Proceedings of the International Symposium on Auditory and Audiological Research
The ISAAR proceedings are a collection of research articles in the fields of auditory and audiological research. Each article is based on work that was presented at the International Symposium on Auditory and Audiological Research, which takes place every other year.
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Timing of turn taking between normal-hearing and hearing-impaired interlocutors
Having a conversation requires more resources than just understanding speech. Previous studies of the timing of turn taking in conversations suggest that in order to sustain normal, fluid turn taking, interlocutors have to predict the end of each other’s turns. Thus, while noise and hearing loss should make understanding speech more difficult, it should also reduce the resources available for speech planning and possibly reduce the saliency of cues used to predict turn ends, resulting in delayed and more variable turn taking. We recorded conversations between 12 pairs of native-Danish young normal-hearing (NH) and older hearing-impaired (HI) listeners with mild presbyacusis in quiet and multitalker babble at three levels. The interlocutors conducted a Diapix task, finding differences in two near-identical pictures. Both HI and NH talkers responded more slowly and with more variability with increasing noise level, and the HI with more variability than the NH. We saw indications that the younger NH adopted a more careful communication strategy, likely to ease the effort on their older HI interlocutor, by adapting their speech rates to their interlocutor and overlapping less.
Copyright (c) 2020 Anna Josefine Munch Sørensen, Ewen N MacDonald, Thomas Lunner
The effect of harmonic number and pitch salience on the ability to understand speech-on-speech based on differences in fundamental frequency
Differences in fundamental frequency (F0) between competing voices facilitate the ability to segregate a target voice from interferers, thereby enhancing speech intelligibility. Although lower-numbered harmonics produce greater pitch salience than higher-numbered harmonics, it remains unclear whether differences in harmonic ranks, and therefore pitch salience, affect the benefit of pitch differences. Earlier studies have not reported an effect of pitch salience, but have generally used only conditions where the difference in average F0 (ΔF0) between the two competing voices was large. It is possible that the effect of pitch salience is greater in more challenging conditions, in which the ΔF0 is relatively small. This study tested speech intelligibility in the presence of one speech masker for ΔF0s of 0, 2, and 4 semitones. The speech was presented in a broadband condition or was highpass or lowpass filtered to manipulate the pitch salience of the voicing. Results showed no interaction between filter type and ΔF0, suggesting little or no effect of harmonic rank or pitch salience in the ability to use F0 to segregate voices, even with smaller ΔF0s between competing voices. The results suggest some benefit of ΔF0 between competing voices, even in the absence of low-numbered spectrally resolved harmonics.
Copyright (c) 2020 Sara Miay Kim Madsen, Torsten Dau, Andrew J Oxenham
Perceptual learning and speech perception: A new hypothesis
Perceptual learning for speech remains substantial even in older adults, but the functional significance of this observation is not well understood. It has been suggested that perceptual learning might serve to support listening in adverse conditions by promoting behavioural and neural plasticity, but this hypothesis is not consistent with the acoustic specificity of learning. Instead, we now suggest that in the context of speech perception, perceptual learning might be best viewed as one of the capacities that, like working memory, support speech perception in an on-line fashion. Consistent with this hypothesis, we present data that rapid perceptual learning of one speech task accounts for substantial individual differences in other speech tasks even after accounting for the potential correlations between different indices of speech perception.
Copyright (c) 2020 Karen Banai, Limor Lavie
The effect of conversational task on turn taking in dialogue
In previous studies, several methods have been used to elicit conversation between talkers. Some involved participants solving a shared task (e.g., describing a map or finding differences between two near-identical pictures), while others have recorded more spontaneous dialogue (e.g., telephone calls). Since the goals of the talkers, and thus the definition of successful conversation, varies across these methods, it is thought likely that turn-taking behaviour will vary depending on how conversations are elicited. The present study investigated this by eliciting English conversations from 7 pairs of native-Danish talkers using two methods: solving a Diapix task and engaging in unguided “small talk”. For each method, in both quiet and 70 dBA babble, two conversations were recorded for each pair. Overall, several differences in conversational behaviour were observed. When engaged in “small talk”, participants spoke more rapidly, produced longer utterances, and replied more quickly than compared to when they were solving the Diapix task. These within-pair differences indicate that comparisons of behaviour across studies should also consider the method by which conversations were elicited.
Copyright (c) 2020 Sam Watson, Anna Josefine Munch Sørensen, Ewen MacDonald
Duration threshold for identifying speech samples for different phonemes
The identification or classification of acoustic objects is important to decide in which way a sound needs to be interpreted and to rate its importance or relevance. In recent studies, it has been shown that the minimal duration of a sound, which is required for a correct identification, could be a useful audiological parameter, e.g. providing information about the hearing ability of a person. In this work, we want to investigate which cues are used by humans to classify a sound correctly as speech. For this purpose, the duration thresholds for the identification of speech samples starting with different phonemes are analyzed for elderly listeners with normal and impaired hearing. To this end, a two-alternative forced choice (2-AFC) method was used, where, as an alternative to speech, a noise signal with a matched frequency spectrum was presented. In contrast to previous studies, there were no frequency cues available and we found no correlation to the pure tone average (PTA) or speech understanding in noise. As one main conclusion, the results suggest that humans primarily exploit the temporal envelope (ENV) rather than the temporal fine structure (TFS) for the identification of short speech samples above hearing threshold and without frequency cues.
Copyright (c) 2020 Hendrik Husstedt, Simone Wollermann, Daniel Bank, Mario Schinnerl, Marlitt Frenz, Jürgen Tchorz
Assessing the impact of fundamental frequency on speech intelligibility in competing-talker scenarios
When only monaural cues are available in competing-talker scenarios, normal- hearing (NH) listeners are able to identify and understand the target speech while hearing-impaired listeners often experience difficulties. A good understanding of the role of monaural cues in speech segregation is therefore essential for developing hearing-aid compensation strategies. Earlier studies with NH listeners showed that differences in fundamental frequency (∆F0) between the target talker and one interfering talker can facilitate the segregation of the speech signals. However, most of these studies used speech materials that bear little resemblance with everyday speech. Furthermore, the F0 was either defined by talker sex or measured as a talker-specific average, thus ignoring the significant F0 variability across sentences. The present study instead used everyday-speech type sentences from the Danish Hearing in Noise Test (HINT) and employed a more accurate method for assessing the impact of F0 on intelligibility for NH listeners. Compared to previous studies, the overall effect of ∆F0 was found to be smaller and it was hypothesised that the previously employed speech materials might have enhanced the effect of ∆F0 beyond its real-life importance.
Copyright (c) 2020 Paolo A. Mesiano, Johannes Zaar, Lars Bramsløw, Niels H. Pontoppidan, Torsten Dau
Effects of noise and L2 on the timing of turn taking in conversation
Previous studies of floor-transfer offsets (FTO), the offset from when one talker stops talking to the next one starts, suggest that normal conversation requires interlocutors to predict when each other will finish their turn. We hypothesized that increasing the difficulty of holding a conversation by adding noise and/or speaking in a second language (L2) would result in longer FTOs. Conversations from 20 pairs of normal-hearing (NH), native-Danish talkers were elicited using the Diapix task in four conditions consisting of combinations of language (Danish vs. English) and noise background (quiet vs. ICRA 7 noise presented at 70 dBA). Overall, participants took longer to complete the task in both noise and in L2 indicating that both factors reduced communication efficiency. In contrast to our predictions, in the presence of noise, the median of the FTO distribution decreased by approximately 40 ms and the standard deviation decreased by approximately 60 ms. However, the average median duration of utterances increased by 40% in noise. These findings are consistent with talkers holding their turn for longer, which may allow more time for their own speech planning. Overall, the results suggest that talkers may prioritise maintaining social norms for turn-taking fluency when communicating in difficult environments.
