spectro-temporal pattern discrimination were generally low-to-moderate and mostly nonsignificant. These findings demonstrate rate-dependent effects of age and hearing loss on the perception of interrupted speech, suggesting complex interactions of perceptual processes across different time scales.

Consonant recognition was assessed following extraction of speech from noise using a more efficient version of the speech-segregation algorithm described in Healy, Yoho, Wang, and Wang [(2013) J. Acoust. Soc. Am. 134, 3029-3038]. Substantial increases in recognition were observed following algorithm processing, which were significantly larger for hearing-impaired (HI) than for normal-hearing (NH) listeners in both speech-shaped noise and babble backgrounds. As observed previously for sentence recognition, older HI listeners having access to the algorithm performed as well or better than young NH listeners in conditions of identical noise. It was also found that the binary masks estimated by the algorithm transmitted speech features to listeners in a fashion highly similar to that of the ideal binary mask (IBM), suggesting that the algorithm is estimating the IBM with substantial accuracy. Further, the speech features associated with voicing, manner of articulation, and place of articulation were all transmitted with relative uniformity and at relatively high levels, indicating that the algorithm and the IBM transmit speech cues without obvious deficiency. Because the current implementation of the algorithm is much more efficient, it should be more amenable to real-time implementation in devices such as hearing aids and cochlear implants.

This study explored a range of training dosages and durations for a word-based auditory-training procedure for older adults with hearing impairment. Three groups received a different "dose": 2x/week; 3x/week; no training. Fifteen training sessions comprised a "cycle" which was repeated three times for each dosage. Groups that completed training performed significantly better than controls for speech-in-noise materials included in the training regimen, with no significant difference observed between the 2x or 3x/week training groups. Based on these results, as well as prior literature on learning theory, training 2x or 3x/week for 5-15 weeks appears to be sufficient to yield training benefits with this training regimen.

The summation of loudness across ears is often studied by measuring the level difference required for equal loudness (LDEL) of monaural and diotic sounds. Typically, the LDEL is ∼5-6 dB, consistent with the idea that a diotic sound is ∼1.5 times as loud as the same sound presented monaurally at the same level, as predicted by the loudness model of Moore and Glasberg [J. Acoust. Soc. Am. 121, 1604-1612 (2007)]. One might expect that the LDEL would be <5-6 dB for hearing-impaired listeners, because loudness recruitment leads to a more rapid change of loudness for a given change in level. However, previous data sometimes showed similar LDEL values for normal-hearing and hearing-impaired listeners. Here, the LDEL was measured for hearing-impaired listeners using narrowband and broadband noises centered at 500 Hz, where audiometric thresholds were near-normal, and at 3000 or 4000 Hz, where audiometric thresholds were elevated. The mean LDEL was 5.6 dB at 500 Hz and 4.2 dB at the higher center frequencies. The results were predicted reasonably well by an extension of the loudness model of Moore and Glasberg.

The aim of this study was to investigate changes in central auditory processing following unilateral and bilateral hearing aid fitting using a combination of physiological and behavioral measures: late auditory event-related potentials (ERPs) and speech recognition in noise, respectively. The hypothesis was that for fitted ears, the ERP amplitude would increase over time following hearing aid fitting in parallel with improvement in aided speech recognition. The N1 and P2 ERPs were recorded to 500 and 3000 Hz tones presented at 65, 75, and 85 dB sound pressure level to either the left or right ear. New unilateral and new bilateral hearing aid users were tested at the time of first fitting and after 12 weeks hearing aid use. A control group of long-term hearing aid users was tested over the same time frame. No significant changes in the ERP were observed for any group. There was a statistically significant 2% improvement in aided speech recognition over time for all groups, although this was consistent with a general test-retest effect. This study does not support the existence of an acclimatization effect observable in late ERPs following 12 weeks' hearing aid use.

The present investigation carried out acoustic analyses of vowels in clear and conversational speech produced by 41 talkers. Mixed-effects models were then deployed to examine relationships among acoustic and perceptual data for these vowels. Acoustic data include vowel duration, steady-state formant frequencies, and two measures of dynamic formant movement. Perceptual data consist of vowel intelligibility in noise for young normal-hearing and elderly hearing-impaired listeners, as reported by Ferguson in 2004 and 2012 [J. Acoust. Soc. Am. 116, 2365-2373 (2004); J. Speech Lang. Hear. Res. 55, 779-790 (2012)], respectively. Significant clear speech effects were observed for all acoustic metrics, although not all measures changed for all vowels and considerable talker variability was observed. Mixed-effects analyses revealed that the contribution of duration and steady-state formant information to vowel intelligibility differed for the two listener groups. This outcome is consistent with earlier research suggesting that hearing loss, and possibly aging, alters the way acoustic cues are used for identifying vowels.

The premise of this study is that models of hearing, in general, and of individual hearing impairment, in particular, can be improved by using speech test results as an integral part of the modeling process. A conceptual iterative procedure is presented which, for an individual, considers measures of sensitivity, cochlear compression, and phonetic confusions using the Diagnostic Rhyme Test (DRT) framework. The suggested approach is exemplified by presenting data from three hearing-impaired listeners and results obtained with models of the hearing impairment of the individuals. The work reveals that the DRT data provide valuable information of the damaged periphery and that the non-speech and speech data are complementary in obtaining the best model for an individual.

