bluuebunny/crossref_metadata_2025 huggingface dataset:
Yanny or Laurel? Acoustic and non-acoustic cues that influence speech perception
Authors: Monson Brian B.
Published: October 2018

“What do you hear?” This question that divided the masses highlights the complex nature of speech perception, which is dependent upon both acoustic and non-acoustic information available to the brain. In this talk, I will discuss some of the most recent advances in our understanding of both types of information that enable the complicated task of speech perception under adverse listening conditions. Recent research from our lab and others has revealed a surprising amount of speech signal information resides in the highest audible frequency range for humans (i.e., beyond 8 kHz), providing localization, phonetic, and other speech source information. These cues, as well as non-acoustic (e.g., experiential, visual, lexical) cues likely become increasingly important when a listener is faced with ambiguity and/or degradation of the acoustic signal. Given the diversity of prior experience from individual to individual, differences in perception of an ambiguous/degraded speech signal are to be expected.
DJ Prof: Mixing instructional modes to improve student learning
Authors: Wage Kathleen E.
Published: October 2018

Active learning encompasses a broad variety of activities designed to engage students with material while instructors are present to answer questions. In contrast, traditional lecture instruction is passive, and students' initial interaction with material is relegated to homework, which is often completed alone. Freeman et al.'s meta-analysis of 225 studies comparing lecture and active learning concluded that active learning is the "preferred empirically validated teaching practice" and showed that active learning courses have significantly lower failure rates than lecture courses [PNAS, 2014]. Incorporating proven active learning techniques into acoustics courses will improve student learning. In Acoustics Today [2016], John Buck, Jill Nelson, and I compared the professor in an active learning course to a DJ mixing different modes of instruction. DJ Prof designs a student-centered environment combining collaborative in-class exercises, short lecture segments, online video examples, and reading assignments. If you're an active learning skeptic, this talk will highlight empirical evidence of its benefits. If you're interested in getting started with active learning, this talk will suggest easy exercises to try. If you're an experienced DJ Prof, this talk will provide examples to amp up your pedagogical playlist.
Simulated wind turbine emissions: Results from additional human response testing
Authors: Nelson Peggy B. , Waggenspack Matthew , Byrne Andrew , Sullivan Michael , Adams Meredith , Marr Jeffrey D.
Published: October 2018

Previous data (Nelson et al., ASA 2017) indicated that healthy human adult participants experienced few symptoms from re-created wind turbine sound and infrasound emissions. That report included more than fifty subjects ages 21–73 years who attended to audible and infrasound signals generated from a wind turbine, recorded at 300 meters and re-created in a laboratory. Stimuli consisted of modulated and unmodulated audible turbine sound at 50 dB SPL, as well as natural and peak-enhanced turbine infrasound at an overall level of approximately 85 dB SPL (peaks up to 100 dB SPL). Participants were tested for their postural stability, detection, and ratings of audible and infrasound emissions randomly presented in one-minute exposure intervals in the laboratory. Very few and minor adverse effects had been noted to date, mostly ear fullness or pressure. Healthy participants showed no evidence of any change in postural sway in the presence of infrasound for the group tested. We have recently begun testing of participants who either a) live near turbines and complain of adverse effects, or b) who have symptoms of dizziness/imbalance as reported to their ENT specialist. Results from postural sway, sound quality judgments, and pre- and post-exposure symptoms will be reported from some of these participants. [Work supported by Xcel Energy RDF14.]
Wind turbine infra-sound penetration inside homes from multiple variable speed turbines
Authors: Metelka Andy
Published: October 2018

Previous measurements in homes with respect to dominant turbines indicated the transmissibility factor calculations were a valid technique for describing wind turbine infra-sound penetration at various locations inside homes. However, when several turbines contribute at different RPM’s and/or are not in phase different techniques need to be considered due to RPM variation and FFT densities. A comparison of the different methods is shown with examples of both cases.
Thresholds of noise in harmonic series maskers
Authors: Labuschagne Ilse B. , Ciocca Valter
Published: October 2018

Numerous studies have investigated the detection of pure tones and harmonic series in noise, but far fewer studied the perception of noise in harmonic series. One such study [Gockel, Moore and Patterson, J. Acoust. Soc. Am., 111(6), 2759–2770 (2002)] demonstrated that the masking thresholds of noise in harmonic series maskers were affected by the fundamental frequency (F0) and the overall level of the series, and by the relative phase of the harmonic components. The maskers used by Gockel et al. comprised unresolved harmonics (10th and higher harmonics below 5 kHz) for which F0 and frequency range covaried. The current study investigated noise detection for three non-overlapping spectral bands of equal auditory filter bandwidths (ERBs). Bands included either resolved harmonics (B1), unresolved harmonics (B3), or both resolved and unresolved harmonics (B2). Masker F0 and overall level were also varied. A Bayesian linear mixed-effects analysis showed that noise detection was better for higher F0s in B1 and B2, and that a higher presentation level resulted in a small improvement in noise detection in B2. Noise detection was better for higher presentation levels in B3. The findings will be discussed in relation to predictions of auditory processing models [Patterson, Allerhand, Giguère, J. Acoust. Soc. Am. 98(4), 1890–1894 (1995)].
An investigation of the training effects in a seven-week auditory localization training using augmented reality
Authors: Chon Song Hui , Kim Sungyoung
Published: October 2018