Copyright (c) 2020 Anna Josefine Munch Sørensen, Michal Fereczkowski, Ewen N MacDonald
Rapid perceptual learning of time-compressed speech and the perception of natural fast speech in older adults with presbycusis
Older people, especially ones with age-related hearing loss (ARHL), often have difficulties understanding naturally fast speech (NFS). This difficulty has been attributed to both sensory and cognitive factors. We now ask if rapid perceptual learning, assessed with a time-compressed speech (TCS) task, also contributes to the perception of NFS in older adults with ARHL, while accounting for the potential contribution of other cognitive factors. 45 participants with and without experience with hearing-aids completed the study. Significant rapid perceptual learning of TCS occurred within the first 20 sentences. This learning was significantly and positively correlated with NFS perception. Additionally, the perception of NFS was positively associated with vocabulary and memory span and negatively correlated with hearing thresholds. We found no significant differences between experienced-users of hearing-aids and non-users. Findings suggest that declines in rapid perceptual learning may play a role in the perception of NFS in people with ARHL, which is additional to the contribution of other cognitive variables.
Copyright (c) 2020 Tali Rotman, Limor Lavie, Karen Banai
Feature-based audiovisual speech integration of multiple streams
Speech perception often involves the integration of auditory and visual information. This is shown in the McGurk effect, in which a visual utterance, e.g., /ipi/, dubbed onto an acoustic utterance, e.g., /iki/, produces a combination percept, e.g., /ipki/. However, it is still unclear how phonetic features are integrated audiovisually. Here, we studied audiovisual speech perception by decomposing the auditory component of McGurk combinations into two streams. We show that auditory /i_i/, where the underscore indicates an intersyllabic silence, dubbed onto visual /ipi/ produce a strong illusion of hearing /ipi/. We also show that adding an acoustic release burst to /i_i/ creates a percept of /iki/. An auditory continuum was created with stepwise temporal alignments of the release burst and /i_i/. When dubbed onto /ipi/, this continuum was perceived mostly as a visually driven response /ipi/ when the burst overlapped with either acoustic vowel. Other temporal alignments frequently produced combination responses. Mostly /ikpi/ combinations were obtained when the burst was closer to the initial vowel, and reverse /ipki/ responses when it was closer to the final vowel. These results are indicative of feature-based audiovisual integration where burst and aspiration are sufficient cues for the consonant /k/, while the perception of /p/ depends on place information in the visual stream.
Copyright (c) 2020 Juan Camilo Gil-Carvajal, Jean-Luc Schwartz, Torsten Dau, Tobias Søren Andersen
Auditory adaptation in real and virtual rooms
Walking from room to room in real listening conditions is a natural process in our everyday life and there is no obvious challenge for our auditory system to cope with. However, in experiments with virtual acoustic environments switching the virtual room or switching from real to virtual rooms can result in auditory confusions which can lead to in-head localization. This effect is known as the room divergence effect. A series of listening tests were conducted to verify this effect under different conditions as well as experiments which studied the effect of prior sound exposure and the time variant behaviour of it. In this paper two of these experiments are described and discussed. The first experiment shows that the extent of the room divergence effect depends on the room acoustics we have just learned. That indicates, that the room divergence effect is diminished during ongoing exposure to a specific room acoustic condition. The second listening test shows further evidence of this time-variant effect and we show that it can be suppressed by interrupting with the adaptation process. These tests raise the question why switching virtual rooms leads to temporary confusions but doing so with real rooms is unproblematic. Different theories are discussed in this publication.
Copyright (c) 2020 Florian Klein, Stephan Werner, Georg Götz, Karlheinz Brandenburg
Audio-visual sound localization in virtual reality
Virtual reality (VR) can be a strong research tool in audio-visual (AV) experiments. It allows us to investigate AV integration in complex and realistic settings. Here, using a VR setup-up in combination with a loudspeaker array, 16 normal-hearing participants were tested on their sound localization abilities. The virtual environment consisted of a 1:1 model of the experimental environment except with the loudspeaker array replaced by a ring. This ring indicated the height, but not the position of the loudspeakers. The visual component of the stimuli consisted of a ball falling and then bouncing once on the ring after which it disappeared. As the ball collided with the ring, an impact sound was played from a loudspeaker. Participants were asked to indicate the apparent sound origin, for both congruent and incongruent visual and audio spatial positions ranging from -30 to 30 degrees. The VR visual stimuli in combination with real auditory stimuli were capable of inducing AV integration. The range of this integration extended, for several participants, over large ranges of AV disparity compared to some earlier studies.
Copyright (c) 2020 Thirsa Huisman, Tobias Piechowiak, Torsten Dau, Ewen MacDonald
A method for evaluating audio-visual scene analysis in multi-talker environments
In cocktail-party environments, listeners are able to comprehend and localize multiple simultaneous talkers. With current virtual reality (VR) technology and virtual acoustics it has become possible to present an audio-visual cocktail-party in a controlled laboratory environment. A new continuous speech corpus with ten monologues from five female and five male talkers was designed and recorded. Each monologue contained a substantially different topic. Using an egocentric interaction method in VR, subjects were asked to label perceived talkers according to source position and content of speech, while varying the number of simultaneously presented talkers. With an increasing number of talkers, the subjects’ accuracy in performing this task was found to decrease. When more than six talkers were in a scene, the number of talkers was underestimated and the azimuth localization error increased. With this method, a new approach is presented to gauge listeners’ ability to analyze complex audio-visual scenes.
Copyright (c) 2020 Kasper Duemose Lund, Axel Ahrens, Torsten Dau
Investigating pupillometry as a reliable measure of individual’s listening effort
Pupillometry as a tool indicating listening effort has been extensively analyzed on a group level, but less is known about how reliable pupil dilation is as an indicator of an individual’s listening effort. The aim of this study was to investigate the reliability of the pupil dilation measured during a speech-in-noise task as an indicator of an individual’s listening effort. The pupil dilation of 27 normal-hearing (NH) and 24 hearing-impaired (HI) participants was recorded while they performed a speech-in-noise test on two different days. Measures of intraclass correlation coefficient (ICC) absolute agreement were considered in the analysis. The ICC was applied to the peak and mean pupil dilation as well as to the different terms resulting from fitting a third-order orthogonal polynomial within growth curve analysis (intercept, 1st order, 2nd order and 3rd order terms), which are assumed to provide further information about temporal changes of the pupil dilation. High values of test-retest reliability were found on some measures of the pupil response. Furthermore, a Bland-Altman analysis was applied as a graphical representation of the reliability of the pupillometry. The results showed different levels of reliability depending on the different features of the pupil response (slope, rise-fall and mean pupil dilation for the HI listeners; rise-fall, delay and mean pupil dilation for NH).
Copyright (c) 2020 Mihaela-Beatrice Neagu, Torsten Dau, Petteri Hyvärinen, Per Bækgaard, Thomas Lunner, Dorothea Wendt
Potential of self-conducted speech audiometry with smart speakers
Speech audiometry in noise based on matrix sentence tests is an important diagnostic tool to assess the speech reception threshold (SRT) of a subject, i.e., the signal-to-noise ratio corresponding to 50% intelligibility. Although the matrix test format allows for self-conducted measurements by applying a visual, closed response format, these tests are mostly performed in open response format with an experimenter entering the correct/incorrect responses (expert-conducted). Using automatic speech recognition (ASR) enables self- conducted measurements without the need of visual presentation of the response alternatives. A combination of these self-conducted measurement procedures with signal presentation via smart speakers could be used to assess individual speech intelligibility in an individual listening environ- ment. Therefore, this paper compares self-conducted SRT measurements using smart speakers with expert-conducted lab measurements. With smart speakers, the experimenter has no control over the absolute presentation level, mode of presentation (headphones vs. loudspeaker), potential errors from the automated response logging, and room acoustics. We present the differences between measurements in the lab and with a smart speaker for normal- hearing, mildly hearing-impaired and moderate hearing-impaired subjects in low, medium, and high reverberation.
