ARCHES conference 2015 Audiological Research Centers in Europe (ARCHES) – network November 16 – 17, 2015 University Medical Center Groningen Hanzeplein 1, 9713 GZ Groningen Room: Ronde Zaal ARCHES 2015 – Groningen NL 1 Index Welcome 2 3 Program Monday 16 4 Program Tuesday 17 7 Summaries of oral presentations 9 Summaries of poster presentations 27 List of participants with email addresses 49 ARCHES 2015 – Groningen NL Welcome We are proud to present you the program for the 9th ARCHES meeting in Groningen. Through the years ARCHES has proven to be a valid platform of leading Audiology groups. And despite the lack of funding, ARCHES continues to be a platform for international exchange of ideas, scientific projects, and friendship. This year we will have: ‐
9 Lab Overviews: Short panoramic views of the work in the different labs (5 minutes)
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17 Oral Presentations: Current research presented in 15‐minutes presentations with 3 minutes of discussion time ‐
21 Posters: The highlights will be presented in short 1‐2 minutes talks, but there is ample time for discussion at the poster site. The venue is at the University Medical Center Groningen (UMCG), Hanzeplein 1, 9713 GZ Groningen. The guided city walk (1 hour) will start in front of “Het Feithhuis”, Martinikerkhof 10, Groningen, monday 19:00 hours. The dinner will also be in “Het Feithhuis”, Martinikerkhof 10, Groningen (tel +3150 313 5335), Monday 20:00 hours The organizers thank the Nadine Tuinman and the other members of the Groningen staff for their help in organizing this meeting. And we hope that there will be a lively exchange of ideas and information, also this year. Pim van Dijk Wouter Dreschler ARCHES 2015 – Groningen NL 3 Program Monday November 16 11.30 12.00 Welcome and registration (Ronde Zaal)
Lunch Session 1 (Ronde zaal) 13.00 Overview Astrid van Wieringen and Jan Wouters
Exp.ORL, Dept. Neurosciences, KU Leuven / ENT University Hospital Leuven 13.05 Overview Theo Goverts Audiology VU Medical Center, Amsterdam, The Netherlands Leuven B Amsterdam NL Speech perception in noise 13.10 Speech recognition in complex listening conditions: effects of age
Cas Smits, Wiepke Koopmans, Theo Goverts 13.28 Validation of a Dutch online speech in noise test for NIHL screening among noise‐exposed workers. Marya Sheik Rashid, Monique Leensen, Wouter Dreschler The FADE approach for predicting speech recognition and other psychoacoustical experiments with normal and hearing‐impaired listeners in a variety of conditions 13.46 Amsterdam NL Amsterdam NL Oldenburg D Marc R. Schädler, A. Warzybok, B. Kollmeier et al.: 14.04 Entrainment to temporal speech‐like fluctuations
Robin Gransier, Michael Hofmann, Marc Moonen, Astrid van Wieringen, Jan Wouters Poster highlights 14.22‐
15.00 21 short poster presentations (max 2 minutes)
15.00 ‐ 16.00 Pause and visits to the posters 4 ARCHES 2015 – Groningen NL Leuven B Posters P1 P2 A profiling system for the assessment of individual needs for rehabilitation with hearing aids. Gijs Hoskam, Inge Brons, Monique Boymans and Wouter Dreschler Amsterdam A comparison between the Dutch and American‐English digits‐in‐noise (DIN) test
Amsterdam Cas Smits, Charles Watson, Gary Kidd, David Moore, Theo Goverts P3 P4 NL Impact of background noise and sentence complexity on cognitive processing demands. Dorothea Wendt Copenhagen Analysis of second‐order modulation processing via sound texture synthesis
Copenhagen Richard McWalter P5 P6 P7 P8 P9 NL DK DK Effect of Frequency Allocation on Vocal Tract Length Perception in Cochlear Implant Users. Nawal El Boghdady, Deniz Başkent, Etienne Gaudrain Groningen How adolescents with cochlear implants perceive learning a second language – a pilot. Dorrit Enja Jung*, Anastasios Sarampalis, Deniz Baskent Groningen A comprehensive survey of the effects of hearing impairment and hearing aids on directional hearing. Michael A. Akeroyd and William M. Whitmer IHR Do individual differences in working memory predict speech‐in‐noise intelligibility in normal hearers? C. Füllgrabe and S. Rosen IHR How previous exposure to environmental noises can aid in maintaining speech intelligibility? Sofie Aspeslagh, D. Fraser Clark, Michael Akeroyd, Owen Brimijoin IHR NL NL UK UK UK P10 Unilateral hearing loss affects language and auditory development Leuven A. Sangen, L. Royackers, C. Desloovere, J. Wouters , A. van Wieringen B P11 Optimal volume settings of cochlear implants and hearing aids in bimodal users
Dimitar Spirrov, Bas van Dijk, Jan Wouters, Tom Francart P12 Template based CI artifact attenuation to measure Electrically Evoked ASSR
Hanne Deprez, Robin Gransier, Astrid van Wieringen, Jan Wouters and Marc Moonen P13 Age and objective measures of functional hearing abilities.
Hamish Innes‐Brown, Renee Tsongas, Colette McKay P14 Interrelations between ABR and EFR measures and their diagnostic power in targeting subcomponents of hearing loss. Anoop Jagadeesh and Sarah Verhulst Oldenburg Speech intelligibility and minimal rotational movements
Oldenburg P15 B P18 B B G G Aided Patient Performance Predictions (APPP) in realistic noise scenarios
Ernst, S.M.A, Völker, C. and Warzybok, A. P17 Leuven Leuven Jan Heeren, Giso Grimm, Volker Hohmann P16 Leuven Oldenburg G Source movement perception in normal and impaired hearing under different levels of acoustic complexity. Micha Lundbeck, Tobias Neher, Volker Hohmann Temporal Processing and Spatial Hearing in Elderly and Hearing Impaired Listeners
Andrew King, Kathryn Hopkins, Christopher J. Plack Oldenburg G Paris F P19 Interaction between AM and FM processing: Effects of age and hearing loss. Paris Nihaad Paraouty; Stephan D. Ewert; Nicolas Wallaert; Christian Lorenzi P20 Effects of age on AM and FM detection Nicolas Wallaert, Brian C. J. Moore, Christian Lorenzi P21 Speech recognition on neural data IHR Alban Levity, Christian J. Sumner, Stephen Coombes, Aristodemos Pnevmatikakis UK F Paris F ARCHES 2015 – Groningen NL 5 Monday November 16, part 2 Session 2 (Ronde zaal) 16.00 Overview Bastian Epp and Torsten Dau
Copenhagen
Hearing Systems, Department of Electrical Engineering, Techn. Univ. of Denmark 16.05 Overview Birger Kollmeier DK Oldenburg
Project Group Hearing, Speech and Audio Technology of the Fraunhofer IDMT, Medizinische Physik and Cluster of Excellence Hearing4all Universität Oldenburg. D Background knowledge for compensation strategies
16.10 Stimulus‐brain activity alignment between speech and EEG signals in cochlear implant users, more than an artifact? Groningen
NL Anita Wagner, Natasha Maurits & Deniz Bașkent 16.28 Functional modelling of neural interaural time difference coding with electric and acoustic stimulation Leuven B Andreas N. Prokopiou, Jan Wouters, Tom Francart 16.46 Perception of vocal‐tract length in cochlear implant users: can it be improved?
Etienne Gaudrain, Nawal El Boghdady, Deniz Başkent
Groningen
NL 17.04 Unilateral hearing loss. Tim Bost, Niek Versfeld, Theo Goverts NL 17.22 Development of a new test for determining binaural sensitivity to temporal fine structure IHR C. Füllgrabe, A.J. Harland, A.P. Sęk and B.C.J. Moore 17.40 – 18.45 Pause and visits to the posters 19.00 20.00 Start of a guided walking tour in city center of Groningen, followed with a dinner in het “Feithhuis” 6 ARCHES 2015 – Groningen NL Amsterdam
UK Tuesday November 17, part 3 Session 3 (Ronde zaal) 8.45 Overview Wouter Dreschler Amsterdam Clinical and Experimental Audiology AMC, Amsterdam, The Netherlands 8.50 Overview Christian Lorenzi NL Paris Laboratoire des Systèmes Perceptifs, CNRS / Dept d'Etudes Cognitives (DEC/IEC), Ecole normale supérieure, Paris, France. F Fitting hearing aids and CI’s 8.55 Dynamic Characterization of Noise Reduction in Hearing Aids
AMC Ilja Reinten, Inge de Ronde‐Brons, Rolph Houben, Wouter A. Dreschler 9.13 Characterizing and compensation of broadband binaural loudness perception in sensorineural hearing impaired listeners OLD Amsterdam NL Oldenburg G Dirk Oetting, Volker Hohmann, Jens‐E. Appell, Birger Kollmeier, and Stephan D. Ewert 9.31 Towards Physiologically based Coding Strategy for Cochlear implants
CH Sonia Tabibi, Andrea Kegel, Wai Kong Lai, and Norbert Dillier Zurich CH 9.48 The Gap Detection Test: Can it be used to diagnose tinnitus?
