bluuebunny/crossref_metadata_2025 huggingface dataset:
An algorithm to increase speech intelligibility for hearing-impaired listeners in entirely novel noises
Authors: Chen Jitong , Wang Yuxuan , Yoho Sarah E. , Wang DeLiang , Healy Eric
Published: May 2016

Supervised speech segregation has been recently shown to improve human speech intelligibility in noise, when trained and tested on similar noises. However, a major challenge involves the ability to generalize to entirely novel noises. Such generalization would enable hearing aid andcochlear implant users to improve speech intelligibility in unknown noisy environments. This challenge is addressed in the current study through large-scale training. Specifically, a deep neural network (DNN) was trained on 10,000 noises to estimate the ideal ratio mask, and then employed to separate sentences from completely new noises (cafeteria and babble) at several signal-to-noise ratios (SNRs). Although the DNN wastrained at the fixed SNR of 2 dB, testing using hearing-impaired listeners demonstrated that speech intelligibility increased substantially following speech segregation using the novel noises and unmatched SNR conditions of 0 dB and 5 dB. Sentence-intelligibility benefit was also observed for normal-hearing listeners in most noisy conditions. The results indicate that DNN-based supervised speech segregation with large-scale training is a very promising approach for generalization to new acoustic environments.
Attenuation of hearing protectors: A systematic comparison of subjective and objective measurement methods
Authors: Nélisse Hugues , Le Cocq Cécile , Boutin Jérôme , Laville Frédéric , Voix Jérémie
Published: May 2016

A key component when selecting a hearing protector is the noise attenuation offered by the device. The subjective Real-Ear Attenuation at Threshold (REAT) test method is the most commonly used procedure to measure attenuation. On the other hand, with the increase popularity of individual fit testing and miniaturization of electronic components, the Microphone-In-Real-Ear approach (MIRE), and its field counterpart F-MIRE, are becoming more appealing and well suited for estimating attenuation in laboratory or in “real world” occupational conditions. In this approach, two miniature microphones are used to measure sound pressure levels in the ear canal under the protector and outside of the protector. This study presents a systematic evaluation of the various factors relating the subjective and objective attenuation values. Experiments on human subjects were carried out where the subjects were instrumented on both ears with microphones outside and underneath their protector. They were then asked to go through a series of subjective hearing threshold measurements followed immediately by microphone recordings using high level broadband noises. Earmuffs, earplugs, and dual-protection were tested for each subject. The various factors relating the subjective and objective attenuation data are first presented. Results showing the relative importance of these factors are presented and discussed as well as various comparisons obtained with the different attenuation values.
Effect of smartphone orientation on sound level measurement
Authors: Klyn Niall A. , Feth Lawrence L.
Published: May 2016

The proliferation of smartphones provides an opportunity for acoustic measurements to be made by citizen scientists. However, there are concerns about the accuracy of data collected by the public. This study reports on the variability in smartphone sound level measurements due to how the device is being carried. A sample of smartphones running either Android or iOS and with various microphone placements were selected and calibrated for dBA measurements. These devices were tested in anechoic and semi-reverberant rooms while being situated to reflect common carrying positions. We found that the measured sound level in the anechoic chamber could vary by >10 dB depending on the orientation of the device, the location of the microphone, and the relationship of the device to the sound source. Smaller but still substantial variation was observed in the semi-reverberant space. These data highlight the variability probable in participatory sensing efforts, and underscore the potential usefulness of providing guidelines for data collection by citizen scientists. [Work supported by the Battelle Engineering, Technology and Human Affairs Endowment.]
Some observations on potential errors in acoustic and electroacoustic systems computer modeling
Authors: Mapp Peter , Boatman Josh
Published: May 2016

Although computer modelling of sound systems is now a mature science, it is surprising how many inaccurate models are still being made. The paper identifies and discusses the most common causes for modeling errors that the authors’ regularly encounter. These include (1) lack of model detail and dimensional accuracy; (2) use of statistical calculations in non-diffuse or inhomogeneous (statistical) spaces; (3) errors due to modeling of materials with unknown absorption and scattering properties; (4) incorrect (or corrupt) loudspeaker data; (5) incorrect consideration of boundary effects and net energy flows; and (6) use of statistical calculations where discrete, long path echoes and reflections are present. Examples of each of the above error mechanisms will be presented together with their typical resultant calculation errors.
The importance of phase, attention, intelligibility, and recall in sound system design
Authors: Griesinger David H.
Published: May 2016