Copyright (c) 2020 Jasper Ooster, Kirsten C. Wagener, Melanie Krueger, Jörg-Hendrik Bach, Bernd T Meyer
“Yes, I have experienced that!” – How daily life experiences may be harvested from new hearing aid users
Both auditory and non-auditory aspects of the rehabilitation process play a role in successful hearing aid uptake. The sound may be experienced differently in the clinic compared to daily life and the skills and knowledge related to HA use vary from patient to patient. The aim of the present study is to assess daily life experiences of new hearing aid users and to explore ways to utilize these assessments in a follow-up situation. The approach is based on online reporting, where the patients over a period of two months “swipe” through 453 possible experiences related to HA use. Seventeen patients volunteered to register experiences for a period of two months, and participated in a follow-up interview, where the registered data were presented. Results suggest that data can shed light on the development within various categories of hearing aid experience and promote reflection on the hearing rehabilitation process.
Copyright (c) 2020 Katja Lund, Rodrigo Ordoñez, Jens Bo Nielsen, Dorte Hammershøi
Speech related hearing aid benefit index derived from standardized self-reported questionnaire data
Speech understanding in noisy environments has been the most desired hearing-aid (HA) benefit sought by HA users. This paper examines the possibility of developing a speech-related HA benefit index from the speech- related questions in the self-reported questionnaire data. One question from Health-Related Quality of Life (HRQoL) instrument 15D and nine questions from the Speech, Spatial and Qualities of Hearing Scale (SSQ) having a direct implication to speech were selected for the analysis. After applying weights relevant to 15D, a delta of base-line (prior to HA fitting) and follow-up (two months after the initial fitting) responses to the selected questions were determined. A principal component analysis (PCA) was performed on the scaled and centered delta values. The resultant principal component scores were used to derive the composite index indicative of speech-related HA benefit.
Copyright (c) 2020 Sreeram Kaithali Narayanan, Tobias Piechowiak, Anne Wolff, Sabina S. Houmøller, Vijay Kumar Narne, Gérard Loquet, Dan Dupont Hougaard, Michael Gaihede, Jesper Hvass Schmidt, Dorte Hammershøi
Applicability and outcomes of a test for binaural phase sensitivity in elderly listeners
Interaural phase difference (IPD) discrimination in the binaural auditory system has been shown to be related to localization abilities and speech intelligibility in background noise. One of the tests for binaural phase sensitivity determines the highest frequency of the test tone for which an interaural phase difference of 180° is detectable (IPD-FR). This test was included in a test battery together with examination of visual and hearing abilities, balance, tactile- and motor-skills, and cognitive abilities. The IPD-FR test was conducted with an adaptive 3-AFC experiment starting with a test-tone frequency of 250 Hz. Sixty-five of 220 participants could not perform the IPD-FR task. The main predictors for inability to perform the IPD-FR task are hearing loss at 250 Hz, fluid intelligence, measurement number, and gender. A linear regression analysis revealed that the test result IPD-FR threshold is related to pure-tone thresholds at low frequencies, composite score of cognition, and composite score of tactile sensitivity, fine motor skills, and vision, as well as gender. A correlation analysis shows that the IPD-FR threshold is not related to speech recognition in non-dynamic listening conditions with speech from the front if low-frequency hearing loss is taken into account.
Copyright (c) 2020 Inga Holube, Theresa Nuesse, Olaf Strelcyk, Annaeus Wiltfang, Petra von Gablenz, Anne Schlueter
Task repetition influence on pupil response during encoding of auditory information in normal-hearing adults
Although numerous behavioural measures to estimate listening effort have been developed in recent years using free recall or dual-task paradigms, relatively little is known about physiological measures, such as pupil dilation, in response to cognitively demanding tasks. This study used a repeated-measure experimental design and aimed to investigate the cognitive resource allocation process of spoken words in an immediate free recall paradigm. Here, ten adults with normal hearing (NH) attended 2 days of trials with 14 trials per day. The listeners heard four-speaker babble noise along with seven sentences and then tried to remember the first words of all seven sentences. Recall performance on the first day only showed a significant serial position effect (p < 0.05). With increasing memory load imposed by the subsequent recall task, baseline pupil size significantly enlarged (p < 0.01), and the PPDs significantly decreased (p < 0.01) during the encoding process, implying that a gradual increase in resources allocated to memory capacity corresponded to a decline in resources allocated to listening. Real-time allocation of cognitive resources during the encoding of spoken words can be monitored independently by the analysis of pupil dilation averaged over multiple trials.
Copyright (c) 2020 Miseung Koo, Myung-Whan Suh, Jun Ho Lee, Seung-Ha Oh, Moo Kyun Park
Development of a Danish test material for assessing speech-in-noise reception in school-age children
For the audiological assessment of the speech-in-noise abilities of children with normal or impaired hearing, an appropriate test material is required. However, there is no standardized speech material for children in Denmark. The purpose of the current study was to develop a Danish sentence material suitable for school-age children. Based on the 600 test sentences from the Danish DAT corpus (Nielsen et al., 2014), 11 test lists comprising 20 sentences each were carefully compiled. These lists were evaluated in terms of their perceptual similarity and reliability with a group of 20 typically-developing normal-hearing children aged 6-12 yrs. Using stationary speech-shaped noise and diotic stimulus presentation, speech reception thresholds (SRTs) were measured twice per list and participant at two separate visits. The analyses showed that six test lists were perceptually equivalent. These lists are characterized by a grand average SRT of −2.5 dB SNR, a test-retest improvement of 0.4 dB, and a within-subject standard deviation of 1.1 dB SNR. The remaining test lists produced slightly higher SRTs but were generally also usable. Altogether, it is concluded that the developed test material is suited for assessing speech-in-noise reception in Danish school-age children.
Copyright (c) 2020 Shno Koiek, Jens Bo Nielsen, Laila Kjørbæk, Maria Baltzer Gormsen, Tobias Neher
Looking for objective correlates between tinnitus and cochlear synaptopathy
Tinnitus is the perception of a sound in the absence of acoustic stimulation. While usually connected to a hearing loss, there exists a subset of tinnitus sufferers with audiologically normal hearing, whose tinnitus was often initiated by a noise trauma. Noise-induced tinnitus might be connected to the noise exposure that leads to a permanent impairment of the hearing system without affecting sensitivity to sound. This is commonly referred to as hidden hearing loss (HHL) and might be connected to cochlear synaptopathy. The hypothesis that HHL is one of the causes underlying tinnitus is based on suppositions that both phenomena are related to deafferentation of auditory nerve fibres and related central gain adjustments. To investigate this connection, a screening procedure consisting of high frequency audiometry (HFA), tinnitus likeness spectrum and loudness, psychophysical tuning curves (PTC) and tinnitus masking curves (TMC), adaptive categorical loudness scaling, and middle-ear muscle reflex test was developed. Pilot results show that all measurements can be completed within a short time frame, due to a Bayesian procedure being adopted to measure HFA, PTC and TMC. These procedures may contribute to investigating the connection between tinnitus and HHL with a large number of outcome measures. This connection will provide important insights toward the development of better diagnoses and treatment methods.
Copyright (c) 2020 Chiara Casolani, James Michael Harte, Bastian Epp
Comparison of clinical feasibility of behavioural and physiological estimates of peripheral compression
Previous research has shown that the rate of peripheral compression (estimated from the slope of the basilar-membrane input-output function) is not correlated with the pure-tone sensitivity (the audiogram). However, efficient estimation of peripheral compression has proven challenging and the methods are based on several assumptions. The aim of this study was to investigate and compare results from three methods of estimating peripheral compression in terms of their accuracy and clinical feasibility. Two psychoacoustic behavioural measures, based on forward (temporal masking curves, TMC) and simultaneous masking with notched-noise (NN), were investigated together with a physiological, distortion-product otoacoustic- emissions (DPOAE), based measure. Forty-five hearing-impaired (HI) listeners with mild-to-moderate hearing loss were tested. Correlation analysis of the data was performed, including partial-correlations, in order to factor out the potential influence of the pure tone-thresholds on the compression estimates. The results demonstrated limitations of each of the considered methods; however, the experiment involving estimates of auditory filters showed good stability and small training requirements across the listeners.