Groningen UMCG Kris Boyen, Deniz Başkent, Pim van Dijk NL Overview 10.05 Overview Pim van Dijk and Deniz Başkent
Groningen Otorhinolaryngology, University Medical Center Groningen and dept of Audiology, Research School of Behavioral and Cognitive Neuroscience, University Groningen NL 10.10 Pause and last opportunity to visits the posters 10.30 – 11.00 Optional: Lab visit and transfer to another lecture hall (Colloquium KNO) 11.00 Posters have to be removed and the program continues in Colloquium KNO ARCHES 2015 – Groningen NL 7 Tuesday November 17, part 4 Session 4 (Colloquium KNO) 11.00 11.05 Overview Michael Akeroyd IHR Institute for Hearing Research, UK UK Overview Norbert Dillier Laboratory of Experimental Audiology, ENT Department, University Hospital and University of Zurich, Switzerland Zurich CH Binaural signals and the environment 11.10 Evaluation of a dereverberation algorithm for CI recipients using a virtual acoustics environment Zurich CH Norbert Dillier, Patricia Bleiker, Andrea Kegel, Eleftheria Georganti, Dietmar Wohlbauer, Wai Kong Lai 11.28 The effect of spectro‐temporal context on computational speech segregation
Thomas Bentsen 11.45 Modelling binaural unmasking of speech in noise with dynamic IPDs Christopher F. Hauth and Thomas Brand 12.03 Effects of incongruent auditory and visual cues on sound externalization
Jens Cubick Final discussion 12.20 – 12.30 Evaluation of this conference Future of ARCHES 12.30 – 13.30 Lunch / end of meeting 8 ARCHES 2015 – Groningen NL Copenhagen DK Oldenburg G Copenhagen DK Oral presentations Audiological Research Centers in Europe"(ARCHES) – network ARCHES 2015 – Groningen NL 9 Speech recognition in complex listening conditions: effects of age Cas Smits, Wiepke Koopmans, Theo Goverts Department of Otolaryngology‐Head and Neck Surgery, Section Ear & Hearing and EMGO Institute for Health and Care Research, VU University Medical Center, Amsterdam, The Netherlands This study examined developmental effects for speech recognition in noise for normal‐hearing children in several listening conditions, using a test that minimizes the appeal on non‐auditory factors. In a first experiment a newly designed pediatric digits‐in‐noise test (pDIN‐test) was compared with the standard DIN‐test for children and adults. Both tests use the same basic stimuli. Compared to the DIN‐test, the pDIN‐test uses 79% instead of 50% as a target point, single digits instead of triplets, and animations in the test procedure in order to further minimize the appeal on working memory and attention. We found no significant difference between the score for the pDIN test and that of the DIN‐test for any age group. A second study with 112 children showed that the DIN test can be accurately and reliably performed by children aged 4 to 12 years. The results provide normative values, and show that speech recognition abilities develop well into adolescence: speech recognition scores improve with age, and children benefit less from binaural and temporal cues than adults. Young children need a more favorable signal to noise ratio than adults for all listening conditions. After correction for the baseline speech reception threshold in stationary noise, age effects on unmasking were smaller but still present. 10 ARCHES 2015 – Groningen NL Validation of a Dutch online speech in noise test for NIHL screening among noise‐exposed workers. Marya Sheikh Rashid1, Monique Leensen1, Jan de Laat2, Wouter A. Dreschler1 1 Academic Medical Center, Dept. of Clinical & Experimental Audiology, Amsterdam, NL LUMC – Leiden, the Netherlands 2
Background: In the Netherlands the internet‐based self‐screening test, the Occupational Earcheck (OEC), has been developed by the Dutch National Hearing Foundation, together with AMC and Leiden University. This test is an adaptive speech‐in‐noise test, specifically designed for the detection of occupational noise‐
induced hearing loss (NIHL). Different versions of an improved OEC (with alternative masking noises) have been evaluated in a controlled lab setting among 18 normal‐hearing listeners, and 15 listeners with different degrees of NIHL. The OEC with a low‐pass filtered stationary noise seemed to be the most appropriate test for the detection of NIHL, with a sensitivity and specificity both greater than 90%. Design: In the current cross‐sectional study, the OEC was validated at the work site in a quiet office‐like room, among a group of 100 noise‐exposed workers. SRT scores were compared to audiometric thresholds obtained by screening audiometry. Furthermore, test‐retest reliability, and sensitivity and specificity values were assessed. Results: SRT results of 87 noise‐exposed workers with normal hearing and different degrees of NIHL were analyzed. Correlation with PTA346 was 0.76. Test‐retest reliability in terms of standard error of measurement was 1.04. ICC was 0.62. Sensitivity and specificity values on monaural basis (ear‐level) were 84% and 82%, respectively. Sensitivity and specificity on binaural basis (individual‐level) was 90% and 73%, respectively. Conclusion: The improved OEC seems to be an appropriate test for occupational hearing screening purposes. However, there is some room for improvement. Retesting after a positive test result may reduce the false‐positive rate. ARCHES 2015 – Groningen NL 11 The FADE approach for predicting speech recognition and other psychoacoustical experiments with normal and hearing‐impaired listeners in a variety of conditions Marc R. Schädler, A. Warzybok, B. Kollmeier et al.: To predict the outcome of basic psychoacoustic experiments and Matrix sentence intelligibility tests, a framework based on an automatic speech recognition (ASR) system is used to simulate auditory discrimination experiments (FADE). Therefore, the ASR system is trained and tested over a wide range of levels/signal‐to‐noise ratios (SNRs) and the lowest achievable level/SNR is reported as the predicted outcome. Because the simulations require neither empirical data nor a‐priori knowledge beyond that of human listeners, the predicted outcomes are reference‐free and primarily constrained by the scarce information in the acoustic stimuli and the ability of the feature extraction to isolate that information. A common parameter set, suitable for a range of features and experiments, is determined with the aim of providing a universal parameter set that might possibly generalize well to other experiments. The broad scope, reference‐free outcomes, and a solid universal parameter set help to prevent an over‐
fitting of the model to specific aspects of sound perception by design. The considered experiments were tone‐in‐noise detection, and speech‐in‐noise recognition in a stationary and a fluctuating noise condition. Comparisons of predicted outcomes and empirical data show the predictive power of this reference‐free modeling approach. 12 ARCHES 2015 – Groningen NL Entrainment to temporal speech‐like fluctuations Robin Gransier1, Michael Hofmann1, Marc Moonen2, Astrid van Wieringen1, and Jan Wouters1 1
ExpORL, Dept. of Neurosciences, KU Leuven, Herestraat 49, bus 721, 3000 Leuven, Belgium 2
STADIUS, Dept. of Electrical Engineering (ESAT), KU Leuven, Kasteelpark Arenberg 10, 3001 Leuven, Belgium Over the last decade there has been a growing interest in the ability of the auditory system to entrain to temporal speech‐like fluctuations. These fluctuations are of particular interest as they contain important information for speech intelligibility. Temporal fluctuations resemble apart from words, syllables and phonemes. Syllabic rates peak roughly, across languages, between 2‐5 Hz, whereas phoneme rates have a broad distribution between ~8‐50 Hz. One of the interesting aspects of speech‐like fluctuations is that they coincide with brain oscillations, and in particular the delta/ theta and beta/gamma range oscillations for syllabic and phoneme rates, respectively. This coincide has resulted in several theories suggesting that the entrainment of brain oscillations to speech‐like fluctuations is an important feature of the auditory system to decode speech. Over the last decade electrophysiological studies have used real speech or modulated signals to gain insight in the ability of the auditory pathway to entrain to speech‐like fluctuations. However, limited attention has been paid to the auditory pathway’s response to the whole range of speech‐like temporal fluctuations, in particular the 0.5‐20 Hz range, and how reliable these entrained responses can be measured. In the present study we investigate the entrainment to speech‐like temporal fluctuations with EEG. A 70 dB SPL modulated one‐octave white noise band, centered at 1 kHz was used as a model for speech. In total, the entrained response to seventy distinct modulation frequencies between 0.5 and 100 Hz was investigated in twenty‐five normal hearing subjects. In addition, the intra‐subject variability of entrained responses to modulation frequencies within the 0.5‐20 Hz range was studied in ten subjects. At the meeting we will present and discus the results of the differences in activation patterns across modulation frequencies and subjects, and the repeatability of entrained responses to speech‐like fluctuations between 0.5‐20 Hz. ARCHES 2015 – Groningen NL 13 Stimulus‐brain activity alignment between speech and EEG signals in cochlear implant users, more than an artifact? Anita Wagner, Natasha Maurits & Deniz Bașkent1,2 1
University of Groningen, University Medical Center Groningen, Dept University of Groningen, Research School of Behavioral and Cognitive Neuroscience; Otorhinolaryngolog;, 2
A number of studies (e.g. Luo & Poeppel, 2007) suggest that synchronization between neural oscillations, as they are observed between different bands of electroencephalographic (EEG) signals, and the temporal amplitude modulations of speech is foundational to speech processing (Giraud & Poeppel, 2012). Speech intelligibility appears to vary with the strength of such alignment between the brain and acoustic signals. These studies are based on vocoded speech materials, and hence stimuli that are similar to the signal transmitted via cochlear implants (CI). EEG studies with CI users suffer from the presence of electrical artifacts induced by the device. This study investigates the phase alignment between EEG signals recorded with CI users and envelopes of natural sentence stimuli, and queries how much such alignment reflects brain signals engaged in speech processing or a CI induced artifact. EEG signals within the theta range (3‐8 Hz) of eight CI users with their own device, and eight normal hearing (NH) participants, recorded while they were listening to naturally spoken sentences, were compared in terms of their alignment with the signal envelopes. The analysis involved a cross‐correlation between the envelopes and the EEG channels to extract the lag between the signals. Coherence between aligned signals was then measured in terms of correlation and phase coherence. Preliminary results show for CI users correlations between 0.29 and 0.69, with higher values observed for channels close to the CI but also for contra‐lateral temporal and central channels. For NH listeners, correlations were found on left temporal and central channels (0.16‐0.33). The EEG signal lagged behind the speech signal for 120 ms to 380 ms for CI users, and for 200 till 450 ms for NH listeners. The correlation between speech envelopes and signals recorded in their respective trials was generally greater than the correlation found between EEG signals correlated with randomly chosen speech envelopes. Greater coherence between the speech signal and the channels in vicinity to the CI, together with the absence of coherence for these channels in NH listeners, suggest that signal transmitted via the device is at the source of this alignment. The question of whether greater coherence reported for NH listeners in studies with vocoded stimuli reflects the less natural amplitude modulations in these signals is currently being tested. The coherence found for the central channels for NH and CI listeners, however, suggests that this alignment may indeed reflect a step in speech processing. References Luo, H., & Poeppel, D. (2007). Phase patterns of neuronal responses reliably discriminate speech in human auditory cortex. Neuron, 54, 1001–1010. Giraud, AL & Poeppel, D. (2012). Cortical oscillations and speech processing: emerging computational principles and operations. Nature Neuroscience,15(4), 511‐517. 14 ARCHES 2015 – Groningen NL Functional neural modelling of just noticeable difference in interaural time detection for normal hearing and bilateral cochlear implant users. Andreas N. Prokopiou, Jan Wouters, Tom Francart Sound localisation is mediated by interaural time differences (ITD) and interaural level differences (ILD). Cochlear implant (CI) users of current clinical systems do not have access to ITD cues because of the loss of phase information after sound processing. This loss of phase is partly caused by the fixed period of the carrier pulse train. However, recent studies have shown that it is possible for bimodal and bilateral CI users to perceive ITDs from interaural phase shifts of the envelope of modulated pulse train stimuli, such as the ones generated by clinical processors. Furthermore the shape of the temporal envelope determines ITD sensitivity. This creates a wide parameter space which has not been fully explored. To aid in the study of this phenomenon a computational model was developed. It amalgamates existing work on acoustic stimulation and electrical stimulation of the auditory nerve with a novel neurometric‐psychometric comparison which aims at estimating the just noticeable difference (JND) in ITD for both normal hearing subjects and CI users. The acoustical and the electrical stimulation of the auditory nerve constitute the peripheral processing part of the model. In both cases the output is a train of action potentials generated by the auditory nerve. The model used for electrical stimulation is an already existing model, developed by Bruce et al. [IEEE Tr Biomed. Eng. 1999] and extended by Goldwyn et al. [J Neurophysiol 2012]. Temporal aspects of the generated spike train, such as neuron refractory period, adaptation and response latency to the electric pulse are well predicted by this model. The model used for acoustic stimulation is also an already existing model developed by Zilany et al. [J. Acoust. Soc. Am. 2014]. It is a cascade of phenomenological description of the major functional components of the auditory‐periphery, from the middle‐ear to the auditory nerve. The neurometric‐psychometric comparison developed takes as inputs the action potentials generated by the left and right auditory nerve stimulation models, either electrically or acoustically, thus permitting the modelling of normal hearing, bilateral CI and bimodal listeners' psychometric estimates, specifically the JND in ITD. This is especially useful, as estimated JND values can be validated against experimental investigations of the human binaural performance, and assess the model's performance. Furthermore the JND estimation can be directly used as a performance metric of novel temporal enhancement stimulation strategies, and as such the model can readily be applied as a test‐
bench for developing strategies for bimodal and bilateral CIs. ARCHES 2015 – Groningen NL 15 Perception of vocal‐tract length in cochlear implant users: can it be improved? Etienne Gaudrain, Nawal El Boghdady, Deniz Başkent1,2 1
University of Groningen, University Medical Center Groningen, Dept Otorhinolaryngolog;, University of Groningen, Research School of Behavioral and Cognitive Neuroscience; 2
When trying to understand a speaker among competing speakers, normal‐hearing listeners take advantage of differences in the voices of the talkers. However, cochlear implant (CI) users do not seem to benefit from such voice differences. Previous work from Patterson’s group in Cambridge allowed to isolate the acoustic cues that allow listeners to identify and segregate voices [e.g. Smith, Patterson, Turner, Kawahara and Irino, 2005, J. Acoust. Soc. Am. 117:305‐318]. In our group, we have extended these methods to CI listeners to fully characterize the limitations CI users experience in utilizing voice characteristics. More specifically, we have shown that while voice pitch (F0) perceived through the implant is sufficiently salient to allow gender categorization, the perception of vocal‐tract length (VTL) is severely degraded, leading to erroneous gender perception (Fuller et al., 2014. J. Assoc. Res. Otolaryn. 15:1037‐1048.). With the help of acoustic simulations mimicking the CI processing, the potential factors explaining this lack of sensitivity for VTL were explored. Poor spectral resolution, spectral quantization, and spectral distortion were all found to affect VTL just‐noticeable‐differences. These results have yielded a number of potential solutions to improve VTL representation — and hence concurrent speech perception — in the implant users. 16 ARCHES 2015 – Groningen NL The influence of peripheral hearing loss on speech recognition: listeners with Unilateral Hearing Loss Tim Bost, Niek Versfeld, Theo Goverts Department of Otolaryngology‐Head and Neck Surgery, Section Ear & Hearing and EMGO Institute for Health and Care Research, VU University Medical Center, Amsterdam, The Netherlands Impaired speech recognition performance is a major factor in the impact of hearing loss on participation in everyday life. Auditory factors, like audibility and supra‐threshold processing, play of course a major role in speech recognition performance in hearing impaired listeners. However, it is widely recognized that non‐
auditory factors like linguistic and cognitive skills also play a role. Furthermore, it is not always obvious to disentangle these factors in speech recognition experiments. We are primarily interested to know the specific influence of peripheral hearing loss on speech recognition and secondary whether this auditory factor can be described in terms of audibility alone or also in terms of supra‐threshold deficits. Here we present results of research were we studied speech recognition (CVCs and Digits in quiet and Digits in Noise at 65 and 80 dBA) in 19 listeners with unilateral sensorineural hearing loss (UHL) between 25 and 70 dB HL. Our basic assumption is that by comparing results between their normal ear (UHL+) and their impaired ear (UHL‐) we can isolate the effect of the peripheral hearing loss, because all non‐auditory factors are constant in each listener. Results show reduced speech recognition scores for UHL‐ on all tests, with a high correlation between the 3 tests. UHL+ scores were in the normal range for all listeners. Especially the results for the DIN at 80 dBA indicate that the peripheral hearing loss also causes supra‐
threshold deficits.