Measures for the accuracy of speech transmission over an audio channel, such as Speech Transmission Index, rely on random word recognition as the standard for quality. In this paper, we propose that the ability to recall information at a later time is a more useful standard for sound quality than simply recognizing words. Recall, particularly in a complex or noisy environment, depends critically on the focused attention of a listener. There is evidence that “proximity,” the auditory perception that a source is close to a listener, enhances both focused attention and the ability to localize and separate sound sources from competing signals. We find that proximity and the benefits it brings depend on the phase alignment of frequencies above 1000 Hz. This alignment is lost when there are excess reflections, reverberation, noise, or multiple loudspeakers. We will present measurement techniques that can quantify proximity from an impulse response, and sound designs that maximize it in practice.
“Voice lift”—Improving sonic perception, including intelligibility, using multi-channel time variant systems
Authors: Barbar Steve
Published: May 2016

For many of our clients, the “optimum” electro-acoustic reinforcement system is one that has a “magic microphone.” This translates to something that they do not have to wear or hold, or stand in a specific spot to use. It does not need to have its battery replaced or cable plugged in. They might not even want to “flip a switch” to turn it on—it would just “be there” and all that they would need to do is talk. And if someone in the audience wanted to ask a question, they too would have a “magic microphone” that would enable everyone to hear them. While little has changed with the development of flying cars, advancements in multi-channel digital signal processing, and microphone arrays have made audio systems that function in this manner attainable. This paper will discuss factors that are important for intelligibility, how this differs from factors that are important for the perception sonic quality, and why both are required in order to achieve a comfortable and engaging listening experience. It will also describe the underlying complexity involved in designing and integrating such systems. Two recent examples will be cited—the Best Buy Theatre in Minneapolis, Minnesota, and the Netlflix corporate auditorium in Los Gatos, CA.
Listening for solutions to a speech intelligibility problem
Authors: Hoover K. Anthony
Published: May 2016

An historic Temple, built in 1929 in Los Angeles with 1800 seats and a 100-ft diameter dome, recently underwent a $150 million renovation, including a new sound system. Even after several system tunings, complaints of poor speech intelligibility remained. The system designers recommended additional sound absorptive treatment, as apparently based on their measurements, but this would be expensive, and would affect the visual aesthetic and perhaps the quality of live unamplified music. Our firm was requested to visit and offer assistance. Using a click track played over the system and simply listening, it was quickly deduced that turning off some of the loudspeakers would result in significant improvement in speech intelligibility throughout the entire Temple. This paper will discuss various aspects of the background, analysis, and client responses, as well as the historicity of this Temple of the Stars.
Generalized energy density and its significance for sound systems in rooms
Authors: Leishman Timothy W. , Jensen Zachary R.
Published: May 2016

The room constant has long been used for fundamental architectural acoustics predictions, including those related to sound systems. It is an integral part of the Hopkins-Stryker equation, which characterizes semi-reverberant fields via total energy density. While many have used potential energy density instead of total energy density with the equation, modern measurement techniques have facilitated the use of the latter. Recently, generalized energy density has been advanced as a metric with even greater spatial uniformity in reverberant fields and attendant benefits. This presentation will review the quantity and discuss how it may be properly used with the Hopkins-Stryker equation. It will also explain how the quantity may improve direct measurement of the room constant rather than requiring its estimation from approximate reverberation times, room geometrical values, and absorption coefficients. Generalized energy density may likewise be used in conjunction with impulse response measurements, which are beneficial in other ways for sound system measurements and optimizations.
Distributed mode loudspeakers—An enigma or just a misunderstood phenomenon?
Authors: Mapp Peter
Published: May 2016