We present the results from a seven-week auditory localization training using augmented reality with Microsoft HoloLens. For each trial, the target and distractor(s) would be randomly placed in space. In training, after the participant estimated the invisible target location, both true and estimated locations would be visually displayed. A localization score was calculated and recorded based on their distance. The test modules were identical to the training modules, except that the true location was not shown. The target was a female singing, and the distractor a piano accompaniment. Eight listeners were divided into two groups after the initial test, keeping the mean and variance of localization scores at the same level. Both groups showed a declining pattern in localization score over the eight tests, unlike the results from our previous study. The decreasing slope was smaller for the train group than the control group, which might reflect some mild effect of training. There was a one-time performance improvement after two trainings. The experiment might have been too simple to maintain participants’ attention for weeks. Possible extraneous factors such as academic schedule might also have had an impact on this decreasing pattern.
Adapting masking thresholds for spatially separated sounds in two dimensional ambisonics
Authors: Adler Yuval , Murgai Prateek
Published: October 2018

Spatial audio reproduction has gained renewed interest in recent years with the increasing presence of virtual reality applications. Ambisonics is currently the most widely adopted format for spatial audio content distributed via internet streaming services, and this has raised the need for audio compression relevant to the format. Current perceptual audio coders used for stereo audio content rely heavily on masking thresholds to reduce data rates, but these thresholds do not take into consideration spatial release from masking. This study begins an effort to update these thresholds for spatially separated sources in ambisonics. The listening tests were performed with sounds encoded in ambisonics to allow for tests to integrate whatever inherent limitations exist in the format. Initial listening tests were carried out for a subset of possible conditions—sound sources were separated along the horizontal plane, with a specific set of separation angles between the masker and maskee. Suggestions are given for continuing the work for the full range of possible conditions.
The lexical representation of second language length contrasts: Native English speakers learning Japanese
Authors: Hayes-Harb Rachel , Barrios Shannon L.
Published: October 2018

Adults experience difficulty using novel second language (L2) phonological contrasts to distinguish words. Indeed, even the ability to perceive and/or produce a novel contrast with relative accuracy does not guarantee an ability to implement the contrast to distinguish words in tasks that require lexical access. These observations lead to questions regarding the phonological content of learners’ lexical representations of difficult L2 contrasts. We employed an artificial lexicon study to examine the lexical encoding and implementation of Japanese consonant and vowel length contrasts by native English speakers. In the first experiment, native English speakers were taught a set of six Japanese-like auditory minimal pairs with pictured meanings. The members of each pair were differentiated by vowel length (e.g., [teki] vs. [teeki]). Participants were then asked to match the pictures to auditory words in four conditions: matched (see picture of ‘teki’, hear [teki]), vowel length mismatch (see ‘teki’, hear [teeki], and consonant length mismatch (see ‘teki’, hear [tekki]). Participants performed accurately on matched items, but were more likely to reject word forms mismatched for consonant than for vowel length. The results from this and subsequent experiments provide insight into the lexical encoding strategies used by learners for difficult novel contrasts.
Effects of L2 experience on L2 vowel production and L1 speakers' perception of the improved vowels
Authors: Oh Grace E.
Published: October 2018

Effects of L2 experience on the production of L2 vowels were investigated and the perceptual accuracy of the L2 vowels by native speakers was tested to examine whether an improvement in vowel articulation leads to higher accuracy. A total of twenty Mandarin Chinese differing in the experience (6 months vs. 2 years) were compared to ten native Korean speakers in their production of seven Korean vowels, /i,ɛ,ɨ,ʌ,o,u,a/. More experienced Chinese speakers were expected to have acquired new Korean vowels, /ɛ,ɨ,ʌ,o/, in a more native-like manner than the inexperienced group. The results showed that Chinese learners were able to produce the three similar vowels, /i, u, a/ in a native-like manner even before any exposure to Korean. After 2 years of experience, they have shown to produce the new mid vowels, /ɛ, ʌ/ with greater height distinctions (F1 values) from adjacent high vowels. Although the two high back vowels, /o, u/ were deviant from the native norm, the Experienced group learned to distinguish the vowels with a distinctive F2 frequency, which has become a critical cue for the distinction of the two merging vowels by young Seoul speakers. When native Korean speakers were asked to identify the vowels produced by Chinese speakers, not only new mid vowels but also similar vowels were more accurately identified for the experienced group. The greatest improvement in perceptual accuracy was shown for the categories that were newly established by the inexperienced learners.
Lexical representation of Mandarin tones in second language learners
Authors: Sun Kuo-Chan
Published: October 2018