ARCHES 2015 – Groningen NL 17 Development of a new test for determining binaural sensitivity to temporal fine structure C. Füllgrabe1, A.J. Harland2, A.P. Sęk2,3 and B.C.J. Moore2 1
MRC Institute of Hearing Research, Nottingham, NG7 2RD, UK Department of Psychology, University of Cambridge, Downing Street, Cambridge, CB2 3EB, UK 3
Institute of Acoustics, Adam Mickiewicz University, 85 Umultowska, 61‐614 Poznan, Poland 2
Speech identification in noise has been shown to be associated with a listener’s sensitivity to the temporal‐
fine‐structure (TFS) of sounds. Both hearing loss and age (Hopkins & Moore, 2011; Füllgrabe, 2013) seem to adversely affect the ability to process TFS information. Hence, there has been keen interest in the development of tests that could be used in the clinic or in large‐scale research studies to assess sensitivity to TFS. Moore and colleagues designed two such tests: the monaural TFS1 test that requires listeners to discriminate complex harmonic tones from the same tones with all components shifted upwards by the same amount in Hertz (Moore & Sek, 2009); and the binaural TFS‐LF test that determines a listener’s detection threshold for an interaural phase difference (IPD) for pure tones of a given low frequency (Hopkins & Moore, 2010). However, several studies have reported that some listeners were unable to reliably perform the tests, so a graded measure of sensitivity to TFS could not be obtained for all listeners. To address this limitation, a new binaural TFS test was developed and validated, the so‐called TFS‐AF test (with AF standing for ‘adaptive frequency’). In this test, listeners are required to detect an IPD in pure tones but, in contrast to the TFS‐LF test, the IPD is fixed while the frequency of the pure tones is adaptively varied to determine the threshold. Normative data for a range of IPDs were obtained for young (19‐25 years) adult listeners with audiometrically normal hearing (≤ 20 dB HL). Results revealed thresholds between 950 and 1600 Hz for IPDs of at least 90º, and no evidence for improved sensitivity following protracted practice (Füllgrabe et al., 2015). In a follow‐up study, IPD sensitivity was assessed in older (> 60 years) adults with normal hearing to investigate if: (i) listeners can perform the TFS‐AF test even when performance on the TFS‐LF test is not possible, and (ii) familiarity with the test procedure and stimuli plays a greater role for listeners from an age range that is more representative of audiological patients. Results indicated that all older listeners could perform the TFS‐AF test and there were no or only small practice effects. The results suggest that IPD sensitivity is fairly stable across frequencies (up to, on average, 1300 and 1000 Hz for the young and older normal‐hearing listeners, respectively) and that the TFS‐AF test is suitable for the rapid assessment of TFS sensitivity in untrained listeners. References Füllgrabe, C. (2013). Age‐dependent changes in temporal‐fine‐structure processing in the absence of peripheral hearing loss. American Journal of Audiology, 22(2), 313‐315. Füllgrabe, C., Harland, A., Sek, A., & Moore, B. C. J. (2015). Development of a new method for determining binaural sensitvity to temporal fine structure. International Journal of Audiology, Submitted. Hopkins, K., & Moore, B. C. J. (2010). Development of a fast method for measuring sensitivity to temporal fine structure information at low frequencies. International Journal of Audiology, 49(12), 940‐946. 18 ARCHES 2015 – Groningen NL Dynamic Characterization of Noise Reduction in Hearing Aids Ilja Reinten, Inge de Ronde‐Brons, Rolph Houben, Wouter A. Dreschler Academic Medical Center, Dept. of Clinical & Experimental Audiology, Amsterdam, NL Hearing Aids (HA) are implemented with complex signal processing techniques among which single microphone Noise Reduction (NR). These algorithms serve to reduce unwanted background noise. NR algorithms should distinguish speech from noise and subsequently reduce gain in frequency bands where speech is absent and/or noise is dominant. Knowledge on the exact decision rules is important for the HA fitting process, but unfortunately most manufacturers do not reveal information on the implemented signal processing features. Recent research in speech intelligibility and listening comfort focusses mainly on temporal aspects of speech and noisy speech. These temporal aspects might also play an important part in successfully implementing a NR algorithm. Therefore, we are looking for a method to characterise the dynamic behaviour of NR algorithms. Previously, clarifying movies of spectral variations and spectrograms were introduced to illustrate the temporal behaviour of HA gain due to NR. Although these images are qualitative, they fail to quantize the results, which is an essential step towards an objective characterization of the dynamic behaviour. Therefore, a different approach was made for testing NR behaviour. A low frequency amplitude modulation spectrum of the HA gain gives an overview of the long‐term temporal behaviour of the NR algorithm. These spectra were analysed in several ways in search for a suitable measure to characterize the dynamic aspects of NR algorithms. Gain was determined as the difference in dB of processed noisy speech with NR either turned on or off. Preliminary results will be shown of measurements with an ideal artificial NR algorithm as well as recordings of hearing aid output signals after NR. ARCHES 2015 – Groningen NL 19 Characterizing and compensation of broadband binaural loudness perception in sensorineural hearing impaired listeners Dirk Oetting1,2, Volker Hohmann2, Jens‐E. Appell1, Birger Kollmeier1,2, Stephan D. Ewert2 1
Project Group Hearing, Speech and Audio Technology of the Fraunhofer IDMT, D‐26129 Oldenburg, Germany; Medizinische Physik and Cluster of Excellence Hearing4all Universität Oldenburg 2
Loudness is one of the key factors related to overall satisfaction with hearing‐aids. Sensorineural hearing loss typically results in steepened loudness function and a reduced dynamic range from elevated thresholds to uncomfortably loud levels for narrowband and broadband signals. In practice restoring the narrowband (NB) loudness perception in hearing‐impaired (HI) listeners can lead to overly loud perception of broadband (BB) signals and it is unclear how binaural presentation affects loudness perception in this case. Individual loudness functions can reliably be measured using categorical loudness scaling (CLS) without any training. Nevertheless, the use of loudness measurement like CLS is by far less common than use of audiometric thresholds to fit hearing aids, although loudness complaints are one of the most mentioned reasons for revisiting the hearing aid dispenser. A possible reason is that loudness measurements are typically conducted with monaural NB signals while binaural BB signals as speech or environmental sounds are typical in daily life. Here individual loudness perception of HI listeners was investigated with a focus on monaural and binaural broad‐band signals, as being more realistic compared to monaural narrow‐band signals. Nine normal‐
hearing listeners served as a reference group in this experiment. Ten hearing‐impaired listeners with similar audiograms were aided with a simulated hearing aid, adjusted to compensate the narrow‐band loudness perception back to normal. As desired, monaural NB UCLs were restored to normal, however large individual deviations of more than 30 dB were found for the binaural BB signal. The individual variations of the increased loudness summation cannot be explained by monaural narrowband measurements as the audiogram or the narrowband loudness functions. Results suggest that BB and binaural loudness measurements add key information about the individual hearing loss beyond the audiogram. Based on the above findings, two hearing aid algorithms were implemented taking into account loudness summation for binaural broadband signals. The processing path of both algorithms is based on the standard multiband compressor using the short‐time fast Fourier transform (FFT). Channel gains are initialized to restore the narrow‐band loudness functions and are modified in real‐time controlled by an additional analysis path. The analysis path estimates the bandwidth of the signal and how binaural the signal is (from monaural to binaural presentation). Based on these two estimates, the first algorithm modifies the input signal (input gain reduction, IGR), the second algorithm reduces the applied gains proportional to the current channel gains (proportional gain reduction, PGR). Both algorithms use the same BB and binaural conditions from CLS for individual fitting. Measurements with 15 HI listeners were conducted and compared to a standard multiband compressor and NAL‐NL2 fitting. Mean gain values for a 65‐dB speech shaped noise were similar to the gains prescribed by NAL‐NL2, however, large individual differences between about ‐12 and +16 dB existed, demonstrating limitations of monaural audiogram‐
based fitting. Acknowledgements: This work was funded by the Cluster of Excellence Hearing4all. 20 ARCHES 2015 – Groningen NL Towards Physiologically based Coding Strategy for Cochlear implants Sonia Tabibi1,2, Andrea Kegel2, Wai Kong Lai2, and Norbert Dillier2 1
Department of Information Technology and Electrical Engeneering, ETH Zurich, Switzerland; Laboratory of Experimental Audiology, ENT Department, University Hospital and University of Zurich, Switzerland. 2
Abstract In recent years, there is a tendency towards physiologically based coding strategies for cochlear implants (CIs) to make the stimulation pattern more realistic and natural; MP3000 strategy that is using a psychoacoustic‐masking model is one example [1]. In our bio‐inspired coding strategy, frequency decomposition of an audio signal was done by a gammatone filterbank which is estimated by reverse correlation functions of neural firing times. A 4th order all‐pole infinite impulse response (IIR) gammatone filterbank [2] was used and its frequency resolution was compared to the standard FFT filterbank implementation. For this purpose, a melody contour test with synthetic clarinet tones in octave 3 was carried out using 3 notes with 3 different intervals [3]. Pilot test results with normal hearing (NH) and CI listeners will be presented for both filterbanks. For the stimulation pattern which is applied via an electrode array, two important neural behaviors were taken into account: channel interaction and refractory properties. These two properties can be characterized by electrophysiological measurements of evoked compound action potential (ECAP). Exciting the neural population by an electrical pulse can change its response to the next pulse. This is largely ignored in frame based strategies such as Advanced Combinational Encoder (ACE). In addition to these spectral (channel interaction) and temporal (refractory) properties, another temporal phenomenon, facilitation is also considered. Animal model studies [4, 5] showed that a subthreshold masker pulse can alter the threshold for the probe pulse depending on the inter‐pulse interval. Thus, in the case of two subthreshold pulses which are close enough, the summation effect (facilitation) will happen which leads to a response to the second pulse. Facilitation can improve the ability of CI users to understand soft speech and leads to better neural representation of the temporal fluctuations or the rapid amplitude modulations in the speech signal. The procedure of implementation will be discussed and preliminary results will be shown in the form of electrodograms. References [1] A. B. Waldo Nogueira, Thomas Lenarz, and Bernd Edler, "A Psychoacoustic “NofM”‐ Type Speech Coding Strategy for Cochlear Implants," EURASIP Journal on Applied Signal Processing, pp. 3044‐3059, 2005. [2] V. Hohmann, "Frequency analysis and synthesis using a Gammatone filterbank " Acta Acustica United with Acustica, vol. 88, p. 10, 2002. [3] Q.‐J. F. John J. Galvin III, and Geraldine Nogaki, "Melodic Contour Identification by Cochlear Implant Listeners," Ear Hear, p. 33, 2007. [4] S. B. C. Dynes, "Discharge characteristics of auditory nerve fibers for pulsatile electrical stimuli," PhD, Massachusetts Institute of Technology, 1996. [5] D. J. S. Leon F. Heffer, James B. Fallon, Mark W. White, Robert K. Shepherd and Stephen J. O'Leary, "Examining the Auditory Nerve Fiber Response to High Rate Cochlear Implant Stimulation: Chronic Sensorineural Hearing Loss and Facilitation " Neurophysiology, vol. 104, pp. 3124‐3135, 2010. ARCHES 2015 – Groningen NL 21 The Gap Detection Test: Can it be used to diagnose tinnitus? Kris Boyen, Deniz Başkent1,2, Pim van Dijk 1