Distributed mode loudspeakers (DMLs) have been commercially available for nearly 20 years, but they are still misunderstood and often misapplied in audio and sound reinforcement installations. The paper will discuss the history and basic properties of DMLs and look at current examples and their potential application in commercial audio and sound reinforcement. In particular, often quoted claims concerning their coverage, diffuse radiation, and intelligibility will be discussed together with a discussion of appropriate measurement parameters and measurement techniques that may be used to characterize these devices.
What is so special about goldfish?
Authors: Edds-Walton Peggy L.
Published: May 2016

The goldfish (Carassius auratus) was the preferred piscine subject for the auditory research conducted by Dick Fay for many years. As one of the otophysine teleosts, the goldfish has been called an “auditory specialist,” and much of the research Fay and his collaborators conducted on goldfish focused on aspects of the sense of hearing (e.g., frequency selectivity and intensity discrimination), using classical conditioning (respiration or heart rate) and a variety of psychophysical procedures to determine what goldfish could hear. Studies of toadfish (Opsanus tau) and mottled sculpin (Cottus bairdii) provided additional insight on frequency tuning and intensity discrimination in “nonspecialist” teleost fishes and unique insights into how the brain of “nonspecialists” encodes source direction, indicating that goldfish are not so special.
Frequency tuning and directional preferences of eighth nerve afferents in the non-teleost bony fish, Acipenser fulvescens
Authors: Meyer Michaela
Published: May 2016

We investigated the coding mechanisms for spectral analysis and sound source location in Acipenser fulvescens, the lake sturgeon. A. fulvescens belongs to one of the few extant non-teleost ray-finned fishes, with a phylogenetic history that dates back about 200 million years. A shaker system was used to simulate the particle motion component of sound during electrophysiological recordings of isolated single units from the eighth nerve. Data were compared to teleosts and land vertebrates. Peripheral coding strategies for spectral analysis and sound source location in A. fulvescens resembled those found in teleosts. Frequency characteristics generally resembled that of low-frequency afferents of other fishes and land vertebrates and the auditory periphery in A. fulvescens appears to be well suited to encode the intensity of sound. Eighth nerve afferents responded to directional stimuli in a cosine-like manner (as in teleosts). However, differences from teleosts were found that may have implications for the mechanisms for sound source location in azimuth. The common physiological characteristics among A. fulvescens, teleosts, and land vertebrates may reflect important functions of the auditory system that have been conserved throughout the evolution of vertebrates.
Sound source localization by fishes
Authors: Sisneros Joseph A.
Published: May 2016

The ability to locate sound sources enables animals to detect prey, avoid predators, and communicate with conspecifics, and is thus basic to the survival of many vertebrate species. Evidence suggests that the capacity to locate sound sources is common to amphibians, reptiles, birds, and mammals, but surprisingly it is not known whether fish locate sound sources in the same manner. Sound source localization by fishes continues to be an important topic in the hearing sciences, but the empirical and theoretical work on this topic has been contradictory and obscure for decades. Studies of sound localization by fishes have been difficult to conceive of primarily because fish are assumed not to use the same binaural acoustic cues as terrestrial animals, and secondarily, because the dominant theories for sound localization by fishes are rather complex and most fish are thought to be unable to detect the sound cues theoretically necessary for localization. Thus, the question of how fish locate sounds remains an open one. Dr. Sisneros’ talk will describe the work of Fay and Sisneros that address sound source localization in the plainfin midshipman (Porichthys notatus), which has proven to be an exceptional species for fish studies of sound localization.
Efferent command may facilitate special frequency channel in the inner ear for vocal communication of Mongolian gerbils
Authors: Riquimaroux Hiroshi
Published: May 2016