Previous studies have shown that a word’s phonological similarity to other words (i.e., phonological neighborhood) can influence its recognition. However, most research concerning lexical representations has been observed for neighbors based on segmental overlap, and little is known about such effects with suprasegmentals such as Mandarin tones. In the present study, two experiments were conducted with forty L2 listeners and 40 native speakers to examine how tone neighborhood density influences Mandarin spoken word recognition. In Experiment 1, speed and accuracy from both groups’ performance in an auditory lexical task were influenced by tone neighborhood density (i.e., fewer words were recognized from dense tone neighborhoods than from sparse tone neighborhoods). However, L2 listeners’ performance was inferior to native listeners’. In Experiment 2, form priming patterns showed that reliable facilitation was observed only when the prime and the target were identical, while monosyllabic Mandarin words differing only in tone failed to speed the response to the target. In addition, only L2 listeners showed an increase in RTs to respond to the target when it was preceded by tone overlap primes. The results of these experiments demonstrate that tone neighborhood is an important factor in L2 Mandarin spoken word recognition.
Acquisition of categorical perception of Mandarin tone
Authors: Li Michelle X. , Rivas Daniel , Pérez Gay Juárez Fernanda , Sicotte Tomy , Harnad Stevan
Published: October 2018

This experiment examined the effect of motor repetition on the learning of Mandarin pitch categories by non-native speakers. In Mandarin, there are four tonal categories, which refer to pitch contours that discriminate words. Differentiating tonal categories is both essential and arduous for non-native speakers to learn. Native speakers do this effortlessly because they have a categorical perception (CP) effect for tones, i.e. they perceive items within a tonal category as more similar to each other and items between categories as more different. Non-native speakers do not have this effect, and this experiment attempted to induce the CP effect for Mandarin tones via a training task, which was the experimental manipulation: Participants either repeated a tone stimulus before categorizing it or listened to a stimulus and categorized it without repetition. Discrimination between and within tonal categories was measured before and after training. All participants demonstrated increased between-category and within-category discrimination after training, except for learners who repeated stimuli in the training phase. They demonstrated a decrease in within-category discrimination, showing a weak CP effect that could be stronger with more training. Implications of these results on auditory category learning and language education will be discussed.
Experimental study of efficient multi-band underwater acoustic communication algorithm
Authors: Lee Hui Su , Baek Chang Uk , Seo Jung-Hyun , Jung Ji Won , Do Dae Won
Published: October 2018

In the underwater acoustic communication environment, multipath transfer characteristics or Doppler spread due to time and space changes such as seabed, sea level, and water depth affect the performance. The multi-band communication technique is effective in terms of performance and throughput efficiency because it can overcome selective frequency fading by allocating the same data to different frequency bands in the environment of rapidly changing channel transfer characteristic. In addition, the transmission distance can be further extended while overcoming various underwater channel environments. However, the multi-band configuration may have worse performance than the single-band one. It is because the performance degradation in a particular band affects the output from the entire bands, which is input into a decoder, thereby decreasing the overall performance. This problem can be solved through a receiving end that analyzes error rates of each band, sets threshold values, and allocates lower weights to inferior bands. In this paper, we proposed an algorithm to set the threshold value using the preamble error rate, which is known data to be transmitted and received. We have analyzed the efficiency of multi-band transmission scheme by applying 1–4 number of multi-bands using turbo pi code through actual underwater experiment.
Measuring speech perception with recovered envelope cues using the peripheral auditory model
Authors: Mamun Nursadul , Akter Khadija , Ali Hussnain , Hansen John H. L.
Published: October 2018

Speech perception refers to how understandable speech produced by a speaker would be by a listener. The human auditory system usually interprets this information using both envelope (ENV) and temporal fine structure (TFS) cues. While ENV is sufficient for understanding speech in quiet, TFS cues are necessary for speech segregation in noisy conditions. In general, ENV can be recovered from the TFS (known as recovered ENV); however, the degree of ENV recovery and its significance on speech perception are not clearly known/understood. In order to systematically assess the relative contribution of the recovered ENV for speech perception, this study proposes a new speech perception metric. The proposed metric employs a phenomenological model of the auditory periphery developed by Zilany and colleagues (J. Acoust. Soc. Am. 126, 283–286, 2014) to simulate the responses of the auditory nerve fibers to both original and recovered ENV cues. The performance of the proposed metric was evaluated under different types of noise (both steady-state and fluctuating noise), as well as several classes of distortion (e.g., peak-clipping, center-clipping, and phase jitter). Finally, to validate the proposed metric, the predicted scores were compared with subjective evaluation scores from behavioral studies. The proposed metric indicates a statistically significant correlation for all cases and accounts for a wider dynamic range compared to the existing metrics.
The Biquaternionic acoustic wave equation
Authors: Oba Roger M.
Published: October 2018