University of Groningen, University Medical Center Groningen, Dept Otorhinolaryngolog;, University of Groningen, Research School of Behavioral and Cognitive Neuroscience; 2
Animals with induced tinnitus showed difficulties in detecting silent gaps in sounds, suggesting that the tinnitus percept may be filling the gap. The main purpose of this study was to evaluate the applicability of this approach to detect tinnitus in human patients and to compare this to control subjects, matched in age and hearing loss, but without tinnitus. To determine the characteristics of the tinnitus, subjects matched an external sound to their perceived tinnitus in pitch and loudness. To determine the minimum detectable gap, a gap detection test was performed by each subject. In line with the literature, an improvement of the gap detection thresholds with increasing stimulus level was found. Interestingly, the tinnitus group did not display elevated gap thresholds in any of the stimuli. Moreover, there was no relation between gap detection performance and perceived tinnitus pitch. These findings show that tinnitus in humans has no effect on the ability to detect gaps in auditory stimuli. Thus, the present results indicate that a simple gap detection is not yet a suitable clinical tool to identify tinnitus. 22 ARCHES 2015 – Groningen NL Evaluation of a dereverberation algorithm for CI recipients using a virtual acoustics environment Norbert Dillier1, Patricia Bleiker2, Andrea Kegel1, Eleftheria Georganti1,3, Dietmar Wohlbauer1, Wai Kong Lai1 1
Laboratory of Experimental Audiology, ENT Department, University of Zurich, Switzerland Department of Information Technology and Electrical Engineering, ETH Zurich, Switzerland 3
Sonova AG, Stäfa, Switzerland 2
Reverberation and noise reduce speech intelligibility significantly and affect especially hearing impaired persons. Several denoising and dereverberation techniques have been developed in the past. The availability of wireless audio streaming options for hearing instruments and CI sound processors provides new options for binaural signal processing schemes in the future. The processing algorithm evaluated in this study consists of three steps: the denoising step, the removal of late reverberation parts and finally a general dereverberation stage based on computed coherence between the input signals at both ears. For the denoising part, a speech distortion weighted multi‐channel Wiener filter (SDW‐MWF) with an adaptable voice activity detector (VAD) is used in order to achieve an optimal trade‐off between noise reduction and speech signal distortion. In the second step a spectral subtraction filter is used in order to reduce late reverberation. Finally, a coherence filter is applied based on the assumption that the reverberated parts of a signal show a low coherence between the left and the right ear. In addition to the basic multi‐channel Wiener filter approach which attenuates low coherent signal parts, an adaptation with a non‐linear sigmoidal coherence to gain mapping is used. The performance of this denoising and dereverberation scheme was evaluated with common objective measures such as signal‐to‐noise ratio (SNR) and signal‐to‐reverberation ratio (SRR) as well as with the perceptual evaluation of speech quality (PESQ). In addition, speech in noise and localization tests in noise with two groups of listeners (normal hearing, NH; cochlear implant, CI) were performed. The virtual acoustics environment test setup used real life multimicrophone sound recordings which were reproduced through a 12 loudspeaker system using ambisonics processing. Reverberant speech was generated from dry recordings using a database of binaural room impulse responses. The dereverberation algorithm was implemented on a Speedgoat xPC Target realtime system which processed two input signals and generated two output signals. The input signals were obtained from hearing instrument microphones placed at the two ears of the subject which was seated in the center of the loudspeaker setup. The processed signals were presented to the two ears of the subjects either via headphones (for NH subjects) or via direct input into the CI sound processors (for CI recipients). Data collection for 10 NH and 10 CI subjects has been completed and results will be presented at the meeting. ARCHES 2015 – Groningen NL 23 The effect of spectro‐temporal context on computational speech segregation Thomas Bentsen Hearing Systems, Department of Electrical Engineering, Technical University of Denmark Computational speech segregation systems are highly relevant for many practical applications, amongst others noise reduction applications (e.g., in hearing aid algorithms) and as a front‐end for robust speech and speaker recognition for human‐computer interfaces. Promising results have been reported by combining auditory‐inspired features with a classification‐based back‐end. The feature extraction stage exploits logarithmically‐scaled amplitude modulation spectrogram (AMS) features to distinguish between speech and noise activity on the basis of individual time‐frequency (T‐F) units. Recently, it has been shown that the accuracy of speech segregation systems can be substantially improved by exploiting spectro‐
temporal context across adjacent T‐F units. Such a stage can be either realized in the feature extraction stage by so‐called delta features, or by exploiting the probability of speech activity across neighboring T‐F units in the classification back‐end. In this contribution, the role and the interaction of these two stages are compared and the influence on speech segregation performance is measured by a set of technical metrics and model‐based intelligibility predictions. 24 ARCHES 2015 – Groningen NL Modelling binaural unmasking of speech in noise with dynamic IPDs Christopher F. Hauth and Thomas Brand Medizinische Physik and Cluster of Excellence “Hearing4All”, Carl von Ossietzky Universität Oldenburg, 26129 Oldenburg, Germany Background Binaural speech intelligibility models have successfully been used to predict speech reception thresholds in scenarios, where the envelope of an interferer changed over time, but the binaural cues were stationary as the acoustical sources had a fixed position. In this study, the effect of an interfering noise with stationary envelope, but time‐varying interaural phase difference (IPD) on speech intelligibility is investigated. Methods Speech reception thresholds (SRT) of 50% correctly understood words are determined for 10 listeners with normal hearing using a sentence test. The speech is presented diotically via headphones together with a noise with the same long‐term frequency spectrum. The IPD of the noise is sinusoidally modulated with frequencies of 0.25, 0.5, 1, 2, 4, 8, 16, 32, and 64 Hz. The outcome of the experiment is predicted using a binaural speech intelligibility model (BSIM) (Beutelmann et al., 2010), which combines an Equalization‐
Cancellation (EC) front‐end with a speech intelligibility index (SII) back‐end. Based on this model, a modified model approach is introduced that estimates the EC parameters blindly from the mixed speech and noise signals. Both versions differ in their underlying assumptions: The original model has access to speech and noise separately and the EC stage is assumed as a top‐down mechanism that maximizes the SNR in each frequency band. Instead the blind model minimizes the overall level in each frequency band, which is equivalent to minimizing the dominating source. This assumption of a signal driven bottom‐up processing achieves a binaural unmasking if the SNR is negative. Results In the perceptual data, the largest binaural unmasking is observed for the slowest IPD modulation frequency, where SRTs are decreased by 3 dB compared to the diotic presentation of speech and noise. This unmasking gradually decreases with increasing modulation frequency and shows a flooring effect above a modulation frequency of 8Hz. The short‐time BSIM is not able to predict this increase. Therefore, a second time constant is introduced in the short‐time model: Additionally to the time‐constant used to account for amplitude modulated signals, an EC time constant simulating binaural “sluggishness” is introduced in order to estimate the EC parameters on a different time scale. By using a binaural window in the order of 100 ms, the decreasing binaural unmasking with increasing modulation frequency can be predicted by both the top‐down and bottom‐up approach. Conclusion Our results indicate that a separate time constant of binaural processing is required to account for effects related to dynamically changing binaural cues on speech intelligibility while keeping the ability of the model to account for interferers modulated in the envelope domain. ARCHES 2015 – Groningen NL 25 Effects of incongruent auditory and visual cues on sound externalization Jens Cubick, Hearing Systems, Department of Electrical Engineering, Technical University of Denmark In natural listening environments, sounds are generally perceived outside the listener's head, i.e., externalized. In contrast, sounds presented via headphones are often perceived inside the head (internalized). However, if headphone‐presented sounds are generated using the virtual auditory space (VAS) technique, where anechoic signals are being convolved with binaural room impulse responses, a very convincing simulation of the natural listening situation with externalized auditory images can be achieved. It has been reported though that the VAS technique works best in the same room where the BRIRs were measured, and that the perception of externalization is reduced, when the playback room differs from the recording room. It has been argued, that this reduction is due to the visual impression of the playback room, which might cause a certain expectation of the acoustics of the room. It has not, however, been tested, if it could be explained by the incongruity between the auditory perception of the recording and the playback room. This study investigated, whether the playback room does indeed affect the perceived externalization of sounds presented via headphones, and whether this is due to incongruent visual or auditory information. Individual BRIRs were recorded for 18 naive listeners in an IEC listening room. The listeners rated the perceived externalization in terms of perceived distance, azimuth angle, and compactness of the auditory image in a) the same IEC listening room, b) a small, reverberant room, and c) a large, anechoic room. The listeners were divided into two groups. One group had limited access to visual cues, the other group had no access to auditory information about the playback rooms during the experiment. It was found that the externalization percept was most stable in the IEC room, whereas especially the perceived distance was reduced in rooms b) and c). The ratings of localization and compactness, on the other hand, were largely independent of the room. Overall, incongruent stimuli reduced externalization, and this reduction was most pronounced when the mismatch between the playback room and the recording room occurred in the auditory modality. 26 ARCHES 2015 – Groningen NL Poster presentations Audiological Research Centers in Europe"(ARCHES) – network ARCHES 2015 – Groningen NL 27 P1: A profiling system for the assessment of individual needs for rehabilitation with hearing aids Gijs Hoskam, Inge Brons, Monique Boymans, Wouter Dreschler Academic Medical Center, Dept. of Clinical & Experimental Audiology, Amsterdam, NL A new profiling system has been developed for the reimbursement of hearing aids, based on individual profiles of compensation needs. The objective is to provide an adequate solution: a simple hearing aid when possible and a more complex aid when necessary. For this purpose we designed a model to estimate user profiles for Human Related Intended Use (HRIU). HRIU is based on self‐report data: a modified version of the AIADH, combined with a COSI‐approach. AIADH results determine the profile of disability and COSI results determine the profile of targets. The difference between these profiles can be interpreted as the profile of compensation needs: the HRIU profile. This approach yields an individual HRIU profile with scores on six dimensions: detection, speech in quiet, speech in noise, localization, focus, and noise tolerance. The HRIU‐profile is a potential means to determine the degree of complexity and/or sophistication of the hearing aid needed, that can be characterized by a Product Related Intended Use profile (PRIU). The PRIU is comprised from 52 scored features of the individual hearing aid. From these features the theoretical benefit (PRIU profile) is determined for the aforementioned six dimensions. Post‐fitting results show improvements in the 6 dimensions and determine whether the hearing aid is adequate. Also it provides well‐standardized data to evaluate the basic assumptions and to improve the system based on practice‐based evidence. This new approach will be highlighted and some first results will be presented. 28 ARCHES 2015 – Groningen NL P2: A comparison between the Dutch and American‐English digits‐in‐noise (DIN) test Cas Smits, Charles Watson, Gary Kidd, David Moore, Theo Goverts The present study investigated differences between Dutch Digits‐In‐Noise test (NL DIN) speech reception thresholds (SRTs) and US DIN SRTs for a group of native‐speaking Dutch listeners. A repeated‐measures design was used to compare SRTs for the NL DIN and US DIN in steady‐state noise and interrupted noise for monaural, diotic and dichotic listening conditions. Surprisingly, the US DIN SRTs were significantly better in monaural and diotic listening conditions than the NL DIN SRTs but only in interrupted noise. In a subsequent experiment a subset of these conditions with additional speech material (i.e., US DIN triplets without inter‐digit coarticulation/prosody) was used to determine if the these better SRTs can be explained by the combined effect of inter‐digit coarticulation and prosody in the English triplets. ARCHES 2015 – Groningen NL 29 P3: Impact of background noise and sentence complexity on cognitive processing demands Dorothea Wendt1,2 1