Mongolian gerbil, Meriones unguiculatus, has been a good model animal for hearing, whose thresholds between 1 and 16 kHz are low. The animal also has a variety of repertoire in communication calls. Frequency range for 80% of communication calls was reported to be 30–50 kHz. However, the audiogram indicates that thresholds for 30–50 kHz are not low, not well suited for vocal communication. The present study examines if sensitivity for 30–50 kHz could be improved when necessary. We made a hypothesis that their attention might enhance inner ear sensitivity around 30–50 kHz for vocal communication. Then, frequency sensitivity of inner ear was examined by the cochlear microphonics (CM), comparing attended and unattended conditions. The findings suggest that attention to the companion animal may enhance CM power spectra around 30 kHz, especially for low intensities. Moreover, when a companion animal exists, even CM to short simple tone bursts of 25–40 kHz was enhanced. Findings from noise induced TTS experiments would also imply that CM responses around 30 kHz appeared to be amplified by the hair cell activities. [Work supported by Grant-in-Aid for Scientific Research (B) 26280064 from MEXT of Japan and Start-up grant from Shandong University.]
Overview of the Engineering Acoustics Technical Committee
Authors: Brown David A.
Published: May 2016

The Engineering Acoustics Technical Committee (EATC) and its members work to advance the methods, tools, and instrumentation used in the study of sound in all its disciplines. It is concerned with the development and improvement of measurement techniques, apparatus, new methods, acquisition of data, and the creation of knowledge in the field of acoustics. Some of the common areas involve: production and reception of sound, transducers and arrays, underwater and in-air acoustic systems, audio engineering and related electronics and amplification, instrumentation and recording, hearing aids, medical and industrial ultrasonics, noise monitoring and cancelation, materials and metamaterials. The talk also presents some hot topics in the EATC and some sample student research projects.
On the use of a correction scale for the effect of intruding environmental sources at specified signal to noise ratios
Authors: Rosenhouse Giora
Published: May 2016

The British Standard BS 4142:2014 is a method for rating noise levels by comparing the rating level of the specific noise source with the existing background noise. This approach includes some corrections, but it does not properly consider the difference in frequency contents of both background noise and the examined specific noise. However, taking into account the influence of the spectrum of the intruding noise corrected by a scale that depends on the signal to noise ratio might solve the issue. It appears that the model of S/N influence on speech intelligibility and the influence of environmental background noise on the effect of specified noise source on people may have a common correction scale. If it is true, as measurements show, the human brain uses the same scale of S/N ratio for the perception of both speech and noise effects, as was shown by the author in the ASA meeting in Providence¸ 2014. This paper shows explicitly how the calculation is done in the case of noise compared with the calculation of the effect of S/N ratio on speech intelligibility. It includes a solved case of using of the correction scale.
Accounting for non-zero asymptotic annoyance prevalence rates in community tolerance level analyses
Authors: Mestre Vincent , Horonjeff Richard , Fidell Sandord
Published: May 2016

Dosage-response relationships between transportation noise exposure and the prevalence of noise-induced annoyance typically assume an asymptotic rate of annoyance of zero at low noise exposure levels. If the annoyance prevalence rate of any sigmoidal function at low exposure levels is non-zero, however, its predictions may not closely match field observations in low exposure ranges. Community tolerance level (CTL) analysis yields a dosage-response relationship that is easily modified to account for non-zero asymptotic annoyance prevalence rates. This presentation describes empirical analyses undertaken to investigate the effects on CTL values of field observations of annoyance at low levels of noise exposure.
Numerical wave simulation for interactive audio-visual applications
Authors: Raghuvanshi Nikunj , Allen Andrew , Snyder John
Published: May 2016

Sound is an integral part of many interactive applications, such as 3D computer games. The generality and complexity of the simulated domains makes wave simulation a natural fit. The opposing goals of real-time execution and compute intensiveness of wave simulation provide a rich ground for designing novel computational methods. Two instances of such audio-visual interactive techniques will be discussed. The first technique employs precomputed wave simulation for providing acoustic effects in computer games. Time-domain 3D simulation is performed from many prospective player positions, resulting in a 7D field containing billions of acoustic responses. Perceptual encoding is then performed to drastically reduce memory requirements. Interactive motion of player and sources in the game can then be accommodated via quickly decoding and auralizing the pre-calculated wave effects. The second technique explores the limits of real-time parallel numerical acoustic computation on today's desktop graphics cards. The user is presented with a 2D “sandbox” environment for designing virtual 2D musical wind instruments that are instantly simulated in real-time for the entire human auditory bandwidth. Instrument shape editing and performance via manipulating tone-holes/valves can be done on-the-fly. Video demonstrations of both techniques will be shown along with a discussion of the core technical ideas.
Time-domain propagation in porous media using the auxiliary differential equation method: Application for outdoor acoustics
Authors: Dragna Didier , Pineau Pierre , Blanc-Benon Philippe
Published: May 2016