Equations for irrotational, linearized Euler fluid, where the mass continuity equation incorporates a linear equation of state, define linear acoustics that can be recast in terms of complexified quaternions, the biquaternions, which satisfy Cauchy-Fueter regularity equations. This gives first order partial differential equations from the four space-time dimensions to the four (complex) dimensional velocity-pressure space, with time and pressure in the biquaternionic scalar parts. The biquaternionic planar exponential provides the basis to elementary solutions as biquaternionic functions of a single biquaternionic variable. Analysis of this form shows significant geometrical algebraic properties including imaginary units for time distinct from those for space. The single biquaternion acoustic field at each point of space-time allows the propagation four complex coefficients in a single geometric algebraic structure, suitable for three or four dimensional Fourier analysis. (Work supported by the U.S. Office of Naval Research.)
Fast two-dimensional censored mean level detector-statistics constant false alarm rate method for multi-beam seafloor terrain detection
Authors: Wang Jiaqi , Li Haisen
Published: October 2018

In order to solve the problem of false terrain caused by environmental interference and “tunneling effect” in conventional multi-beam seafloor terrain detection, this paper addresses a seafloor topography detection method based on fast two-dimensional (2D) CMLD-CFAR (censored mean level detector-Statistics Constant False Alarm Rate). A cross-sliding window is used in this method. The target occlusion phenomenon which occurs in multi-target environments can be eliminated by censoring the large cells of the reference cells, and the rest reference cells are used to calculate the local threshold. The conventional 2D CMLD-CFAR methods need to estimate the background clutter power level for every pixel, which costs a lot of computational load. This paper proposes a fast algorithm with a global threshold to reduce the computational load. Only the regions of interest (ROI) which are selected by a global threshold will be detected by 2D CMLD-CFAR method, and the rest pixels are distinguished as clutters directly. The proposed method has been evaluated by the real multi-beam experimental data of which the background follows a distribution of Exponential. The results show that the novel method can effectively solve the problem of false terrain in multi-beam terrain survey and has a high detection accuracy.
Nonsingular EB-ESPRIT for the localization of early reflections in a room
Authors: Jo Byeongho , Choi Jung-Woo
Published: October 2018

The eigenbeam-ESPRIT (EB-ESPRIT) can directly identify the direction of arrivals (DoAs) of sound sources through the parametric estimation process. This parametric estimation is beneficial for identifying directions of early echoes in an enclosed space, because it does not need an extra peak searching step for finding dominant peak locations from a beamforming power map. Nevertheless, the use of EB-ESPRIT for echo localization has been hindered due to its ill-conditioning problem for sources positioned near the equator of the spherical coordinates. Room reflections in a reverberant room can impinge from any direction, and hence, the ill-conditioning is highly likely to occur. In this work, a nonsingular EB-ESPRIT technique is utilized for the echo localization without the ill-conditioning problem. The technique is based on new recurrence relations of spherical harmonics that express DoA parameters in form of sinusoidal functions. Therefore, the ill-conditioning due to the use of arctangent function can be avoided. The performance is demonstrated using the room impulse responses simulated and measured for a rigid spherical microphone array. Frequency smoothing is applied to localize echoes that are highly coherent to each other, and results demonstrate that accurate localization of echoes is possible irrespective of their incidence angles.
Modal reconstruction of the sound field in a room at low frequencies
Authors: Fernandez-Grande Efren
Published: October 2018

Capturing the spatio-temporal (or spatio-spectral) properties of the sound field in a room is valuable for its characterization. To measure the full spatio-spectral properties within the volume of the room, i.e., the frequency response functions at all points, a very large number of measurements is required. Even at low frequencies, sampling the three-dimensional space enclosed by a room is challenging. It is therefore of interest to employ sound field reconstruction methods to predict (interpolate/extrapolate) the sound field over the volume of the room, based on a limited number of measurements. The aim of this study is to develop a sound field reconstruction method that exploits the modal structure of the sound field in a room, to reconstruct the sound field at low frequencies over the entire volume of the space. The proposed methodology makes use of the fact that the spatial structure of the acoustic field in a room is inherently block-sparse and can be extracted from spatially distributed measurements. The modeshapes are represented as the superposition of a small number of plane waves, and the modal basis is used to reconstruct the sound field.
Riverscape meets underwater soundscape: Acoustic habitat selection by brook char in a small stream
Authors: Kacem Zaccaria , A. Rodríguez Marco , Proulx Raphaël
Published: October 2018