Copyright (c) 2020 Michal Fereczkowski, Torsten Dau, Ewen Neale MacDonald
Characterizing the speech-in-noise abilities of school-age children with a history of middle-ear diseases
Recently, a number of studies have indicated that recurrent or chronic middle-ear disease during early childhood may lead to long-term supra-threshold hearing deficits. The current study followed up on this by investigating differences in monaural and binaural hearing abilities in noise among school-age children with or without a history of middle-ear diseases. Groups of children aged 6-12 years with either a history of recurrent otitis media (OM) with infection or effusion or without any previous ear diseases participated. All participants had normal middle-ear function and normal audiometric hearing thresholds at the time of testing. Measurements included monaural and binaural speech reception thresholds in the presence of stationary noise or competing speech. Sensitivity to binaural phase information was also assessed. Preliminary analyses based on the data from the first 31 participants suggest that, on average, OM children have poorer thresholds in conditions with binaural or spatial differences compared to children without any previous middle-ear problems. Follow-up analyses based on a larger dataset will substantiate these initial findings and relate them to information obtainable from the OM children’s medical records (e.g., age of onset or duration of conductive hearing loss).
Copyright (c) 2020 Shno Koiek, Jens Bo Nielsen, Christian Brandt, Jesper Hvass Schmidt, Tobias Neher
Analysis of a forward masking paradigm proposed to estimate cochlear compression using an auditory nerve model and signal detection theory
The healthy human auditory system has a large dynamic range. An “active mechanism”, presumably due to the electromotility of the outer hair cells in the cochlea, leads to level-dependent amplification of basilar membrane (BM) vibration and a compressive BM input/output function. Different methods for estimating this compressive function based on behavioural forward masking have been suggested. These methods are based on the assumption that BM processing can be isolated from the response of the overall system and that the forward masking onto the probe is different for on- and off-frequency maskers. In the present study, a computational model of the auditory nerve (AN) in combination with methods from signal detection theory was used to test these assumptions. The simulated AN response was quantified in terms of rate and synchrony for different AN fibre types. Contribution of different tonotopic regions to overall sensitivity to the stimuli were analysed. The results show that on- and off-frequency maskers produce similar forward masking onto the probe. The simulation results suggest that the estimate of compression based on the behavioural experiment cannot be derived from sensitivity at the level of the AN but requires additional contributions, consistent with physiological studies.
Copyright (c) 2020 Jens Thuren Lindahl, Gerard Encina-Llamas, Bastian Epp
Physiological correlates of masking release
Masking release is one example of auditory object segregation where the masked threshold of a target sound decreases in the presence of beneficial cues. Two such cues are comodulation and interaural phase disparity (IPD) underlying the phenomena of comodulation masking release (CMR) and binaural masking level difference (BMLD) respectively. While the effect of these cues have been shown in behavioral studies, little is known about the underlying physiological mechanisms of masking release. In this study, we postulated an ”internal signal-to-noise ratio (iSNR)” that reflects neuronal representation of a masked tone. As the proxy for iSNR, we investigated the applicability of late auditory evoked potentials (LAEPs). We added an onset asynchrony cue with comodulation and IPD cues. Results showed that onset asynchrony had a negative effect on CMR while it did not affect BMLD. The P2 component of the vertex LAEPs was suggested to be an objective measure of iSNR. This will provide us information about whether temporal contexts affect the neuronal representation of CMR and BMLD at the level of the auditory cortex.
Copyright (c) 2020 Hyojin Kim, Bastian Epp
A comparison of two measures of subcortical responses to ongoing speech: Preliminary results
Neural responses in the auditory brainstem and midbrain are traditionally obtained with repetitions of basic stimuli such as clicks and tones. However, two different methods to measure subcortical responses to ongoing speech with non-invasive electroencephalography (EEG) have recently been published: one based on regularised linear regression (Maddox and Lee, 2018), and the other based on cross-correlation (Etard et al., 2009; Forte et al., 2017). Here, we compare these two methods using the same EEG data set. For both measures, we found prominent peaks in the response functions at latencies consistent with wave V of the auditory brainstem response (ABR; mean latency: 8.19 and 5.97 ms, respectively). The peak response latencies in individual participants were correlated between the regression approach and conventional click-evoked auditory brainstem responses (click- ABRs), suggesting a common underlying neural source. However, similar correlations were not found between the two speech-based methods, nor between the correlation approach and click-ABRs. This could arise from either differences in the methodologies or from variability in the measures.
Copyright (c) 2020 Florine Lena Bachmann, Ewen MacDonald, Jens Hjortkjær
Provoking and minimising potentially destructive binaural stimulation effects in auditory steady-state response (ASSR) measurements
An aided sound-field auditory steady state response (ASSR) has the potential to be used to verify the quality of fit of hearing aids on infants. Each aided ear should ideally be tested independently, but it is suspected that binaural testing may be used by clinics to reduce test time. This study simulates ‘clinically conceivable’ dichotic ASSR sound-field conditions to examine the risk of making false judgements due to unchecked binaural effects. Unaided ASSRs were recorded with a clinical two channel EEG system for 15 normally hearing subjects using a three-band CE-ChirpTM stimulus. It was found that the noise corrected power of a response harmonic can be reduced by up to 10 dB by introducing large ITDs equal to half the time period of the stimulus envelope. This could lead to concluding that a hearing aid fitting is poor, even though the fitting would have passed separate monaural ASSR tests (false referral). No effect was detected for simulated lateralisations of the stimulus, which is beneficial for a proposed aided ASSR approach. Full-scalp ASSR recordings show distinct SNR reductions and topographical changes in response to the large ITDs, and demonstrate the vulnerability of ASSR to montage and inter-subject variation. Findings suggest that multi-harmonic detectors could make binaural measurements robust to artificial reductions of response harmonics cause by large ITDs.
Copyright (c) 2020 Sam David Watson, Søren Laugesen, Bastian Epp
Using response times to speech-in-noise to measure the influence of noise reduction on listening effort
Single microphone noise reduction (NR) can lead to a subjective benefit even when there is no objective improvement in speech intelligibility. A possible explanation lies in a reduction of listening effort. In a previous study, we showed that response times (a proxy for listening effort) to a simple arithmetic task with spoken digits in noise were reduced (i.e., improved) by NR for normal-hearing (NH) listeners. In the current study we complemented the data set with data from twelve hearing-impaired (HI) listeners, the target group for NR. Subjects were asked to add the first and third digit of a digit triplet in noise. Response times to this task were measured, subjective listening effort was rated, and speech intelligibility of the stimuli was tested. Stimuli were presented at three signal-to-noise ratios (SNR; -5, 0, +5 dB) and in quiet. Stimuli were either processed with ideal or non-ideal NR, or unprocessed. In contrast to the previous results with NH listeners, a significant effect of NR on response times was for HI listeners restricted to conditions where speech intelligibility was also affected (-5 dB SNR). We cannot confirm a positive effect on response times to speech-in-noise after applying NR for HI listeners.