Temporal processing ability for the senses of hearing and touch was examined through the measurement of gap-duration discrimination thresholds (GDDTs) employing the same low-frequency sinusoidal stimuli in both modalities. GDDTs were measured in three groups of observers (normal-hearing, hearing-impaired, and normal-hearing with simulated hearing loss) covering an age range of 21-69 yr. GDDTs for a baseline gap of 6 ms were measured for four different combinations of 100-ms leading and trailing markers (250-250, 250-400, 400-250, and 400-400 Hz). Auditory measurements were obtained for monaural presentation over headphones and tactile measurements were obtained using sinusoidal vibrations presented to the left middle finger. The auditory GDDTs of the hearing-impaired listeners, which were larger than those of the normal-hearing observers, were well-reproduced in the listeners with simulated loss. The magnitude of the GDDT was generally independent of modality and showed effects of age in both modalities. The use of different-frequency compared to same-frequency markers led to a greater deterioration in auditory GDDTs compared to tactile GDDTs and may reflect differences in bandwidth properties between the two sensory systems.

Hearing-impaired (HI) listeners often show less masking release (MR) than normal-hearing listeners when temporal fluctuations are imposed on a steady-state masker, even when accounting for overall audibility differences. This difference may be related to a loss of cochlear compression in HI listeners. Behavioral estimates of compression, using temporal masking curves (TMCs), were compared with MR for band-limited (500-4000 Hz) speech and pure tones in HI listeners and age-matched, noise-masked normal-hearing (NMNH) listeners. Compression and pure-tone MR estimates were made at 500, 1500, and 4000 Hz. The amount of MR was defined as the difference in performance between steady-state and 10-Hz square-wave-gated speech-shaped noise. In addition, temporal resolution was estimated from the slope of the off-frequency TMC. No significant relationship was found between estimated cochlear compression and MR for either speech or pure tones. NMNH listeners had significantly steeper off-frequency temporal masking recovery slopes than did HI listeners, and a small but significant correlation was observed between poorer temporal resolution and reduced MR for speech. The results suggest either that the effects of hearing impairment on MR are not determined primarily by changes in peripheral compression, or that the TMC does not provide a sufficiently reliable measure of cochlear compression.

Despite considerable effort, monaural (single-microphone) algorithms capable of increasing the intelligibility of speech in noise have remained elusive. Successful development of such an algorithm is especially important for hearing-impaired (HI) listeners, given their particular difficulty in noisy backgrounds. In the current study, an algorithm based on binary masking was developed to separate speech from noise. Unlike the ideal binary mask, which requires prior knowledge of the premixed signals, the masks used to segregate speech from noise in the current study were estimated by training the algorithm on speech not used during testing. Sentences were mixed with speech-shaped noise and with babble at various signal-to-noise ratios (SNRs). Testing using normal-hearing and HI listeners indicated that intelligibility increased following processing in all conditions. These increases were larger for HI listeners, for the modulated background, and for the least-favorable SNRs. They were also often substantial, allowing several HI listeners to improve intelligibility from scores near zero to values above 70%.

Dynamic range compression is widely used to reduce the difference between the most and least intense portions of a signal. Such compression distorts the shape of the amplitude envelope of a signal, but it is unclear to what extent such distortions are actually perceivable by listeners. Here, the ability to distinguish between compressed and uncompressed versions of a noise vocoded sentence was initially measured in listeners with normal hearing while varying the threshold, ratio, attack, and release parameters. This narrow condition was selected in order to characterize perception under the most favorable listening conditions. The average behavioral sensitivity to compression was highly correlated to several acoustical indices of modulation depth. In particular, performance was highly correlated to the Euclidean distance between the modulation spectra of the uncompressed and compressed signals. Suggesting that this relationship is not restricted to the initial test conditions, the correlation remained largely unchanged both (1) when listeners with normal hearing were tested using a time-compressed version of the original signal, and (2) when listeners with impaired hearing were tested using the original signal. If this relationship generalizes to more ecologically valid conditions, it will provide a straightforward method for predicting the detectability of compression-induced distortions.

Estimating audiometric thresholds using objective measures can be clinically useful when reliable behavioral information cannot be obtained. Transient-evoked otoacoustic emissions (TEOAEs) are effective for determining hearing status (normal hearing vs hearing loss), but previous studies have found them less useful for predicting audiometric thresholds. Recent work has demonstrated the presence of short-latency TEOAE components in normal-hearing ears, which have typically been eliminated from the analyses used in previous studies. The current study investigated the ability of short-latency components to predict hearing status and thresholds from 1-4 kHz. TEOAEs were measured in 77 adult ears with thresholds ranging from normal hearing to moderate sensorineural hearing loss. Emissions were bandpass filtered at center frequencies from 1 to 4 kHz. TEOAE waveforms were analyzed within two time windows that contained either short- or long-latency components. Waveforms were quantified by root-mean-square amplitude. Long-latency components were better overall predictors of hearing status and thresholds, relative to short-latency components. There were no significant improvements in predictions when short-latency components were included with long-latency components in multivariate analyses. The results showed that short-latency TEOAE components, as analyzed in the current study, were less predictive of both hearing status and thresholds from 1-4 kHz than long-latency components.