Stream habitats are characterised by geophysical descriptors such as water temperature, depth, substrate type, and flow speed. So far, few studies have focused on underwater sounds as an important feature of habitat selection by fish. In this study, we described stream habitats at high resolution to evaluate the relative importance of the underwater soundscape and other geophysical descriptors for understanding the distribution of brook char densities. Our results showed the high acoustical heterogeneity of stream habitats (ranging from 40 dB up to 150 dB re 1uPa), which was related to differences in water velocity and depth as expected from theory. Brooks char densities were nevertheless positively related to sound intensities, irrespective of water velocity, depth, or facies type. Our findings showed that underwater sounds integrate the many environmental dimensions of stream and may be used as cues for habitat selection. The positive relationship between brook char densities and sound intensities could be related to the high auditory threshold of Salmonidae.
Long-term characterization of the marine soundscape in Kona, Hawaii
Authors: Merkens Karlina , Oleson Erin M. , Baumann-Pickering Simone
Published: October 2018

Soundscape analysis of marine environments is still a new field of inquiry, with the majority of studies focusing on individual species or acoustic functional groups. Additionally, the challenges of making long term acoustic recordings in marine habitats, particularly in deep water, have limited the duration of many analyses. The Pacific Islands Fishery Science Center has been collecting passive acoustic data at a depth of 630 m off the Kona coast of the Island of Hawaii since 2007, and has recently begun a soundscape analysis effort to characterize the contributors to the local acoustic environment. This project seeks to examine not only low frequency noise patterns, below 1 kHz, but also higher frequency sounds that may overlap in frequency with the signals of many cetacean species. We have begun by identifying the relative contribution of odontocete echolocation, and also close approaches of boats and the presence of echosounders. The temporal patterns in these signals reveal daily and seasonal cycles in biological and anthropogenic sounds, which provide both detailed insight into the interactions between humans, animals and their environment, and long term trends in activity of those sound sources.
Preclinical assessment and enhancement of passive acoustic mapping in clinically realistic propagation environments
Authors: Gray Michael , Coussios Constantin
Published: October 2018

Passive acoustic mapping (PAM) techniques have been developed to detect, localize, and quantify cavitation activity during therapeutic ultrasound treatments. Building on a decade of in vitro and small animal studies, this paper presents a series of pre-clinical experiments and simulations conducted in preparation for an upcoming clinical trial of ultrasound-mediated targeted drug delivery. The effects of tissue attenuation and refraction, array shape and element diffraction, soundspeed uncertainty, and imaging algorithm will be discussed. Critically, techniques will presented for removing factors that would otherwise bias PAM results both within and between patients. Together, the results indicate the potential to significantly enhance both the qualitative and quantitative capabilities of PAM for ensuring clinical therapeutic safety and efficacy. [Work supported by the National Institute for Health Research Oxford Biomedical Research Centre.]
Estimation of the thermo-physical property of biological tissues from temperature rise due to ultrasound exposure
Authors: Tsujimoto Yukako , Akiyama Iwaki
Published: October 2018

This study discusses the feasibility of a method for ultrasonic tissue characterization according to the thermo-physical properties of biological tissues. Since this method measures the ratio of sound velocity variation due to ultrasonic heating, it is significant to clarify the relationship between the ratio and thermo-physical properties. The ratio of sound velocity variation of a tissue sample was measured from phase shifts of echoes before and after the temperature rise by transmitting an ultrasonic pulsed wave. In the experiments the transducer used for heating was a resonance frequency of 5.0 MHz, concave ring shape, inner diameter of 5 mm, and outer diameter of 12 mm. It was arranged coaxially on the outer perimeter of the transducer used for the measurement. This transducer was a resonance frequency of 10 MHz, a focal distance of 15 mm, and a diameter of 4 mm. The exposure time of heating ultrasound is 100 ms. As a result of heating sound pressure of 1.0 MPa, 1.5 MPa and 2.0 MPa, the estimated values of temperature rise were 0.56 ˚C, 1.1 ˚C and 2.0 ˚C, respectively. [This study was supported by MEXT-Supported Program for the Strategic Research Foundation at Private Universities, 2013-2017.]
Spherical harmonic expansions of high-resolution directivity data
Authors: Bellows Samuel D. , Leishman Timothy W.
Published: October 2018

Quantifying the directivity patterns of sound sources provides not only deeper insights into understanding the source and its acoustical properties but also the potential to simulate the propagation of its sound in various settings. Being able to describe the detailed 3-dimensional directivity of highly complex and dynamic sound sources such as musical instruments and human speech in a simple way while at the same time preserving the fine details of the directivity patterns has broad applications. By using seventeenth-order spherical harmonic expansions of data collected on a 2,664-point spherical array, large amounts of the measured directivity of musical instruments and human speech was condensed in a simple, accurate, and powerful tool for understanding these complex systems.
Energetic and informational masking effects of hospital noise
Authors: Gallun Frederick J.
Published: October 2018