Copyright (c) 2020 Ilja Reinten, Inge de Ronde-Brons, Maj van den Tillaart-Haverkate, Rolph Houben, Wouter Dreschler
Hearing examinations in Southern Denmark (HESD): Database description and preprocessing
All hearing examinations from the public health system of the Region of Southern Denmark have been electronically recorded from 1996 to 2018 and merged into a single database, named the Hearing Examinations of Southern Denmark (HESD) database. This database contains hearing information for more than 143,000 adults, totaling 271,575 valid pure-tone audiograms. The use of this dataset, however, needs to be preceded by an intensive preprocessing procedure in order for the data to be used for research purposes. This study is aimed at describing the HESD database, as well as the preprocessing steps and rules used to classify different types of hearing loss. An initial overview of the different types of hearing profiling and their distribution among our sample is also provided.
Copyright (c) 2020 Manuella Lech Cantuaria, Ellen Raben Pedersen, Mette Sørensen, Frans Boch Waldorff, Jesper Hvass Schmidt
Investigating the relationship between spectro-temporal modulation detection, aided speech perception, and directional noise reduction preference in hearing-impaired listeners
In analogy to the restoration of reduced audibility via hearing-aid amplification, supra-threshold speech processing deficits may be partially compensated for by using state-of-the-art directional noise reduction (NR) techniques. However, while amplification is usually prescribed based on classical audiometry, a clinical test that represents supra-threshold speech processing and is thus useful for prescribing NR settings is yet to be established. The present study explored the potential of a suitably adapted spectro-temporal modulation detection (STMD) test for this purpose by means of laboratory-based tests and field tests with 30 hearing-impaired participants. In particular, it was investigated whether STMD performance (i) predicts aided speech intelligibility measured in a spatial multi-talker set up with different degrees of NR and (ii) predicts preference for moderate vs. aggressive NR. STMD thresholds were strongly correlated with (i) speech scores measured without NR, (ii) speech intelligibility benefit induced by aggressive NR, and (iii) the individual participants’ NR preference. The latter relationship was mediated by performance in a reverse digit span task, which measures working memory capacity. Overall, the results suggest that a clinical test that assesses STMD sensitivity may be useful for prescribing NR settings in hearing-aid fitting.
Copyright (c) 2020 Johannes Zaar, Lisbeth Birkelund Simonsen, Thomas Behrens, Torsten Dau, Søren Laugesen
Effects of directional hearing aid processing and motivation on EEG responses to continuous noisy speech
Arguably, the next frontier in hearing aid (HA) development are devices that can infer (or learn) the needs of the user via non-invasive physiological measurements such as electroencephalography (EEG) and adjust themselves accordingly. A promising approach to translating EEG signals into HA control signals is the analysis of EEG impulse responses to running speech, as obtained by cross-correlating the audio stimulus with the concurrently recorded EEG signal. Here, we used this method for examining neural correlates of the effects of directional HA processing and listener motivation on speech comprehension in noise. Groups of older participants with normal or impaired hearing listened to audiobook material embedded in realistic cafeteria noise while their EEG was recorded using mobile hardware. A HA simulator was used for (dis)engaging a directional microphone setting and for providing amplification. Motivation was manipulated by offering a monetary reward for good speech comprehension in half of the trials. Motivation influenced the participants’ listening performance but not their EEG responses. Directional HA benefit, however, was reflected in both the behavioural and EEG data, thereby illustrating the potential of the tested approach with respect to enabling online HA control.
Copyright (c) 2020 Tobias Neher, Bojana Mirkovic, Stefan Debener
Adaptation to hearing-aid microphone modes in a dynamic localisation task
New technology can foster new ways of listening. A new hearing-aid programme can alter how we hear not only sources of sound but also their locations. While previous research has established how different hearing aid types and microphone modes affect static localisation ability, the current study explored the effects of introducing unfamiliar devices and microphone modes on dynamic localisation ability. Twelve experienced users of bilateral behind-the-ear (BTE) hearing aids oriented themselves to a target sound. Each trial consisted of 5-s segments of a target talker in a continuous background of far-field babble at the same overall level as the target. Targets were presented at either ±30, ±75 or ±120°. Head-orientation trajectories were measured with infra-red cameras. Participants first wore their own hearing aids for one block of 60 trials, then wore a new hearing aid and completed five more blocks in three different directional-microphone modes. In general, results showed trajectory differences between modes, and a modest influence of the preceding mode (i.e., adaptation). Three additional participants experienced with in-the-ear hearing aids oriented poorly with the new BTE device for the first two blocks, then returned to their baseline performance. This suggests that such a form-factor change requires additional time for spatial adaptation.
Copyright (c) 2020 William M Whitmer, Nadja Schinkel-Bielefeld
The vent effect in instant ear tips and its impact on the fitting of modern hearing aids
Today, approximately 70-80% of hearing aid fittings are made with instant ear tips. This may be due to ease of fit, improved physical comfort and the reduction in occlusion compared to custom earmolds. These tips can be completely open, vented, or closed. The acoustic properties of the ear tip depend on its type, size and the fit to the individual ear canal. Depending on the resulting real ear occluded gain (REOG) and vent effect (VE), the sound quality and aided benefit provided by the hearing aid may vary among individuals fitted with the same ear tip type. This study explored five Widex instant ear tips both in relation to REOG and VE using real ear measurements on 60 ears and in relation to subjective occlusion ratings. The results showed a large variation in REOG and VE both between ear tips and across subjects within the same ear tip type, and a high correlation between VE and perceived occlusion. These results imply that the acoustics of instant tips need to be assessed and considered as part of the hearing aid fitting process to ensure that fitting targets are matched.
Copyright (c) 2020 Sueli Aparecida Caporali, Jens Cubick
Individual hearing aid benefit: Ecological momentary assessment of hearing abilities
Questionnaires are often used to address the subjective perspective on hearing abilities in the course of hearing aid (HA) fitting. Weaknesses of this approach are, e.g., memory bias and possible mismatch of the pre-defined and individually experienced listening situations. Ecological momentary assessment (EMA) including in-situ surveys in real-life, could tackle these issues. We conducted an EMA study to examine how HA uptake changes the perception of everyday hearing abilities. In collaboration with local hearing aid acousticians, 16 first-time and follow-up HA wearers were recruited. They used the smartphone-based EMA system olMEGA for 3-4 full days before HA fitting and after HA acclimatization. This system allows for specifying situations and sound sources as well as for assessing hearing related dimensions like speech understanding and listening effort. Nine hundred thirty-three surveys out of a total of 1705 surveys related to speech listening events. Results showed a considerable individual variability regarding the type of reported events, the distribution and position of assessments. Overall, speech understanding improved by 1.1 scores and listening effort decreased by 1.3 scores on 7-point scales in post-intervention EMA compared to pre- intervention EMA.
Copyright (c) 2020 Petra von Gablenz, Ulrik Kowalk, Jörg Bitzer, Markus Meis, Inga Holube
Hearing-aid settings in connection to supra-threshold auditory processing deficits
Plomp (1986) described the consequences of hearing impairment in speech communication as the sum of two components: attenuation and distortion. Recent studies have shown that the sensitivity to spectro-temporal modulations (STM) might be linked to speech intelligibility in noise, suggesting that supra-threshold, or “internal”, distortions would affect both speech and STM perception similarly. Furthermore, reduced sensitivity to STM may also affect a listener’s preference for a hearing aid (HA) compensation strategy. Here, speech intelligibility and STM sensitivity were measured in 20 hearing-impaired (HI) listeners. One group of the listeners (Group A) showed an inability to detect STM, whereas the other listeners (Group B) exhibited similar thresholds as the control group with young normal-hearing (NH) listeners. The two HI groups participated in a perceptual evaluation experiment using multi-stimulus comparisons (MUSHRA). The audio files were processed by a HA simulator fitted to the individual hearing loss and the performance was rated in terms of four attributes: clarity, comfort, preference and listening effort. A correlation analysis showed that clarity and preference were correlated in Group A whereas comfort and listening effort were correlated in Group B. The classification of HI listeners in auditory profiles might be valuable for efficient HA fitting.