Time-domain sound propagation equations in porous media usually involve convolutions, which can be computationally cumbersome. Indeed, a naïve approach would require to store the acoustic variables at every time-step, which would lead to an unacceptable memory space for long range propagation. To reduce the computational burden associated to convolutions, an efficient numerical method, referred to as the auxiliary differential equation (ADE) method, originated from the computational electromagnetic community and presented in Dragna et al. [J. Acoust. Soc. Am. 138, 1030–1042 (2015)] is proposed. The idea behind is to approximate the relaxation functions, which depend on the porous media properties, by rational functions in the frequency-domain. The time variation of the convolution is thus given by first-order differential equations which can be solved using standard time-marching schemes. The accuracy of the method is investigated and compared to that of recursive convolution methods. The ADE method is then applied for outdoor sound propagation using the time-domain rigid-frame model proposed by Wilson et al. [Appl. Acoust. 68, 173–200 (2007)] in the ground. Results obtained with Wilson's model are compared at long range to those obtained with Zwikker and Kosten's equations and with an equivalent impedance surface for different flow resistivities.
Rapid computation of waves diffracted by an absorbing plane with precise error bounds
Authors: Li Kai M. , Wang Yiming
Published: May 2016

The current study is motivated by a recent paper [L. N. Trefethen and J. A. C. Weideman, “The exponentially convergent trapezoidal rule,” SIAM Rev., 36, 385–458 (2014)]. It is well known that waves diffracted by an absorbing plane due to a point or line source can be expressed in Fourier integrals. At low frequencies and short ranges, numerical solutions may be obtained by evaluating the integral, for example, by the fast field formulation (FFP). However, it is found prohibitive expensive to use FFP at high frequencies and long ranges. On the other hand, the solutions can be efficiently evaluated along the steepest descent path at all situations. A common approach involves expansion of the kernel function about the saddle point and application of the pole subtraction method. By retaining the first term of the Maclaurin series, this method gives a fairly accurate analytic solution. This paper explores the trapezoidal rule for evaluating the Fourier integral along the steepest descent path with a prescribed precision. An error bound analysis has been conducted to confirm the predicted accuracy. The required computational time is comparable to those calculated by the asymptotic solution except that the trapezoidal rule yields even more accurate numerical results.
An acoustics library for the Modelica multidomain modeling language
Authors: Thompson Stephen C.
Published: May 2016

The Modelica® modeling language is especially well-suited for modeling the complicated electrical, mechanical, magnetic, and acoustical connections within an acoustic transducer or other acoustic system. The Modelica Standard Library includes many of the components that are needed for modeling electromechanical systems, but does not include piezoelectric or acoustical components. This paper describes methods and modeling code to provide an acoustical domain and several methods to include piezoelectric length expander pieces in the Modelica language.
The musical saw and the flexatone: An experimental study of confined vibrational modes in metal plates of variable curvature
Authors: Worland Randy
Published: May 2016

The musical saw and the flexatone are two quite different musical instruments that rely on similar physical principles for their sound production. The musical saw typically consists of an ordinary sawblade bent by the player into an “S”-shape and then bowed, producing a very pure tone. Skilled players produce melodies, vibrato, and glissando effects by dynamically varying the blade’s curvature as well as the bowing location. The flexatone is a percussion instrument in which wooden beaters strike a flexible metal sheet. The resulting sounds can again be modulated by varying the curvature of the metal with slight hand pressure. Both instruments rely on the confinement of vibrational bending modes between regions of blade curvature. These confined vibrational patterns do not extend to the ends of the blade, and are therefore not influenced by the end conditions, including the damping that would otherwise be caused by the player’s hand. The mode shapes and frequencies of these resonances are investigated as a function of blade geometry and curvature, using electronic speckle-pattern interferometry.
Reconstructing the piano hammer force from measurements and filtering of the string velocity
Authors: Chaigne Antoine
Published: May 2016