Hearing Systems, Department of Electrical Engineering, Technical University of Denmark Eriksholm Research Centre, Snekkersten, Denmark 2
Speech comprehension in adverse listening conditions requires cognitive processing demands. Processing demands typically increase with acoustically degraded speech but also depend on linguistic aspects of the speech signal, such as syntactic complexity. While it has been shown that both factors, i.e. acoustic degradation and sentence complexity, affect processing demands, only little is known about the interplay between these two factors when presented in combination. In the present study, pupil dilations were recorded in 20 normal‐hearing participants while processing sentences that were either syntactically simple (subject‐first sentence structure) or complex (object‐first sentence structure) and presented in either high or low level background noise. Pupil response was recorded (as a physiological correlate of processing effort) in an audio‐visual picture‐matching task. Furthermore, participants were asked to subjectively rate the perceived effort during the sentence comprehension task. The results indicated that the perceived effort was affected by the noise level, whereas the effect of the sentence complexity on subjectively rated effort was rather small. Conversely, pupil dilation increased with sentence complexity whereas only a small effect of the noise level on pupil dilation was observed. Thus, the results indicated that acoustic noise and linguistic complexity have distinctly different impact on pupil responses and rated effort. Finally, the results showed that speech comprehension performance decreased with more complex sentences. The results suggest that perceived effort appears to be strongly dependent on the noise level, and less on sentence complexity. Pupil dilations, in contrast, may reflect working memory processing that is required for speech comprehension but may not necessarily be perceived as effortful. In order to examine whether and to what extent processing demands are linked to working memory, both pupil dilation and perceived effort will in a next step be correlated with working memory capacity in the individual participants. 30 ARCHES 2015 – Groningen NL P4: Analysis of second‐order modulation processing via sound texture synthesis Richard McWalter Hearing Systems, Department of Electrical Engineering, Technical University of Denmark Sound textures, such as a waterfall or galloping horses, are composed of the superposition of many similar acoustic events. It has been suggested that the perception of sound textures depend on time‐averaged statistics measured from a standard model of the auditory system. However, the perception of sound textures containing more advanced amplitude modulations is relatively unstudied. Previous work by Lorenzi et al. 2001, Ewert et al. 2002 and Füllgrabe et al. 2005 revealed the sensitivity of the auditory system to second‐order modulations, and noted the rhythmic like perceptual qualities. Based on this work, an auditory model was developed to account for second‐order modulation masking data that consisted of three linear filter stages – frequency selective peripheral filtering, first‐ and second‐order modulation filtering – with inter‐stage envelope extraction. Texture statistics, including marginal moments and pair‐
wise correlations, were measured at the output of the peripheral, first‐ and second‐order filtering stages. The auditory model was integrated into a version of the sound texture synthesis system of McDermott and Simoncelli (2011). Synthetic sound textures were generated with graduated texture statistic groups at each stage for the model. Listeners evaluated the synthetic textures via an identification task to reveal the contribution of different statistic groups in texture perception. In each trial, listeners were presented with synthetic sound textures and selected a corresponding text descriptor from a list of five choices. Realistic synthetic textures could be generated from the statistics measured from a broad range of original sound textures using the extended texture synthesis system. Increased identification performance was observed for each stage of the model, and the inclusion of the second‐order modulation processing yielded the highest identification performance, approaching that of the original textures. The results suggest the sensitivity of the auditory system to second‐order modulations may contribute to sound texture perception. ARCHES 2015 – Groningen NL 31 P5: Effect of Frequency Allocation on Vocal Tract Length Perception in CI Users Nawal El Boghdady1,2, Deniz Başkent1,2, Etienne Gaudrain3,1,2 1
University of Groningen, University Medical Center Groningen, Dept Otorhinolaryngolog;, University of Groningen, Research School of Behavioral and Cognitive Neuroscience; 3
Lyon Neuroscience Research Center, CNRS UMR 5292, INSERM U1028, University Lyon 1, Lyon, France 2
Cochlear implant (CI) users experience tremendous difficulties in understanding speech in cocktail‐party settings. A necessary first step to understanding speech in these scenarios involves tracking the voice of the target speaker, which usually depends on the perception of speaker‐specific features. These can be characterized along two largely independent dimensions, namely the Glottal Pulse Rate (GPR), or voice pitch (F0), and the Vocal Tract Length (VTL), related to speaker size. Fuller et al., 20141 have shown that while normal hearing (NH) listeners can utilize both F0 and VTL cues to identify the gender of a speaker, CI users rely solely on F0 to perform the same task. One possible hypothesis for this is that, in the implant, VTL cues are lost due to spectral distortions caused by sub‐optimal frequency‐to‐electrode mapping. In the present study, the effect of varying frequency‐to‐electrode allocation on VTL perception was investigated using vocoder simulations of CIs. Twenty‐four normal hearing (NH) participants were tested. Stimuli consisted of triplets of Dutch consonant‐vowels (CVs) uttered by a female talker, manipulated only along the VTL dimension using STRAIGHT, while F0 was held constant. The VTL just noticeable differences (JNDs) were tracked using a 2‐down/1‐up adaptive procedure, both for positive and negative VTL differences, in a 3‐alternative‐forced‐choice task (3‐AFC). Positive VTL changes represent a male voice direction, while negative VTL changes point towards the child voice direction. Four different frequency allocation maps were implemented using a 16‐channel noise‐band vocoder: 1) a map based on the Greenwood formula, 2) a purely linear map, 3) a 16‐channel version of the Cochlear CI24 clinical map, and 4) an Advanced Bionics (AB) HiRes 90K frequency map. Vocoder synthesis filters were always distributed according to the Greenwood function, and were shifted to simulate a deep electrode array insertion depth of 21.5 mm and a shallow insertion depth of 18.5 mm, according to the specifications of the AB HiFocus Helix electrode. Results from this experiment show that there is a significant effect of frequency map on VTL thresholds, which vary according to the direction of the speaker. Specifically, the Greenwood map yields very similar JNDs for the two voice directions (child and male) tested compared to the other frequency maps, for which the two voice directions showed very different thresholds. However, the effect size of such an interaction was rather small. No significant effect was observed for the insertion depths simulated in this study. These results indicate that the frequency‐to‐electrode mapping may indeed be sub‐optimal in CI users, especially for extracting VTL cues from different voices. Hence, the presence of this effect will be further investigated in CI participants, whose JNDs will be measured while varying the frequency allocation map of a research processor. However, since the effect size of the interaction between frequency mapping and VTL JNDs was small in the NH sample, additional directions will be explored for the CI group, such as the effect of stimulation patterns and different coding strategies on VTL JNDS. Funding: The University Medical Center Groningen (UMCG), and Advanced Bionics. 1
Fuller, C. D., Gaudrain, E., Clarke, J. N., Galvin, J. J., Fu, Q.‐J., Free, R. H., & Başkent, D. (2014). Gender Categorization Is Abnormal in Cochlear Implant Users. Journal of the Association for Research in Otolaryngology, 15, 1037–1048. 32 ARCHES 2015 – Groningen NL P6: How adolescents with cochlear implants perceive learning a second language – a pilot Dorrit Enja Jung*1,2, Anastasios Sarampalis2,3, Deniz Baskent1,2 * Presenting author 1
University of Groningen, University Medical Center Groningen, Dept Otorhinolaryngology; University of Groningen, Research School of Behavioral and Cognitive Neurosciences; 3
University of Groningen, Faculty of Behavioral and Social Sciences, Dept Psychology 2
Mastering a second spoken language (L2), most importantly English, has direct advantages for adolescents from non‐English‐speaking countries, such as the Netherlands. To date, cochlear implant (CI)‐related factors and challenges for L2 learning after completion of native language acquisition have not been identified. We postulate that two CI‐related factors, sensory and cognitive, may limit the ability to learn a L2 (English) successively to the native language (Dutch). It would be of great interest to support L2 learning in implanted adolescents, as they could develop closer to their full academic potential. Also, they might be able to benefit from secondary effects regularly observed in speakers of multiple languages, for example, greater cognitive control. This project includes a two parts, a questionnaire study and an experimental study. We present preliminary results from the questionnaire study. This study aims at investigating the CI and non‐CI adolescents’ self‐perception regarding their English learning experience. The questionnaire battery will be administered to high school students (age 12 – 17 years) attending English classes. Three participant groups will be included in the final study: adolescents with pediatric cochlear implantation, hearing‐
impaired adolescents without implants, and normal‐hearing adolescents. The questionnaire battery will cover relevant domains for L2 learning, for example: language exposure, motivation to learn a second language, language background, attention and fatigue, and the role of hearing‐ and lip‐ reading abilities. We expect that adolescents with implants report little difficulties regarding reading, as well as grammar‐ and vocabulary acquisition. They are expected to report greater difficulties regarding understanding spoken communication and pronunciation of the second language. Implanted adolescents and hearing‐