Copyright (c) 2020 Raul H Sanchez-Lopez, Torsten Dau, Morten Løve Jepsen
Hearing aid feature profiles: The success of rehabilitation
We recently developed a method to objectively classify hearing aids, using technical data (e.g., compression, noise reduction, etc.) from over 3900 different devices. This yielded hearing aid subgroups called ‘modalities’, that were characterized as distinct feature profiles, independent of manufacturer or type. Our present study aims to combine these objectively defined modalities with audiologically relevant rehabilitation needs, using data including audiological diagnostic tests and two questionnaires for subjective ratings. We investigated which hearing aid modalities contribute to successful rehabilitation results, and to which extent these modalities can be associated with specific rehabilitation needs. Our results indicate that more adjustable hearing feature channels or levels do not necessarily lead to better rehabilitation results.
Copyright (c) 2020 Simon E Lansbergen, MSc, Wouter A Dreschler, PhD
Using BEAR data to obtain reduced versions of the SSQ-12 and IOI-HA-7 questionnaires
The Speech, Spatial and Qualities of Hearing scale (SSQ-12) and the International Outcome Inventory for Hearing Aids (IOI-HA-7) are questionnaires containing 12 and 7 items, respectively. They are designed to subjectively assess hearing ability and are complementary to behavioral measures. Both questionnaires have been applied across a range of clinical and clinical research-related contexts, for example for assessing outcomes of e.g., cochlear implants and hearing aids. However, due to time constraints neither of the questionnaires seem to be an inherent part of standard clinical quality control. The Better Hearing Rehabilitation (BEAR) database contains SSQ- 12 and IOI-HA-7 scores of around 1600 subjects. Applying an Exploratory Factor Analysis (EFA) on the data from the 2nd visit allowed us to reduce the SSQ-12 to 5 questions and the IOI-HA to 3 remaining questions. The SSQ-5 explains 79% of the variance in the SSQ-12 data while the IOI-HA-3 accounts for 70% of the variance in the original IOI-HA-7. These new versions have the potential to be used more efficiently by shortening time and focusing on the items that are most effective to reflect individual benefit. Furthermore, the analysis seems to confirm the validity of such a reduction from similar findings in the literature that were done on different datasets.
Copyright (c) 2020 Tobias Piechowiak, David Zapala
Perceptual evaluation of six hearing-aid processing strategies from the perspective of auditory profiling: Insights from the BEAR project
The current study forms part of the Better hEAring Rehabilitation (BEAR) project, which aims at developing new clinical tools for characterizing individual hearing loss and for assessing hearing-aid (HA) benefit. Its purpose was to investigate potential interactions between four auditory profiles and three measures of HA outcome obtained for six HA processing strategies. Measurements were carried out in a realistic noise environment at signal-to-noise ratios that were set based on individual aided speech reception thresholds (SRT50). Speech recognition scores and ratings of overall quality and noise annoyance were collected in two spatial conditions. The stimuli were generated with the help of a HA simulator and presented via headphones to 60 older, habitual HA users who had previously been profiled based on a data-driven approach (Sanchez-Lopez et al., 2019). The four auditory profiles differed significantly in terms of mean aided SRT50 and interacted significantly with the HA processing strategies for speech recognition in one spatial condition. Moreover, the correlation-pattern between the speech recognition scores and subjective ratings differed among the auditory profiles.
Copyright (c) 2020 Mengfan Wu, Raul Sanchez Lopez, Mouhamad El-Haj-Ali, Silje Grini Nielsen, Michal Fereczkowski, Torsten Dau, Sébastien Santurette, Tobias Neher
Assessing daily-life benefit from hearing aid noise management: SSQ12 vs. ecological momentary assessment
In audiological research, assessing daily-life benefit from hearing aid (HA) noise management (NM) is a challenge. While ecological momentary assessment (EMA) using smartphone-connected HAs has recently emerged as a promising tool for real-life data acquisition, there is a lack of research linking this method to established ones such as the SSQ12 questionnaire. In the current study, 12 hearing-impaired participants were asked to assess two HA fittings using a well-known questionnaire and a smartphone-based EMA method combining soundscape logging with momentary self-reports. The two HA fittings differed in terms of their NM settings (no NM vs. cardioid microphones and noise reduction activated). The participants were aged 23- 75 years and had different occupations and lifestyles. The testing period for each fitting was 2 weeks. Overall, the EMA and SSQ12 scores were higher for the setting with NM activated, but this difference was only statistically significant in case of the SSQ12. The soundscape data showed that only few participants experienced noisy surroundings frequently. Future work on EMA-based HA assessment should therefore address the interplay between the tested HA features and the auditory ecology of the participants.
Copyright (c) 2020 Line Storm Andersen, Klaudia Edinger Andersson, Mengfan Wu, Niels Pontoppidan, Lars Bramsløw, Tobias Neher
Robust auditory profiling: Improved data-driven method and profile definitions for better hearing rehabilitation
Currently, the clinical characterization of hearing deficits for hearing-aid fitting is based on the pure-tone audiogram only. This relies on the assumption that the audiogram can predict performance in complex, supra-threshold tasks. Sanchez-Lopez et al. (2018) hypothesized that the hearing deficits of a given listener, both at threshold and supra-threshold levels, result from two independent types of auditory distortions. The authors performed a data-driven analysis of two large datasets with results from several tests, which led to the identification of four auditory profiles. However, the definition of the two types of distortion was challenged by differences between the two datasets in terms of the tests and listeners considered. In the Better hEAring Rehabilitation (BEAR) project, a new dataset was generated with the aim of overcoming these limitations. A heterogeneous group of listeners was tested using measures of speech intelligibility, loudness perception, binaural processing abilities and spectro-temporal resolution. As a consequence, the auditory profiles of Sanchez-Lopez et al. (2018) were refined. The updated auditory profiles, together with the investigation of optimal hearing-aid compensation strategies, are expected to form a solid basis for improved hearing-aid fitting.
Copyright (c) 2020 Raul H Sanchez-Lopez, Michal Fereczkowski, Tobias Neher, Sébastien Santurette, Torsten Dau
Subjective loudness ratings of vehicle noise with the hearing aid fitting methods NAL-NL2 and trueLOUDNESS
Subjects with similar hearing thresholds showed large differences in loudness summation of binaural broadband signals after narrowband loudness compensation. Based on these findings, the fitting method trueLOUDNESS was developed to restore the individual binaural broadband loudness perception. In the present study, the trueLOUDNESS fitting method was compared with NAL-NL2. Loudness judgements of different vehicles’ sounds were compared with average judgements by normal-hearing subjects. The loudness judgements with trueLOUDNESS fittings were closer to normal compared to the loudness judgements with NAL-NL2 fittings. This study shows that the lab measurement of binaural broadband loudness perception has validity beyond the laboratory.
Copyright (c) 2020 Dirk Oetting, Jörg-Hendrik Bach, Melanie Krueger, Matthias Vormann, Michael Schulte, Markus Meis
A word elicitation study including the development of scales characterizing aided listening experience
The purpose of the present study was to identify the terms hearing aid professionals and their patients use in the communication about the aided listening experience and develop scales that would help characterize this experience in the domain of corrective actions that a hearing care professional may apply. The study comprised a word elicitation task based on observations and interviews from consultations at the Aalborg University Hospital. The results were analyzed by developing an affinity diagram. The resulting 80 words were then sorted by three hearing professionals in a supervised card sorting session. The resulting attributes were included in a 63-point scale design, which (in a usability test including eight hearing-aid users) were considered easy to survey and use, but also including some redundancy and ambiguities. The results suggest that it is possible to develop scales based on the voluntary statements expressed during actual consultations, but it remains uncertain whether the expressions will be interpreted the same way by other patients and professionals.