A method is presented for reconstructing the piano hammer force from measurements and filtering of the string velocity. The construction of the filter is based on the propagation of the pulses generated by the hammer blow in two opposite directions along the string. For the first five octaves of most pianos, and for a measuring point situated between hammer and agraffe, the hammer force can be derived by considering the incoming wave from the hammer and its first reflection by the agraffe only. For the higher notes, 4- or 8-waves schemes must be considered in order to reconstruct the hammer force over the complete range of the piano. The theory is first validated on simulated string velocities, by comparing imposed and reconstructed hammer forces. The simulations are based on a nonlinear damped stiff string model (JASA 134, pp. 648–665). In a second step, the method is applied to string velocities measured of real pianos. Its efficiency is further validated through simulation of the string velocity with the reconstructed hammer force at the input, and comparison with the measured velocity. [Work supported by the Lise-Meitner-Fellowship M1653-N30 of the Austrian Science Fund (FWF).]
Experimental active control of cylindrical shells using the weighted sum of spatial gradients control metric
Authors: Aslani Pegah , Sommerfeldt Scott D. , Blotter Jonathan D.
Published: May 2016

Cylindrical shells are often used in industry and there are many applications in which it is desired to reduce the sound radiated from the vibrating structure. At low frequencies, active structural acoustic control (ASAC) is able to offer solutions for this purpose. The technique of using the weighted sum of spatial gradients (WSSG) as a control metric has been developed previously for cylindrical shells as well as for flat structures. This metric has been demonstrated to be correlated with the radiated sound power and is therefore effective in attenuating the radiated sound. The method has also been shown to be robust with respect to the error sensor location. This paper will discuss the experimental implementation of WSSG on cylindrical shells and show experimental control results that have been obtained. The effectiveness of the method has been investigated by comparing the attenuation of the radiated sound power that is achieved both in the numerical model and the experimental results. The radiated sound power in the model is calculated through using the radiation modes for cylindrical shells, while for the experiment the sound power is measured using the ISO 3741 standard. A comparison of the experimental and numerical results will also be presented. [Work supported by NSF.]
Calculation of flanking sound transmission according to ISO 15712-1—Comparison between the simplified and the detailed method
Authors: Hoeller Christoph , Mahn Jeffrey , Quirt David
Published: May 2016

The 2015 edition of the National Building Code of Canada (NBCC) specifies sound insulation requirements between dwelling units in terms of Apparent Sound Transmission Class (ASTC). The ASTC includes both the direct sound transmission through the separating element between adjacent rooms as well as the sound transmission via flanking paths. One of the ways to establish compliance with the NBCC involves a calculation procedure based on ISO 15712-1, in which the flanking sound transmission is predicted from the measured sound transmission through individual building elements combined with the attenuation at their junction. The calculation can be performed in third-octave bands (“Detailed Method”) or using single-number ratings such as the STC (“Simplified Method”). This presentation will describe the two calculation procedures, before focusing on the differences between. In extended studies at the National Research Council Canada, it was found that the simplified method sometimes leads to misleading results. An alternative method for calculating the ASTC of walls with linings was proposed, which ensures that the simplified method yields more conservative results than the detailed method. To achieve the best possible estimate of the sound insulation performance of buildings systems with linings, the detailed method should be used.
Relationship between perceptual accuracy and information measures: A cross-linguistic study
Authors: Kang Shinae , Johnson Keith
Published: May 2016

Variation in the amount of information transmitted by speech elements correlates with (and helps explain) a number of speech patterns, such as phonetic reduction during production and sound change. However, the cross-linguistic effect of information content on speech perception is relatively understudied. This study fills this gap and investigates the relationship between perceptual accuracy and two information measures in English, Korean, and Japanese. We computed the informativity (weighted negative contextual predictability) and functional load (the contribution of a speech element to lexical contrast) of /p/, /t/, and /k/ in sub-syllabic speech chunks in English, Korean, and Japanese sub-lexical corpora. Native speakers of the three languages identified consonant clusters in VC.CV stimuli in a perception experiment. A logistic model predicting perceptual accuracy from informativity and functional load found that informativity is generally negatively correlated with perceptual accuracy, while functional load is positively correlated. Furthermore, the strength of this correlation is different for the three languages studied. We interpret the result as reflecting differences in syllable structure across the three languages.
American English schwa characteristics in the context of vowel reduction
Authors: Kuo Christina , Weismer Gary
Published: May 2016