The ability to understand speech in a hospital setting is limited by the overlap of speech and noise energy at the level of the cochlea, sometimes called “energetic masking.” However, there are also additional “informational masking” effects that can occur when the noise is comprised of informational sounds such as speech, alarms, or other meaningful auditory events (e.g., Kidd and Colburn, 2017). While energetic overlap has been well-studied and is quantified in a number of international standards such as the Speech Intelligibility Index (ANSI, 2004), there is no similar metric for informational masking. Two of the main difficulties with predicting informational masking, and thus the intelligibility of speech in real world environments, are 1) the lack of tools that defines the meaningfulness of an interfering sound and 2) the much greater variability in susceptibility of individual listeners to informational as opposed to energetic masking. These ideas will be illustrated with data from several recent studies of speech intelligibility with various types of maskers, including hospital noise, and several current and emerging techniques for assessing speech intelligibility for individual listeners in simulated environments will be described.
Do orienting responses predict annoyance from irrelevant sounds?
Authors: Oliver Jordan N. , Sollmann Lea , Niehl Annika , Francis Alexander L.
Published: October 2018

Unexpected sounds are distracting and can be annoying but individuals may differ in susceptibility to them. Irrelevant sounds occurring at sparse temporal intervals induce a psychophysiological orienting response reflecting involuntary capture of attention away from the primary task. We hypothesize that the frequency and/or magnitude of individual listeners’ orienting responses to irrelevant sounds will predict annoyance ratings and task performance in distracting noise. Participants read essays while seated in a comfortable chair in a sound-shielded booth facing a semicircular array of 6 speakers located 1.5 m away at 30°, 60°, and 90° to the left and right. Unintelligible background speech (ISTS) played at 60 dB(A) SPL from each loudspeaker (unsynchronized). At 50–70 s intervals one of 12 non-speech sounds (IADS) played for 6 s from one loudspeaker at approximately 70 dB(A) SPL. Order and location of sounds were randomized but each sound played from each speaker exactly once over the experiment (72 trials, ~80 min). Cardiovascular, electrodermal, electro-ocular, and bilateral posterior auricular muscle activity were recorded from participants to quantify orienting response. Behavioral measures of reading comprehension, noise sensitivity, personality traits, and subjective effort, frustration and annoyance were also collected and will be related to physiological measures.
Particle assembly and object propulsion using acoustic holograms
Authors: Melde Kai , Fischer Peer
Published: October 2018

The contact-free manipulation of particles using ultrasound fields is an active field of research promising a number of applications. Conventional acoustic tweezers use strongly focused beams or higher order Bessel beams to provide a trap for single particles to be manipulated with. Other more mature methods use resonators to create elongated potential wells for collective particle trapping or separation. The resulting assemblies have a limited complexity, because the fields are highly symmetric. We recently introduced the acoustic hologram as an alternative way to create arbitrary ultrasound fields. In this talk, I will present two concepts, one for particle trapping and one for propulsion of objects, that have been enabled by this new method. The first is parallel assembly of microparticles at a surface in the shape of a projected acoustic image. Using a global trigger, these particles can be fused together to form a mechanically stable object. The second demonstration is a seemingly dynamic effect resulting from our static hologram. By defining the phase gradient (essentially the wave vector) along the water-air interface, it is possible to continuously propel objects along predefined trajectories. The necessary complexity to create such ultrasound fields with defined amplitude and phase distribution is easily managed using acoustic holograms.
Vowel formant trajectories in naturalistically produced loud speech
Authors: Fuchs Susanne , Koenig Laura L.
Published: October 2018

Louder speech, relative to typical speech, may show changes in addition to the expected increased SPL, such as higher values of the first formant (F1), longer durations, and higher articulatory velocities. Most past work has assessed vowel midpoints only. To what extent can any formant differences be observed across the full time course of the vowel? Here we evaluate formant trajectories in loud and conversational speech. Adult female speakers of German produced acronyms containing /a: i: u:/, and words that put various vowels in a bilabialalveolar context. Greater loudness was elicited by increasing speaker-experimenter distance. Preliminary analyses extracting values at 25%, 50%, and 75% of the vowel duration show that formant trajectories in normal and loud speech are usually quite parallel throughout the vowel, particularly for high and tense vowels (where loudness differences are minimal overall). In some cases normal-loud differences may increase from the 50% to 75% time points. In general, it does not appear that articulatory/acoustic differences between loud and normal speech are restricted to vowel midpoints. We will also carry out analyses over shorter time intervals to evaluate very early and late regions in the vowels.
Positional allophony in ejective stops: A case study of Georgian
Authors: Gfeller Chloé
Published: October 2018