Copyright (c) 2020 Dorte Hammershøi, Anne Wolff, Lykke J. Andersen, Rikke L. Mortensen, Mads D. Nielsen, Stefanie A. Larsen
Improving robustness of adaptive beamforming for hearing devices
Fixed beamforming for hearing aids is suboptimal due to mismatches in real-world situations between the assumed and encountered sound fields. Adaptive beamforming potentially provides better performance but may degrade it if the characteristics of the signal required by the design procedure are inaccurately estimated. This paper proposes a straightforward but sufficiently rich model for the sound field that can be used to increase the robustness of adaptive beamformer design. A method for estimating the model parameters is also presented. In reverberant acoustic conditions, the proposed method improves performance by > 1 dB even at −16 dB SNR, the lowest signal to noise ratio (SNR) tested. Furthermore, it is shown to be robust in a variety of acoustic conditions which do not conform to the sound field model, and to inaccurate steering of the array.
Copyright (c) 2020 Alastair Moore, Patrick Naylor, Mike Brookes
How to compare hearing-aid processing of real speech and a speech-modified stimulus for objective validation of hearing-aid fittings?
A method is proposed to evaluate whether a modern hearing aid with all automatic features enabled processes a speech-modified stimulus for objective validation of hearing-aid fittings as speech. The proposed method measures short-term coupler gains from brief snippets of steady-state probe noise, crossfaded into either the International Speech Test Signal (ISTS) or the speech-modified stimulus, which thus act as conditioning signals. For reference, the method is also applied to a steady-steady noise signal, which drives the hearing aids into noise mode. Results for a selection of hearing aids show that the method classifies the hearing aids’ mode of processing according to expectations, with all three conditioning signals.
Benefit from different beamforming schemes in bilateral hearing aid users: Do binaural hearing abilities matter?
Using a hearing aid simulator and virtual acoustics, Neher et al. (2017) recently showed that binaural hearing abilities influence speech-in-noise reception through different bilateral directional processing schemes. The current study aimed to extend this finding to real acoustic environments and commercial devices. Three beamforming schemes were tested – they differed in signal-to-noise ratio (SNR) improvement and binaural cue preservation. The participants were 38 elderly experienced hearing aid users. Speech understanding and localisation performance were measured. Binaural hearing abilities were assessed using the binaural intelligibility level difference (BILD). The analyses revealed a clear effect of the BILD on speech understanding in noise, but no interaction with the beamformer conditions. Greater SNR improvement was generally beneficial. In contrast, localisation of static and dynamic stimuli was more accurate when low-frequency binaural cues were preserved. Furthermore, the interaction with the BILD was marginally significant for dynamic stimuli (p = 0.054). Altogether, these results suggest that when selecting directional processing schemes in bilateral hearing aid fittings both speech understanding and aspects of spatial awareness perception should be considered.
Copyright (c) 2020 Matthias Latzel, Kirsten Christine Wagener, Matthias Vormann, Tobias Neher
The implementation of efficient hearing tests using machine learning
Time-efficient hearing tests are important in both clinical practice and research studies. Bayesian active learning (BAL) methods were first proposed in the 1990s. We developed BAL methods for measuring the audiogram, conducting notched-noise tests, determination of the edge frequency of a dead region (fe), and estimating equal-loudness contours. The methods all use a probabilistic model of the outcome, which can be classification (audible/inaudible), regression (loudness) or model parameters (fe, outer hair cell loss at fe). The stimulus parameters for the next trial (e.g. frequency, level) are chosen to yield maximum reduction in the uncertainty of the parameters of the probabilistic model. The approach reduced testing time by a factor of about 5 and, for some tests, yielded results on a continuous frequency scale. For example, auditory filter shapes can be estimated for centre frequencies from 500 to 4000 Hz in 20-30 minutes. The probabilistic modelling allows quantitative comparison of different methods. For audiogram determination, asking subjects to count the number of audible tones in a sequence with decreasing level was slightly more efficient than requiring Yes/No responses. Counting tones yielded higher variance for a single response, but this was offset by the higher information per trial.
Copyright (c) 2020 Josef Schlittenlacher, Richard E. Turner, Brian C. J. Moore
From derived-band envelope-following responses to individualized models of near- and supra-threshold hearing deficits
Auditory models which include frequency-dependent profiles of near and supra-threshold hearing deficits can aid the design of individualized hearing- aid algorithms. However, determining individual auditory-nerve (AN) fiber loss parameters is controversial as diagnostic metrics are presently based on auditory brainstem responses (ABRs) or envelope following responses (EFRs). These measures do not necessarily yield a frequency-specific quantification and might be affected by both outer-hair-cell and AN damage. We developed a derived-band EFR (DBEFR) metric to offer a frequency- specific assessment and complemented these with click-evoked otoacoustic emissions and audiometry. Cochlear-gain-loss profiles were derived from the latter measurements and inserted into individualized models, in which different synaptopathy profiles were introduced and DBEFRs simulated. Using a clustering technique, the best match between experimental and simulated synaptopathy profiles was determined and validated using the ABR data collected from the same listener. Results showed promise in offering a method to determine individualized sensorineural hearing-loss profile given a limited number of objective metrics.
Copyright (c) 2020 Sarineh Keshishzadeh, Sarah Verhulst
Neural health in cochlear implant users with ipsilateral residual hearing
Studies in cochlear implant (CI) users have shown a correlation between neural health and speech reception performance. Recently, electrically evoked compound action potentials (eCAP) with varying interphase gaps (IPG) have been used to estimate neural health. In the present study, we investigated eCAP characteristics in CI users with ipsilateral residual hearing (electric-acoustic stimulation, EAS). We hypothesized that neural health is better in apical areas in EAS users than in basal areas, due to increased hair cell survival. Amplitude growth functions (AGF) with varying IPGs of 2.1 and 10 μs were measured in 19 MED-EL Flex recipients with residual hearing. The eCAP characteristics slope, N1 latency and stimulus level at 50% maximum eCAP amplitude were investigated for the effect of IPG across electrode positions and were correlated to speech perception outcomes and duration of hearing loss. CI users without residual hearing were used as a control group to compare the patterns of slope, latency and 50% maximum amplitude between both IPGs. IPG showed a significant effect on the eCAP characteristics. The change in stimulus level for the 50% maximum amplitude showed a significant difference between electrode 1 and 3 as well as 1 and 4 in EAS users, maybe indicating impaired neural health in the medial region and validating the measurement in EAS users.
Copyright (c) 2020 Marina Imsiecke, Andreas Büchner, Thomas Lenarz, Waldo Nogueira
Towards unblinding the surgeons: Complex electrical impedance for electrode array insertion guidance in cochlear implantation
The complications during electrode array insertion in scala tympani for cochlear implantation may cause trauma, residual hearing loss and affect speech outcomes. The inner ear is like a black box for surgeons during the insertion process with no real-time feedback and must rely on radiation- based extraoperative imaging. Impedance measurement of electrodes during insertion is a simple yet effective method to assess array position. For this, an impedance meter has been designed which can measure magnitude (|Z|), phase (θ), real (R) and imaginary (Xc) parts of impedance. A switching circuit can sequentially scan all electrode pairs at regular intervals during insertion. An Evo straight electrode array is inserted in a transparent 2:1 scaled up 2D cochlear model (11 trials) filled with 0.9% saline using a 3-degrees-of- freedom actuation system. Bipolar impedance measurements of 8 pairs (40 samples each) are taken at regular intervals during 25 mm insertion at speed of 0.05mm/sec. A notable increase in |Z| and R is observed in the apical 3 electrode pairs when they first get in to contact with the lateral wall. At the same time, the phase gets less negative (more resistive impedance) and Xc increases (less capacitance). These results show that impedance can be used for electrode array localization in cochlea and impedance change due to electrode proximity to different materials can have application in other electrode implants.