The mid-central unrounded vowel schwa has been understood as a vocalic produced with a uniform cross-sectional area of the vocal tract that is correlated with formant frequencies occupying the center of the acoustic vowel space. As such, schwas have also been associated with reduced vowels. Vowel reduction occurs in a variety of contexts and factors, and it is commonly described as centralization toward the middle of the vowel space or a migration of formant frequencies toward those associated with a tube of uniform cross-sectional areas. Nonetheless, little is known regarding the extent to which reduced vowels approximate schwa values. The purpose of this study was to establish a within-speaker reference of schwa in the context of vowel reduction. Two questions were addressed. First, do the acoustic characteristics of schwa correspond to the centroid of vowel space? Second, how do schwa values compare to those of reduced vowels? Formant frequencies of vowels in citation and connected speech tasks were compared to schwa productions for the same speaker. It was hypothesized that schwa can be characterized as a centroid region defined within individual speakers’ vowel spaces. It was further hypothesized that reduced vowels in connected speech would closely approximate this centroid region.
Assessment of the starting point of the Lombard effect
Authors: Bottalico Pasquale , Ipsaro Passione Ivano , Graetzer Simone , Hunter Eric J.
Published: May 2016

The Lombard Effect is an involuntary tendency to raise the level of the voice with an increase in the background noise level. The aim of this study was to determine whether there is a starting point of the Lombard Effect in terms of background noise level (Ln).Twenty subjects were instructed to read a passage with the goal of being understood by a listener at 2m. A loudspeaker, also located at 2m, emitted pink noise. Ln varied in level from 25 dBA to 70 dBA in 5 dB increments. A regression model was fit with segmented relationships to the data, estimating the slopes and the change point(s) in the relationship between the response variable, within-subject normalised SPL, and the explanatory variable, Ln. One change point was identified at Ln 48.3 dBA. The rate of change in speech level below 48.3 dBA of noise was about 0.24 dB per dBA of noise while the rate of change above 48.3 dBA was nearly 0.5 dB per dBA of noise. Hence, a change point was identified in the speaker’s adaptation to the background noise level, which could be said to mark the starting point of the Lombard Effect, as typically described.
Acoustic comparison of /t/ glottalization and phrasal creak
Authors: Garellek Marc , Seyfarth Scott
Published: May 2016

In American English, the presence of creaky voice can derive from distinct linguistic processes, including phrasal creak (prolonged irregular voicing, often at edges of prosodic phrases) and coda /t/ glottalization (when the alveolar closure for syllable-final /t/ is replaced by or produced simultaneously with glottal constriction). Garellek (2015) showed that listeners can differentiate words in phrasal creak from those with /t/ glottalization, which suggests that the creaky voice derived from phrasal creak and /t/ glottalization differ acoustically. To test this, we analyzed vowels preceding syllable-final /t/ in the Buckeye Corpus, which is comprised of audio recordings of spontaneous speech from 40 speakers of American English. Tokens were coded for presence of phrasal creak (prolonged irregular voicing extending beyond the target syllable) and /t/ glottalization (whether the /t/ was produced only with glottal constriction). Spectral measures of voice quality, including both harmonic and noise measures, were extracted automatically using VoiceSauce (Shue et al., 2011), and discriminant analyses were performed. The results indicate several differences among non-creaky vowels, vowels before glottalized /t/, and vowels in phrasal creak. The discussion will focus on how different sub-categories of creaky voice can contrast with one another in a given language.
Acoustic classification of velar fricatives in Assamese
Authors: Redmon Charles
Published: May 2016