impaired adolescents are likely to report greater importance of lip‐reading and of the amount of background noise for successful L2 learning than normal‐hearing peers. In the presented pilot data, at least one of two well‐functioning CI users show deficits in L2 learning outcomes (i.e., L2 proficiency) compared to the control. Acknowledgements The authors are supported by a Rosalind Franklin Fellowship from the University Medical Center Groningen, University of Groningen, and the VIDI Grant No. 016.096.397 from the Netherlands Organization for Scientific Research (NWO) and the Netherlands Organization for Health Research and Development (ZonMw). ARCHES 2015 – Groningen NL 33 P7: A comprehensive survey of the effects of hearing impairment and hearing aids on directional hearing Michael A. Akeroyd1 and William M. Whitmer2 1
MRC Institute of Hearing Research, Nottingham, UK MRC/CSO Institute of Hearing Research – Scottish Section, Glasgow, UK 2
We report here a comprehensive survey of the experiments that have been done since the 1980s on how accurate hearing‐impaired listeners are at determining the spatial direction of sound sources, presented either using loudspeakers or using virtual acoustics. To allow for across‐experiment comparisons RMS error was used as the common metric for all data; where a study did it was not reported empirical relationships were used to calculate it from what other data was reported. Data on ITDs or ILDs per se, width, or spatial release from masking were not considered. There is considerable variation across listeners and experiments, but, overall, the survey results demonstrate that the performance of hearing‐impaired listeners is somewhat worse than normal hearing listeners no matter what the direction. Performance is especially worse for changes in elevation or for sounds presented from the side. Normal‐hearing listeners made few front/back confusions in the surveyed experiments but hearing‐impaired listeners were particularly prone to them. Statistically significant effects of aided localization have been observed experimentally, but few of them are generalizable as they often occurred for just some source directions, stimuli, or groups of listeners. Across the survey, hearing aids do not improve performance – but equally they do not interfere that much. For example, in the domain of left/right directional accuracy, the across‐experiment mean difference between aided and unaided results was just 1 degree of RMS error. There is overall little effect of differences in hearing aid features or designs. The exception was unilateral fitting, which results in a substantial localization deficit of as much as 20 degrees of RMS error. Overall, there is no experimental evidence for a substantial, universal benefit from hearing aids to directional accuracy, but in some circumstances, especially unilateral fitting, hearing aids can give a substantial deficit. Acknowledgements Work supported by MRC (U135097131) and the Chief Scientist Office (Scotland). 34 ARCHES 2015 – Groningen NL P8: Do individual differences in working memory predict speech‐in‐noise intelligibility in normal hearers? C. Füllgrabe1 and S. Rosen2 1
MRC Institute of Hearing Research, Nottingham, NG7 2RD, UK UCL Speech, Hearing & Phonetic Sciences, London, WC1N 2PF, UK 2
With the advent of cognitive hearing science, increased attention has been given to individual differences in cognitive functioning and their explanatory power in accounting for inter‐listener variability in understanding speech in noise (SiN). The psychological construct that has received most interest is working memory (WM), representing the ability to simultaneously store and process information. Common lore and theoretical models assume that WM‐based processes subtend speech processing in adverse perceptual conditions, such as those associated with hearing loss and background noise. Empirical evidence confirms the association between WM capacity (WMC) and SiN identification in older hearing‐
impaired listeners. To assess whether WMC also plays a role when listeners without hearing loss process speech in acoustically adverse conditions, we surveyed published and unpublished studies in which the Reading‐Span test (a widely used measure of WMC) was administered in conjunction with a measure of SiN identification. The survey revealed little or no evidence for an association between WMC and SiN performance. We also analysed new data from 132 normal‐hearing participants sampled from across the adult lifespan (18 to 91 years), for a relationship between Reading‐Span scores and identification of matrix sentences in noise. Performance on both tasks declined with age, and correlated moderately even after controlling for the effects of age and audibility (r = 0.39, p ≤ 0.001, one‐tailed). However, separate analyses for different age groups revealed that the correlation was only significant for middle‐aged and older groups but not for the young participants (< 40 years). A possible explanation for this increasing cognitive involvement with age is the accumulation of age‐
related deficits in supra‐liminary auditory processing (e.g. sensitivity to temporal‐fine‐structure and temporal‐envelope cues; Füllgrabe, 2013; Füllgrabe et al., 2015), resulting in under‐defined and degraded internal representations of the speech signal, calling for WM‐based compensatory mechanisms to aid speech identification. References Füllgrabe, C. (2013). Age‐dependent changes in temporal‐fine‐structure processing in the absence of peripheral hearing loss. American Journal of Audiology, 22(2), 313‐315. Füllgrabe, C., Moore, B. C., & Stone, M. A. (2015). Age‐group differences in speech identification despite matched audiometrically normal hearing: contributions from auditory temporal processing and cognition. Frontiers in Aging Neuroscience, 6, 347. Acknowledgements Work supported by the Medical Research Council (grant number U135097130) and the Oticon Foundation (Denmark). ARCHES 2015 – Groningen NL 35 P9: How previous exposure to environmental noises can aid in maintaining speech intelligibility Sofie L.G. Aspeslagh1,2, D. Fraser Clark2, Michael A. Akeroyd3 and W. Owen Brimijoin1 1
MRC/CSO Institute of Hearing Research – Scottish Section, Glasgow, UK University of the West of Scotland, Paisley campus, UK 3
MRC Institute of Hearing Research, Nottingham University, UK 2
Listeners often face changes in their acoustic environment. Previous research has demonstrated that when listeners experience a sudden change in environment (both reverberatory and in background noise), they suffer a momentary drop in speech intelligibility that recovers rapidly to optimum performance again with experience in that new environment. In reality, however, it is unlikely that listeners will experience an instantaneous change in reverberatory properties, whereas an instantaneous change in background noise is more plausible. The purpose of the current research was to investigate how previous exposure to background noise can aid in maintaining speech intelligibility. In one experiment, we gave listeners some partial information about a noise before presenting it to measure whether prior exposure to features of a new noise stimulus could influence the drop in intelligibility listeners experience when confronted with it. Results demonstrated that if listeners were given no information (white noise), or only amplitude information about the test noise stimulus, they experienced a 25% drop in intelligibility upon the switch. When listeners were given spectral information they only experienced a 9% drop in intelligibility upon the switch. Thus listeners appear to be less capable of maintaining speech intelligibility when there is a change in the spectral content of background noise than when there is a change in the temporal content. These results suggest that if listeners use a model to represent background noise in order to aid with intelligibility, then the model may be more based on spectral information than on temporal information. In another experiment, we aimed to measure how long listeners retain this putative background noise model. In order to do this we introduced an intervening noise in the middle of a test noise. Results showed that if the intervening noise was short (<1 s), then listeners experienced little or no drop in intelligibility when the test noise resumed. If the intervening noise was 9 s long, however, then listeners experienced a drop in intelligibility when the test noise was reintroduced. These results suggest that when an intervening noise is short, listeners tend to retain their background noise model and as the intervening noise gets longer, listeners are more likely to discard it. Acknowledgements Work Supported by the MRC and CSO (Scotland) in conjunction with the University of the West of Scotland. 36 ARCHES 2015 – Groningen NL P10: Unilateral hearing loss affects language and auditory development A. Sangen1, L. Royackers2, C. Desloovere2, J. Wouters1, A. van Wieringen1 1
ExpORL, Dept. Neurosciences, KU Leuven Belgium ENT University Hospital Leuven Belgium 1
An increasing body of research suggests that children with unilateral hearing impairment lag behind with respect to their normal hearing peers. In view of possible interventions it is necessary to document their developmental outcomes. The aim of the present research is to examine auditory, linguistic and cognitive outcomes of children with unilateral hearing loss compared to those of age‐matched normal hearing children of similar age. A case–control study was carried out with 22 children with unilateral sensorineural hearing loss between 5 and 15 years of age and age‐matched normal hearing controls. Language, working memory, and speech in noise (presented to the good ear through headphones) were assessed by means of behavioral measures and aspects of hearing disability and academic performance by means of questionnaires. Our results show that children with unilateral hearing loss score comparably to children with normal hearing with regard to speech in noise presented to the good ear, working memory and morphological language abilities, but lag behind in expressive vocabulary and syntactic language skills. Furthermore, the speech, spatial and qualities of hearing questionnaire (SSQ) indicates that in daily life, the unilaterally hearing impaired children experience problems in spatial hearing and in understanding speech in noisy situations, and that the effort they have to put into listening and in understanding speech is considerably greater than in normal hearing children. Our data suggest early intervention for children with unilateral hearing loss to prevent speech language delays. Keywords: Unilateral hearing loss, children, speech‐ or language delay ARCHES 2015 – Groningen NL 37 P11: Optimal volume settings of cochlear implants and hearing aids in bimodal users Dimitar Spirrov, Bas van Dijk, Jan Wouters, Tom Francart ExpORL, Dept. Neurosciences, KU Leuven, Belgium The goal of our study was to find the optimal relation between the volume settings of a cochlear implant (CI) and a hearing aid (HA) so that the user's loudness is balanced. Bimodal users have a CI in one ear and a HA in the other, non‐implanted ear. This combination increases the speech understanding in noise compared to the CI only. However, there are differences between the two devices, caused by the stimulation mode (acoustic versus electric) and the processing algorithms. The devices are not made with combined use in mind. Often even similar parts like the compressors are not coordinated. This leads to an unbalanced sound perception that decreases user comfort and potentially limits the speech understanding benefit. When the bimodal user changes the volume (or sensitivity) of the CI or the volume of the HA, the loudness changes for that ear. However, the contralateral device does not adapt its volume accordingly. Furthermore, one step volume change in the CI will not have the same perceptual effect as one step volume change in the HA. This makes it difficult for the users to correctly set their volume controls on both sides. Therefore, an automatic setting procedure is needed. In this study we investigated the possibility to use loudness models and to compute a function that relates the volume controls of the two devices. In the HA, the overall gain is used as a volume control parameter. In the CI, either microphone sensitivity or the so‐called volume, which controls the amplitude of the electrical pulses, are used as volume control parameters. By using loudness models parameterized for individual subjects, a function to relate the HA overall gain with the HA caused loudness was calculated. Similarly, for the CI a second function was calculated to relate the sensitivity or volume with the loudness caused by the CI. In order to have a balanced sound percept the loudness caused by the HA and the CI have to be equal. Therefore, the first two functions were combined in a new function to relate the HA overall gain with CI sensitivity or volume. As an input signal to the models steady state noise filtered according to the international long‐term average speech spectrum (LTASS) was used. The obtained function was validated with loudness balancing experiments with bimodal users. The stimulus level at the CI side was fixed and the level at the HA side was varied by interleaved 1up/2down and 2up/1down adaptive procedures. Balancing results were achieved for different stimuli and stimulus levels. The balancing results of LTASS was used to determine the model parameters. The mean difference between the model prediction and the balancing results was 3 dB. Without the model prediction the mean difference was 6.2 dB. From previous studies it is known that just noticeable level differences for bimodal users are on average in the interval 1.5 to 3 dB. Therefore, the average bimodal users will perceive electrical and acoustical stimuli as equally loud. The achieved function can be used to link the volume control of the two devices, and/or to relate their step sizes. The automated volume control settings will lead to an important usability improvement for the bimodal users. Acknowledgements: We would like to acknowledge the support from the Agency for Innovation by Science and Technology in Flanders (IWT R&D 110722 and IWT Baekeland 140748). 38 ARCHES 2015 – Groningen NL P12: Template based CI artifact attenuation to measure Electrically Evoked Auditory Steady State Responses Hanne Deprez1,2, Robin Gransier2, Astrid van Wieringen2, Jan Wouters2, Marc Moonen1 1