Many phonetic studies have dealt with ejective stops, but allophonic differences between initial and intervocalic positions have not been investigated. Ejectives have been typologically classified as either “stiff” or “slack,” depending on their glottal tension (Kingston 1985). Since stops in general tend to show positional acoustic differences superficially similar to those induced by differences in glottal tension (notably differences in VOT duration), we ask whether positional allophony of ejectives manipulates glottal tension. In an acoustic study of Georgian ejectives, we observed longer mean VOTs in initial position for labial (28 ms vs 23 ms) and coronal ejectives (33 ms vs 25 ms), consistent with increased glottal tension. Additionally, in a subset of cases, the vowel following an intervocalic ejective is completely glottalized, consistent with decreased glottal tension in intervocalic position. However, we found no significant difference overall in the mean duration of glottalized vowel phonation after ejectives between initial and intervocalic positions, notably because, in the other intervocalic tokens, the duration of vowel glottalization is comparable to that in initial position. This suggests that the observed positional differences in VOT in Georgian ejective stops are not reducible to differences in glottal tension.
Regional variation in West Yorkshire filled pauses: Implications for forensic speaker comparisons
Authors: Gold Erica , Ross Sula , Earnshaw Kate
Published: October 2018

In the current linguistic literature, West Yorkshire (a county in Northern England) has received relatively little commentary, as it is often overshadowed by other bigger regions and cities like Manchester or Newcastle-upon-Tyne. However, in forensic phonetics, literature on regional variation is often vital to forensic casework. Sociophonetic studies aid forensic phoneticians in making judgments regarding whether speaker characteristics are typical of a region or not. For both forensic and sociophonetic motivations, this paper begins to look at the variation present in West Yorkshire by analyzing variation in filled pauses across speaking styles from three boroughs within West Yorkshire (Bradford, Kirklees, and Wakefield) from the West Yorkshire Regional English Database (WYRED; Gold et al., 2016). This paper analyses 60 speakers from WYRED. All speakers are male, aged 18-30, and English is their first and only language. This study measures the vocalic portion of all “uh” /V/ and “um” /V + N/ productions in over 8,000 tokens across four different tasks. Filled pauses were manually segmented in Praat and F1, F2, and F3 midpoints were extracted. Results suggest that filled pauses may be influenced by speaking style, but more importantly the three regions exhibit some significant differences in their filled pause realizations.
Registration and fusion of 3D head-neck MRI and 3D/4D tongue ultrasound
Authors: Lulich Steven M.
Published: October 2018

Static 3D head-neck MRI can provide excellent coverage of the entire vocal tract and surrounding anatomy with good spatial resolution and signal-to-noise ratios. Real-time MRI takes advantage of these strengths and adds temporal resolution for dynamic speech production, but depends on regular repetition of the same speech utterances for several minutes. (Dynamic or cine MRI also depends on regular repetitions, but the process is stroboscopic rather than reconstructive, and the end result is a reduced number of time frames with longer frame durations.) Real-time MRI is not suitable for non-repetitive speech utterances. In contrast, 3D/4D ultrasound can provide excellent spatial resolution with good temporal resolution of non-repetitive speech utterances, but with lower signal-to-noise ratios and poor anatomical coverage. This study presents methods for registering and fusing static head-neck MRI and 3D/4D tongue ultrasound data, taking advantage of the strengths of both MRI and 3D/4D ultrasound to investigate the production of non-repetitive speech utterances.
Accidental gaps in Mandarin Chinese tones
Authors: Jin Shao-Jie , Lu Yu-an
Published: October 2018

Mandarin is a tone language with four phonemic tones (i.e., high-level Tone 1 [55]), rising Tone 2 [35], falling-rising Tone 3 [214], and falling Tone 4 [51] and with the maximum (C)(G)V(G)/(C) syllable structure. However, not all syllables can be combined with each of the tones (i.e., accidental gaps). For example, the syllable [tsʰu] is allowed to be combined with T1 ([tsʰu]55 “coarse”), T4 ([tsʰu]51 “vinegar”), but not T2 and T3. A calculation of all the 391 allowable syllables showed that there are 131, 185, 155, 110 accidental gaps in each of the four tones. A one-way chi-square test revealed that the accidental gaps in T2 were over-represented (stats here). A further investigation into these gaps in T2 according to different syllable types (CV: 52, CVN: 43, CGV: 38, and CGVN: 29) showed that these gaps were marginally under-represented in CGVN (stats here). We attributed these findings partially to the marked status of contour tones and to the typological preference for complex tonal targets to complex rimes due to their inherent longer durations (cf. Zhang 2001).
Tracking a surface ship via cascade of blind deconvolution and array invariant using a bottom-mounted horizontal array
Authors: Byun Gihoon , Song H. C. , Kim Jea Soo , Park Ji Sung
Published: October 2018

The array invariant developed for robust source-range estimation typically requires a priori knowledge about the source signal (i.e., cooperating source) to estimate the Green's function. However, the array invariant method has been recently extended to a surface ship radiating random signals (200-900 Hz) by extracting the Green's function via blind deconvolution using a vertical array [J. Acoust. Soc. Am. 143. 1318–1325 (2018)]. In this paper, the blind deconvolution is applied to a 60-m long, bottom-mounted horizontal array to extract the Green's function of the same ship circling around the array in a square spiral pattern at ranges of 300–1500 m. The overall tracking performance shows good agreement with GPS measurements except when the ship is towards the broadside with respect to the horizontal array. Further, simultaneous localization of multiple ships is discussed.
Application of preprocessing methods to improve time delay estimation for synthetic aperture sonar motion estimation
Authors: Gazagnaire Julia , Beaujean Pierre-Philippe
Published: October 2018