Copyright (c) 2020 Nauman Hafeez, Xinli Du, Nikolaos Boulgouris, Philip Begg, Richard Irving, Chris Coulsan, Guillaume Tourrels
The next generation of audio intelligence: A survey-based perspective on improving audio analysis
Computer audition has made major progress over the past decades; however it is still far from achieving human-level hearing abilities. Imagine, for example, the sounds associated with putting a water glass onto a table. As humans, we would be able to roughly “hear” the material of the glass, the table, and perhaps even how full the glass is. Current machine listening approaches, on the other hand, would mainly recognise the event of “glass put onto a table”. In this context, this contribution aims to provide key insight into the already made remarkable advances in computer audition. It also identifies deficits in reaching human-like hearing abilities, such as in the given example. We summarise the state-of-the-art in traditional signal-processing-based audio pre-processing and feature representation, as well as automated learning such as by deep neural networks. This concerns, in particular, audio diarisation, source separation, understanding, but also ontologisation. Based on this, concluding avenues are given towards reaching the ambitious goal of “holistic human-parity” machine listening abilities – the next generation of audio intelligence.
Copyright (c) 2020 Björn Schuller, Shahin Amiriparian, Gil Keren, Alice Baird, Maximilian Schmitt, Nicholas Cummins
Prediction of speech intelligibility with DNN-based performance measures
In this paper, we present a speech intelligibility model based on automatic speech recognition (ASR) that combines phoneme probabilities obtained from a deep neural network and a performance measure that estimates the word error rate from these probabilities. In contrast to previous modeling approaches, this model does not require the clean speech reference or the exact word labels during test time, and therefore, less a priori information. The model is evaluated via the root mean squared error between the predicted and observed speech reception thresholds from eight normal-hearing listeners. The recognition task in both cases consists of identifying noisy words from a German matrix sentence test. The speech material was mixed with four noise maskers covering different types of modulation. The prediction performance is compared to four established models as well as to the ASR-model using word labels. The proposed model performs almost as well as the label-based model and produces more accurate predictions than the baseline models on average.
Copyright (c) 2020 Angel Mario Castro Martinez, Constantin Spille, Birger Kollmeier, Bernd T. Meyer
Evaluation of a notched-noise test on a mobile phone
The ability to conduct hearing tests and estimate auditory function at home or in the workplace can be useful for screening or longitudinal studies and allow the collection of diagnostic data from tests that are too time consuming to be feasible in the clinic. However, moving away from an acoustically controlled environment may influence the results of a test (e.g., through masking due to higher levels of background noise), increasing the uncertainty in the test measurements. In this study, 9 normal-hearing participants completed a notched noise masking experiment with 3 different experimental setups: in a psychoacoustic test booth with a standard laboratory PC; in a psychoacoustic test booth with a mobile device; and in a quiet office room with a mobile device. The accuracy and reliability of the mobile implementation was compared to results obtained with the laboratory setup. The effect of the test environment was investigated by comparing the mobile platform results between booth and office. The mobile device implementation corresponded well with the laboratory results for a notch width of zero, but showed a systematic bias when the width of the notch was increased. The reliability of the mobile implementation was comparable to the laboratory. Moving outside the sound-insulated booth did not affect the mobile platform results.
Copyright (c) 2020 Petteri Hyvärinen, Michal Fereczkowski, Ewen N MacDonald
Using a deep neural network to speed up a model of loudness for time-varying sounds
The “time-varying loudness (TVL)” model calculates “instantaneous loudness” every 1 ms, and this is used to generate predictions of short-term loudness, the loudness of a short segment of sound such as a word in a sentence, and of long-term loudness, the loudness of a longer segment of sound, such as a whole sentence. The calculation of instantaneous loudness is computationally intensive and real-time implementation of the TVL model is difficult. To speed up the computation, a deep neural network (DNN) has been trained to predict instantaneous loudness using a large database of speech sounds and artificial sounds (tones alone and tones in white or pink noise), with the predictions of the TVL model as a reference (providing the "correct" answer, specifically the loudness level in phons). A multilayer perceptron with three hidden layers was found to be sufficient, with more complex DNN architecture not yielding higher accuracy. After training, the deviations between the predictions of the TVL model and the predictions of the DNN were typically less than 0.5 phons, even for types of sounds that were not used for training (music, rain, animal sounds, washing machine). The DNN calculates instantaneous loudness over 100 times more quickly than the TVL model.
Copyright (c) 2020 Josef Schlittenlacher, Richard Turner, Brian C. J. Moore
Learning about perception of temporal fine structure by building audio codecs
The goal of audio coding is to efficiently describe an auditory experience while enabling a faithful reconstruction to the listener. The subjective quality compared to the original is measured by established psychoacoustic tests (BS.1116, 2015; BS.1534, 2015) and the description cost is measured in number of bits. As it is much cheaper to describe coarse scale signal properties than temporal fine structure (TFS), tools like noise fill, spectral extension, binaural cue coding, and machine learning have increased performance of audio codecs far beyond the first generation based on masking principles (e.g., mp3). In this evolution, implicit knowledge on hearing has been acquired by codec developers, but it has become increasingly difficult to construct tools to predict subjective quality. For example, it is yet unknown which aspects of the TFS that are essential for the listening impression to be preserved. To explore these issues, we study models of auditory representations with the mindset from audio coding. Given a method to solve the inverse problem of creating a signal with a specified representation, evaluating by listening can immediately reveal strengths and weaknesses of a candidate model.
Copyright (c) 2020 Lars Villemoes, Arijit Biswas, Heiko Purnhagen, Heidi-Maria Lehtonen
Computational investigation of visually guided learning of spatially aligned auditory maps in the colliculus
The development of spatially registered auditory maps in the external nucleus of the inferior colliculus in young owls and their maintenance in adult animals is visually guided and evolves dynamically. To investigate the underlying neural mechanisms of this process, we developed a model of stabilized neoHebbian correlative learning which is augmented by an eligibility signal and a temporal trace of activations. This 3-component learning algorithm facilitates stable, yet flexible, formation of spatially registered auditory space maps composed of conductance-based topographically organized neu- ral units. Spatially aligned maps are learned for visual and auditory input stimuli that arrive in temporal and spatial registration. The reliability of visual sensory inputs can be used to regulate the learning rate in the form of an eligibility trace. We show that by shifting visual sensory inputs at the onset of learning the topography of auditory space maps is shifted accordingly. Simulation results explain why a shift of auditory maps in mature animals is possible only if corrections are induced in small steps. We conclude that learning spatially aligned auditory maps is flexibly controlled by reliable visual sensory neurons and can be formalized by a biological plausible unsupervised learning mechanism.
Copyright (c) 2020 Timo Oess, Marc O. Ernst, Heiko Neumann
“Psychophysical” modulation transfer functions in a deep neural network trained for natural sound recognition
Representation of amplitude modulation (AM) has been characterized by neurophysiological and psychophysical modulation transfer functions (MTFs). Our recent computational study demonstrated that a deep neural network (DNN) trained for natural sound recognition serves as a good model for explaining the functional significance of neuronal MTFs derived physiologically. The present study addresses the question of whether the DNN can provide insights into AM-related human behaviours such as AM detectability. Specifically, we measured “psychophysical” MTFs in our previously developed DNN model. We presented to the DNN sinusoidally amplitude-modulated white noise with various AM rates, and quantified AM detectability as d′ derived from the model’s internal representations of modulated and non-modulated stimuli. The overall d′ increased along the layer cascade, with human-level detectability observed in the higher layers. In a given layer, the d′ tended to decrease with increasing AM rates and with decreasing AM depth, which is reminiscent of a psychophysical MTF. The results suggest that a DNN trained for natural sound recognition can serve as a model for understanding psychophysical AM detectability. Since our approach is not specific to AM, the present paradigm opens the possibility of exploring a broad range of auditory functions that can be evaluated by psychophysical experiments.
Copyright (c) 2020 Takuya Koumura, Hiroki Terashima, Shigeto Furukawa