STADIUS, Dept. of Electrical Engineering (ESAT), KU Leuven ExpORL, Dept. of Neurosciences, KU Leuven 2
Electrically evoked auditory steady‐state responses (EASSRs) are currently being investigated for objective cochlear implant fitting. EASSRs can be detected in the electro‐encephalogram (EEG) in response to periodic (modulated) pulse trains presented through the cochlear implant. However, the EEG is obscured by electrical artifacts caused by (1) the implant's radiofrequency link, and (2) the electrical stimulation pulses. These CI artifacts can also be present at the response frequency. Their characteristics are subject and stimulus dependent, and vary depending on the stimulation mode. In monopolar mode, the CI artifacts are larger in amplitude and longer in duration than in bipolar mode. CI artifacts can be removed by blanking which applies a linear interpolation over the duration of CI artifact. This method only works if the CI artifact duration is shorter than the interpulse interval, which is the case for low‐rate (<<500 pulses per second, pps) pulse trains, or stimulation in bipolar mode. We recently showed that the CI artifact can also be removed at contralateral recording electrodes for stimulation at 500 pps in monopolar mode. At ipsilateral recording electrodes or for stimulation at higher pulse rates, a different method is needed to remove the CI artifacts. We therefore developed a new method based on template subtraction. EASSRs were measured in eight subjects at sub‐threshold stimulation intensities and for stimulation at 500 pps in monopolar mode. Based on these recordings we constructed CI artifact templates for each subject in each recording electrode. We evaluated the method in two steps, by first verifying that the CI artifacts can be removed in the absence of a neural response and second verifying that artificially generated responses can be reliably reconstructed. (1) We subtracted a template, constructed based on a sub‐threshold recording, from the same recording. If the CI artifacts are correctly removed, we expect the detection rate to be about 5%, which is the significance level of the response detection test. With blanking, the obtained median false alarm rate was 69% (IQR=47%) which reduced to 4%(IQR=11%) after template subtraction. (2) We added a simulated response, with varying amplitude and phase, to the sub‐threshold recordings used above. The same template as in (1) was subtracted from the data. If the CI artifacts were correctly removed, we expect to obtain the correct amplitude and phase. The percentage of channels with correct amplitudes (POCA) was calculated for each subject. The median POCA over all subjects was 29% (IQR=44%) with blanking, and increased to 94% (IQR=12%) after template subtraction. Based on artificially generated data with recorded CI artifacts and simulated neural responses, we conclude that template subtraction is a promising method for CI artifact attenuation. Acknowledgements: Research was funded by the Research Foundation Flanders (G.0662.13) and a Ph.D. grant to the second author by the Agency for innovation by Science and Technology (IWT, 141243). ARCHES 2015 – Groningen NL 39 P13: Age and objective measures of functional hearing abilities. Hamish Innes‐Brown, Renee Tsongas, Colette McKay Bionics Institute Melbourne, Australia & ExpORL, Dept. Neurosciences, KU Leuven Belgium The hearing impaired often have difficulties understanding sounds in a noisy background. This ability relies on the capacity of the auditory system to process temporal information. In this study we examined relationships between age and sensitivity to temporal fine‐structure, brainstem encoding of harmonic and modulated sounds, and understanding speech in noise. Understanding these links will allow the detection of changes in functional hearing before permanent threshold shifts occur. We measured TFS sensitivity, brainstem responses and speech in noise performance in 34 adults aged from 18 to 60 years. Cross‐
correlating the stimulus waveforms and scalp‐recorded brainstem responses generated a simple measure of stimulus encoding accuracy. Speech‐in‐noise performance was negatively correlated with TFS sensitivity and age. TFS sensitivity was also positively correlated with stimulus encoding accuracy for the complex harmonic stimulus, while increasing age was associated with lower stimulus encoding accuracy for the modulated tone stimulus. The results show that even in a group of people with normal hearing, increasing age was associated with reduced speech understanding, reduced TFS sensitivity, and reduced stimulus encoding accuracy (for the modulated stimulus). People with good TFS sensitivity also generally had more faithful brainstem encoding of the harmonic tone. 40 ARCHES 2015 – Groningen NL P14: Interrelations between ABR and EFR measures and their diagnostic power in targeting subcomponents of hearing loss Anoop Jagadeesh and Sarah Verhulst Medizinische Physik and Cluster of Excellence “Hearing4all”, Dept. of Medical Physics and Acoustics, Oldenburg University Given the recent classification of sensorineural hearing loss in outer‐hair‐cell loss, and a temporal coding deficit due to auditory‐nerve fiber loss, this study evaluates how brainstem response measures can be more effectively used in the diagnostics of subcomponents of hearing loss. We studied the relationship between auditory‐brainstem response (ABR) and envelope‐following response (EFR) measures, and how they relate to threshold and compression (DPOAE) measures in listeners with normal to mild hearing losses. The relationships between the resulting click ABR wave‐I and V level‐series and EFRs to 75‐dB‐SPL broad‐
band and narrow‐band noises of different modulation depths indicate that the EFR strength‐vs‐
modulation‐depth‐reduction and ABR measures are likely to inform about different, but complementary aspects of hearing loss. Because ABR latency and strength correlated with each other, and the ABR latency‐
vs‐level slope with hearing thresholds, we suggest that cochlear spread of excitation, and to a lesser extent other effects such as neuropathy, is responsible for differences in the ABR measures across listeners. The EFR slope measure did not correlate with any other metric tested, and might reflect temporal coding (or other) aspects of hearing, irrespective of the degree of cochlear excitation changes due to outer‐hair‐cell loss. ARCHES 2015 – Groningen NL 41 P15: Speech intelligibility and minimal rotational movements Jan Heeren, Giso Grimm, Volker Hohmann Speech reception thresholds (SRT) measured using head related transfer functions (HRTF) and headphones are usually higher than SRTs measured with loudspeakers. This discrepancy is commonly assumed to be caused by dynamic binaural cues, which are introduced by minimal head movements in the loudspeaker condition. These cues are not present when headphones are used. Thus, the influence of dynamic binaural cues was investigated by testing diotic listening conditions against minimal movements. Two spatial configurations were tested (S0N0, S0N180) on headphones and loudspeakers. In both cases 11th order Ambisonics was used for audio panning. Movements were implemented as modulations of the nominal azimuths. These modulations were either in‐phase or anti‐phase for S and N and resulted in rotational or counter‐rotational movements. Results indicate that dynamic binaural cues lead to a release from masking, but they do not provide evidence for the stated assumption (funded by DFG FOR1732). 42 ARCHES 2015 – Groningen NL P16: Aided Patient Performance Predictions (APPP) in realistic noise scenarios Ernst, S.M.A, Völker, C. and Warzybok, A. Technical progress, both in the algorithmic processing as well as in signal presentation, offers an extensive repertoire of possibilities in today’s compensation of hearing impairment. One major drawback of this diversity, resulting in highly individualized fittings and optimizations of hearing aids, is the vastly increasing inherent complexity, which can only be countered by substantial staff and time resources. Individualized aided patient performance prediction (APPP), e.g. for different types of hearing devices or fittings, is a possible way to reduce complexity and lower costs. The aim of this study was to evaluate a selection of established instrumental measures, which might be candidates for an APPP‐test‐platform and compare them with a model approach able to account for individual loss in audibility and for binaural processing, i.e. the binaural speech intelligibility model (BSIM). All instrumental measures, i.e. iSNR, STOI, PESQ and BSIM, were aided with 11 signal pre‐processing strategies and tested in multitalker babble, cafeteria, and a single competing talker noise condition. The pre‐processing schemes comprised directional microphones, coherence filter, single channel noise reduction and binaural beamformer as well as combinations of them. The predictions were compared with speech reception thresholds (SRTs) of 10 normal‐hearing and 12 hearing‐impaired participants. To compensate for hearing loss a multiband compressor scheme was used, individually fitted following the NL1 procedure. In general, the computed correlations between the instrumental measures and the mean subjective data from this study were low and only significant (p<0.001) in the single competing talker noise condition. However, the model predictions using BSIM can explain up to 83% of the measured variance of the individual SRTs in the no‐pre‐processing condition, i.e. the speech reception of normal hearing as well as aided hearing‐impaired listeners without additional noise cancellation. Furthermore, the individualized model scheme was able to estimate the possible benefits of the noise reduction algorithms for a subset of the participants, resulting in an individual Kendall rank correlation coefficient of up to 0.93 in the single competing talker noise condition. However, at this stage of development, the model failed to give reliable predictions of signal processing benefits for each individual and all noise scenarios. Therefore, further model development is needed for accurate evaluation of different hearing aid algorithms or their settings. This should include individual listener characteristics others than the audiogram. ARCHES 2015 – Groningen NL 43 P17: Source movement perception in normal and impaired hearing under different levels of acoustic complexity Micha Lundbeck, Tobias Neher, Volker Hohmann So far, very little is known about the perception of spatially dynamic sounds, especially under more complex acoustic conditions. Therefore, this study investigated the influence of reverberation and the number of concurrent sources on movement perception of listeners with normal and impaired hearing. Virtual listening environments were simulated with the help of a higher‐order Ambisonics‐based system that allows rendering complex scenarios with high physical accuracy. Natural environmental sounds were used as the stimuli. Both radial (near‐far) and angular (left‐right) movement perception were considered. The complexity of the scenarios was varied by adding stationary sound sources as well as reverberation. As expected, hearing‐impaired listeners were less sensitive to source movements than normal‐hearing listeners, but only for the more complex acoustic conditions. Furthermore, adding sound sources generally resulted in reduced sensitivity to both angular and radial source movements. Reverberation influenced only radial movement detection, for which elevated thresholds were observed. Altogether, these results illustrate the basic utility of the developed test setup for studying factors related to spatial awareness perception. 