Synthetic aperture sonar (SAS) provides the best opportunity for side-looking sonar mounted on unmanned underwater vehicles to achieve high-resolution images at longer ranges. However, SAS processing requires maintaining a coherent phase history over the entire synthetic aperture, driving strict constraints on resolvable platform motion. This has driven the development of motion estimation and compensation techniques that use the received ping data, in addition to the onboard navigation solution, to resolve ping-to-ping platform motion. The most common approach is to use the redundant phase center technique. Here the ping interval is set, such that a portion of the array is overlapped. The accuracy of the motion estimation depends on the accuracy of the time delay estimation between the data received on the overlapping channels. Given the stochastic nature of the operational environment some level of decorrelation between these two signals is likely, even without residual platform motion. This decorrelation results in inaccurate time delay estimation and image quality degradation. In this research various preprocessing techniques have been applied to the sonar data to reduce the influence of stochastic noise with the goal of improving the accuracy of the time delay estimates.
Learning about hall acoustics from multi-directional audio recordings
Authors: Markham Benjamin , Sacks Jonah , Hochgraf Kelsey
Published: October 2018

Multi-channel audio recording, for playback in ambisonic format, is common when documenting a newly completed auditorium or an existing auditorium to be renovated. These recordings provide opportunities for comparative listening of different auditoria, and also between an acoustical model and the built auditorium. To allow meaningful comparison, recordings have been made using the same musicians playing the same musical excerpts on the same instruments. Anechoic recordings made similarly are used for convolution with synthetic (modelled) impulse responses. Comparative listening experience has yielded a number of lessons for acoustical design of auditoria and for acoustical modeling in design, including (a) the importance of instrument directivity to the sound of music in an auditorium, and perceptual responses to variations thereof, (b) the importance and challenge of loudness calibration during comparative listening, (c) inter-dependency of room size, perceived intensity and distribution of reverberation, and source directivity, (d) limitations in typical geometric acoustical modeling techniques, and (e) usefulness of directive loudspeakers when acquiring measured impulse responses for the purpose of convolving with anechoic audio. The process and results of these efforts will be presented and discussed. Selected listening segments will be available following the presentation.
Regression and correlation analysis of measured and simulated reverberation time in two Taiwanese Basilica churches
Authors: Lee Chia-Fen , Tsaih Lucky S.
Published: October 2018

There are few historical Basilica churches in Taiwan. None of them have been studied thoroughly on their acoustic characteristics with on-site measurements and model predictions. Wanchin Basilica of the Immaculate Conception and Tainan Theological College and Seminary Chapel were built between 1860 to 1960 AD. On-site measurement with ISO-3382 were performed to obtain reverberation time and background noise level. To obtain and document additional acoustic parameters and mappings of C80, D50, and STI, Odeon Acoustics was used. Since both churches are the historical buildings, it is difficult to identify the absorption coefficient of materials used in these churches for model simulation purpose. Thus, after selecting the approximate materials for the simulated model, validating the measured and simulated reverberation time with Linear Regression model is necessary. With 392 data points; standard error, least squares, and RMSD were calculated to derive the regression model with correlation (R) and determination coefficient (R2). R2 of regression models for both churches are 0.73 and 0.84. R values for both churches are 0.86 and 0.92. Both R and R2 indicated that the high correlation between predicted model and measured data. Thus, the confidence of predicated C80, D50 and STI can be raised and reliable.
Sound insulation of a single finite panel—Comparison, validation, and parametric studies
Authors: Huang Yihe , Chen Fangliang , DeGanyar Tejav
Published: October 2018

Sound transmission characteristics of a single finite panel is of significant value for both theoretical development and engineering application in building acoustics. Extensive studies and numerous models have been developed to study the sound transmission loss behavior of a single panel over the past decades. Though most of them have been validated separately with specific test results in their developing phases, evidences showed that considerable discrepancy exit from one model to another. Numerous comparisons among different prediction models have been presented in the literature. However, quantitative comparisons among them is still lacking. To guide the design of real practice, an impartial evidence of the applicability and accuracy of different models is highly desired. For this purpose, a quantitative comparison among more than ten widely adopted existing models are presented in this study. Test results collected from different labs for different materials, that found in the literature and conducted by the authors as well, are provided first to validate different models for different scenarios. Comprehensive parametric studies are then conducted to investigate the effects of different design parameters, such as boundary conditions, damping factor, panel dimension, and aspect ratio, on the sound transmission loss of a single panel.