44 ARCHES 2015 – Groningen NL P18: Temporal Processing and Spatial Hearing in Elderly and Hearing Impaired Listeners Andrew King, Kathryn Hopkins, Christopher J. Plack Audiology and Deafness Research Group, School of Psychological Sciences, University of Manchester, Manchester, M13 9PL. UK Study 1: The effects of age and absolute threshold on envelope and temporal fine structure (TFS) interaural phase difference (IPD) processing were tested by measuring IPD discrimination thresholds from 46 listeners varying from 18 to 83 years old. Amplitude‐modulated (AM) tones were used so that the IPDs were manipulated either in the envelope or the TFS. A modulation rate of 20 Hz was used with carriers of 250 and 500 Hz. At these carrier frequencies, the listeners had absolute thresholds ranging from −1 to 73 dB SPL. Absolute threshold correlated with TFS‐IPD thresholds, but not envelope‐IPD thresholds, even when age was partialled out. Age was correlated moderately with envelope‐IPD thresholds and weakly with TFS‐IPD thresholds, even when absolute threshold was partialled out. Study 2: These IPD results were compared to frequency following responses (FFR) to AM tones modulated at 16, 27, 115 and 145 Hz with carriers of 307, 537, 357 and 578 Hz respectively. Phase coherence in the FFR reflecting phase locking to the envelope and TFS was derived. Absolute threshold did not relate to FFR. Phase coherence in TFS FFR and envelope FFR at 145 Hz deteriorated with age, irrespective of absolute threshold. Correlations between FFR and IPD thresholds were weak and not significant when age was accounted for. Age was still correlated with IPD thresholds when FFR was accounted for. Study 3: To determine whether disruption of IPDs in speech affected speech understanding, spatial release from masking (SRM) in a speech‐on‐speech task (Dantale II) was measured for 20 older listeners (aged 64 to 86) with bilateral, gently‐sloping hearing loss. A Master Hearing Aid system was used to either amplify the signal at the listener’s ear, or amplify and tone‐vocode the signal in phase across the ears (removing the IPDs). SRM was significantly greater without vocoding than with vocoding. Discrimination thresholds for IPDs in pure tones and for frequency shifts in harmonic and inharmonic complexes were also measured. Individual differences in the effect size of vocoding on SRM were related to audiometric thresholds at low frequencies and discrimination of frequency shifts in inharmonic complexes, but not to IPD discrimination. At the Ecole Normale Supérieure I shall turn my focus to investigating whether low‐rate frequency modulation (FM) detection is partly due to phase locking or whether FM is simply converted into AM at the auditory filter output. ARCHES 2015 – Groningen NL 45 P19: Interaction between AM and FM processing: Effects of age and hearing loss. Nihaad Paraouty1; Stephan D. Ewert2; Nicolas Wallaert1; Christian Lorenzi1 1
Ecole Normale Superieure, Paris, LSP UMR 8248, France; Medizinische Physik and Cluster of Excellence Hearing4All, Universität Oldenburg, 26111 Oldenburg 2
Background Many psychophysical studies have attempted to disentangle the roles of temporal‐envelope (ENV) and temporal fine‐structure (TFS) cues in frequency modulation (FM) detection. Interference effects between amplitude modulation (AM) and FM have been used to assess the relative strengths of these cues in FM detection. These studies measured FM detection thresholds with and without an interfering AM intended to disrupt ENV cues resulting from cochlear filtering. The current study extends this interference paradigm by systematically assessing the effects of interfering AM on FM detection and conversely, interfering FM on AM detection. Methods FM and AM detection thresholds were measured at 40 dB sensation level for 3 groups of listeners: young normal‐hearing (<30 years), older normal‐hearing (40‐65 years) and older hearing‐impaired listeners (40‐65 years; mild‐moderate hearing loss at 500 Hz) for a carrier frequency of 500 Hz and a modulation rate of 5 Hz. In the AM detection task, FM at the same rate as the AM (5 Hz) was superimposed at varying FM depths. In the FM detection task, a 5 Hz‐AM was superimposed at varying AM depths. Frequency selectivity was also assessed for each listener using the notched‐noise masking method. The thresholds for detecting a 500‐Hz pure tone in a 600‐Hz wide notched‐noise centered at 500 Hz were measured using three spectral notch widths: 0, 150 and 300 Hz. A simple model which used the output of an ENV‐processing pathway was developed in order to support the behavioural data. Results The data shows clear perceptual interference effects between AM and FM. AM detection was degraded by interfering FM similarly across the 3 groups of listeners. Hence, the interference effect measured this way seems to be independent of age and hearing loss. FM detection was deteriorated by interfering AM for all groups of listeners and was also globally degraded by age and hearing loss. This interference effect was stronger in the hearing‐impaired group. Conclusion The relative roles of ENV and TFS cues in modulation detection, as well as the effects of age and hearing loss on FM detection will be discussed in light of behavioural and modelled data. Acknowledgements Work supported by ANR‐Heart, ANR‐11‐0001‐02 PSL, ANR‐10‐LABX‐0087 & Entendre‐SAS. 46 ARCHES 2015 – Groningen NL P20: Effects of age on AM and FM detection Nicolas Wallaert1, Brian C. J. Moore2, Christian Lorenzi1 1
Ecole Normale Superieure, Paris, LSP UMR 8248, France; Department of Experimental Psychology, University of Cambridge Downing street, Cambridge CB2 3EB, United Kingdom 2
Abstract Amplitude‐modulation (AM) and frequency‐modulation (FM) detection thresholds were measured for a carrier frequency of 500 Hz and modulation rates of 2 and 20 Hz for young and older listeners with normal absolute thresholds below 3 kHz. FM detection thresholds were measured in the presence of uninformative AM in both intervals of a forced‐choice trial, to disrupt the use of excitation‐pattern cues. The number of modulation cycles ranged from 2 to 9. The results show that for both groups and for each number of modulation cycles, AM and FM detection thresholds were lower for the 2‐Hz than for the 20‐Hz rate. For both groups, AM and FM detection thresholds decreased with increasing number of modulation cycles, this effect being greater for AM than FM. Thresholds were higher for older than for younger listeners, especially for FM detection at 2 Hz. This result is interpreted as reflecting a detrimental effect of age on the use of temporal‐fine‐structure cues for low‐rate FM detection. The effect of increasing number of modulation cycles was similar across groups for both AM and FM for the 2‐Hz rate. For the 20‐Hz rate, the older listeners showed a slightly greater effect of increasing number of modulation cycles than the younger listeners for both AM and FM. These findings suggest that ageing spares temporal integration of the cues used to detect AM and FM. ARCHES 2015 – Groningen NL 47 P21: Speech recognition on neural data Alban Levity1,2, Christian J. Sumner2, Stephen Coombes2 and Aristodemos Pnevmatikakis3 1
MRC Institute of Hearing Research, Nottingham, UK Department of Mathematical Sciences, University of Nottingham, Nottingham, UK 3
Athens Institute of Technology, Paiania, Greece 2
The representation of speech along the human auditory pathway varies from a pure spectrotemporal representation at the cochlea to one containing more abstract features in the auditory cortex. Presumably in part due to this processing, speech understanding is remarkably robust to factors such as environmental noise compared with artificial speech recognition (ASR). Attempts to incorporate cochlear models into ASR have shown benefits to noise robustness, but these are fairly modest (Stern, 2011), suggesting that processing by the central auditory system must also play a role. The use of machine learning tools to assess coding in populations of central auditory nuclei is common place, but simple linear decoding predominates, which may well not capture the sophistication of upstream processing required for recognising complex sounds such as speech. A potential tool to simulate more ‘real‐world’ recognition, is to decode neural activity using ASR systems such as Hidden Markov Models (HMM) or Artificial Neural Networks (ANNs). Such a system could be used, for example, to assess how changes in the representation across the auditory pathway impact on speech recognition. We present our early attempts at developing an ASR suitable for recognising speech from spike trains, using the HMM‐
based software HTK. Specifically, here, we aim to systematically compare several spike pre‐processing schemes, applied to decoding phonemes and words from a non‐linear cochlear model (Sumner et al., 2003). 48 ARCHES 2015 – Groningen NL List of participants with email addresses: Name
Affiliation
Email address
Michael Akeroyd
IHR – UK
[email protected]
Sofie Aspeslagh
IHR – UK
[email protected] Deniz Baskent
UMCG ‐ Groningen
[email protected] Thomas Bentsen
DTU ‐ Copenhagen [email protected]
Nawal El Boghdady
UMCG ‐ Groningen
[email protected] Tim Bost
VUmc‐Amsterdam
[email protected]
Kris Boyen (17/11)
UMCG ‐ Groningen
[email protected]
Monique Boymans
AMC‐ Amsterdam
[email protected]
Jens Cubick
DTU ‐ Copenhagen [email protected]
Torsten Dau
DTU ‐ Copenhagen [email protected]
Hanne Deprez
KU ‐ Leuven
[email protected]
Pim van Dijk
UMCG ‐ Groningen
[email protected]
Mirjan van Dijk
UMCG ‐ Groningen
[email protected] Norbert Dillier
USC ‐ Zurich
[email protected] Wouter Dreschler
AMC – Amsterdam [email protected] Bastian Epp
DTU ‐ Copenhagen [email protected]
Stephan Ernst
Uni‐Oldenburg / Hearing4All
stephan.ernst@uni‐oldenburg.de Stephan Ewert
Uni‐Oldenburg / Hearing4All
stephan.ewert@uni‐oldenburg.de Christian Fullgrabe
IHR – UK
[email protected] Etienne Gaudrain
UMCG ‐ Groningen
[email protected] Mirjam van Geleuken (17/11)
AMC ‐ Amsterdam
[email protected]
Theo Goverts
VUmc – Amsterdam
[email protected]
Robin Gransier
KU ‐ Leuven
[email protected]
Christopher Hauth
Uni‐Oldenburg / Hearing4All
christopher.hauth@uni‐oldenburg.de Jan Heeren
Uni‐Oldenburg / Hearing4All
jan.heeren@t‐online.de Hynek Hermansky
Uni‐Oldenburg / Hearing4All
[email protected] Jolien Hessels (16/11)
UMCG ‐ Groningen
[email protected] Gijs Hoskam (16/11)
AMC – Amsterdam [email protected] Hamish Innes‐Brown
KU ‐ Leuven
[email protected]
ARCHES 2015 – Groningen NL 49 Name
Affiliation
Email address
Anoop Jagadeesh
Uni‐Oldenburg / Hearing4All
anoop.jagadeesh@uni‐oldenburg.de D. Enja Jung
UMCG ‐ Groningen
[email protected]
Andrew King
ENS ‐ Paris
[email protected] Emile de Kleine
UMCG ‐ Groningen
[email protected]
Thomas Koelewijn
VUmc‐Amsterdam
[email protected] Birger Kollmeier
Uni‐Oldenburg / Hearing4All
birger.kollmeier@uni‐oldenburg.de Elouise Koops UMCG ‐ Groningen
[email protected] Peter Jan Laverman
VUmc – Amsterdam
[email protected]
Alban Levity
IHR – UK
[email protected]
Jacqueline Libert
UMCG ‐ Groningen
[email protected] Christian Lorenzi ENS ‐ Paris
[email protected] Annika Luckmann (16/11)
UMCG ‐ Groningen
[email protected] Micha Lundbeck
Uni‐Oldenburg / Hearing4All
micha.lundbeck@uni‐oldenburg.de Richard McWalter
DTU ‐ Copenhagen [email protected] Nihaad Paraouty
ENS ‐ Paris
[email protected]
Andreas Prokopiou
KU ‐ Leuven
[email protected]
Ilja Reinten
AMC – Amsterdam [email protected] Anouk Sangen
KU ‐ Leuven
[email protected] Mark René Schaedler
Uni‐Oldenburg / Hearing4All
marc.r.schaedler@uni‐oldenburg.de Marya Sheikh Rashid
AMC – Amsterdam [email protected] Cas Smits (16/11)
VUmc – Amsterdam
[email protected]
Dimitar Spirrov
KU – Leuven
[email protected]
Sonia Tabibi
USC ‐ Zurich
[email protected]
Paolo Toffanin
UMCG ‐ Groningen
[email protected] Anita Wagner
UMCG ‐ Groningen
[email protected]
Dorothea Wendt
DTU ‐ Copenhagen [email protected]
Nicolas Wallaert
ENS ‐ Paris
[email protected] Astrid van Wieringen
KU ‐ Leuven
[email protected] Dietmar Wohlbauer
USC ‐ Zurich
[email protected]
50 ARCHES 2015 – Groningen NL ARCHES 2015 – Groningen NL 51 52 ARCHES 2015 – Groningen NL