Previous work on the acoustic discrimination of fricative contrasts has commonly studied categories which are not held to vary phonemically; instead the focus is placed on accounting for variation in the salience and stability of the acoustic cues mapping onto those categories (cf. Forrest et al., 1988; Jongman et al., 2000; McMurray and Jongman, 2011). The present study, like Jannedy et al. (2015), addresses a set of contrasts—the voiceless posterior obstruents /x, h, kh/ in Assamese—where /h/-voicing and /kh/-lenition processes interact with the discriminability of the velar fricative differentially according to positional (CV, VCV, and VC) and speaker (gender, dialect) variables. Results of principal components logistic regression (PCLR) models applied at both fixed and cumulative time windows over consonant and transition intervals are presented. Overall, /x/-/h/ discrimination was significantly more accurate in intervocalic position (96%) relative to word-initial position (84%). Data from intervocalic productions further suggest a significant dialectal difference in the /x/-/kh/ contrast, with discrimination poorer in Nalbari (68%) than in Jorhat (86%) data, in line with Sarma’s (2012) analysis. Implications for the integration of lexical information by means of phonotactic priors derived from a corpus (Baker et al., 2002) are also discussed.
Automatic classification of English fricatives using cepstral coefficients
Authors: Lilley Jason , Spinu Laura
Published: May 2016

We use a classification tool previously tested on Romanian fricatives to categorize the front (non-sibilant) fricatives of English by place of articulation. Labiodental and interdental fricatives are difficult to distinguish acoustically, posing problems even for human perception. Prior classification work with English front fricatives has not been very successful with this contrast, with correct classification rates ranging from 40 to 60%. The feature set we use for coding the acoustic properties of the fricatives and their following vowels comprise the first six cepstral coefficients (c0–c5). The acoustic features are measured at 10-ms intervals across each segment; the measures obtained are then binned into three contiguous intervals for both the fricative and the vowel, representing the onset, steady state, and offset of each segment. The boundaries between regions are set by using a hidden Markov model to determine three internally uniform regions with respect to their acoustic properties. The mean value of each acoustic feature within each region is obtained; thus, each CV production yields six measurements per coefficient. We are testing this model on fricatives from the TIMIT corpus and classifying them using multinomial logistic regression models. While this investigation is underway, we expect higher correct classification rates than previous work.
Using gated audiovisual speech perception to identify the temporal onset of coarticulation in production
Authors: Redford Melissa A. , Bogdanov Sergei V. , Vatikiotis-Bateson Eric
Published: May 2016

The goal of the present work was to develop a sensitive, non-invasive method for accurately assessing the temporal scope of coarticulation in speech production. Building on work in audiovisual speech perception (Munhall and Tohkura, 1998; Moradi et al., 2013), we used a gating paradigm and human judges to identify the temporal onset of anticipatory lip rounding in simple SVO sentences produced by five adult females. The sentences were gated based on acoustic landmarks from the midpoint of the verb through to the midpoint of the object noun. Full sentences were also included. Judges were asked to decide whether the object noun rhymed with rounded “oop” or unrounded “ack.” Results indicated an earlier correct identification of rounding in the audiovisual condition compared to the control, audio-only condition. The audiovisual judgments also provided greater temporal specificity regarding the onset of coarticulation than could obtained with acoustic measurement or with a video-based kinematic measure. Insofar as human judges appear to anticipate and even benefit from head movement in audiovisual speech tasks (Munhall et al., 2004), our method should prove especially useful for measuring the temporal extent of coarticulation in younger, more rambunctious speakers.
The role of harmonic spectral structure in the cocktail party problem
Authors: McDermott Josh , Popham Sara , Boebinger Dana
Published: May 2016

Harmonicity is believed to provide an important acoustic grouping cue underlying sound segregation, though the mechanisms by which this occur, and its importance in real-world conditions, remain unclear. To test the role of harmonicity in the segregation of speech, we used a modified version of the STRAIGHT methodology for speech resynthesis to manipulate the fine-grained spectral structure of otherwise natural-sounding speech tokens. We then measured the ability of human listeners on two tasks: speech comprehension—of individual speech tokens in isolation, of speech in noise, and of speech presented concurrently with other tokens—and detection of a mistuned harmonic. We tested the importance of harmonicity in both tasks by jittering harmonic frequencies up or down by a small amount, rendering the speech inharmonic. We tested the importance of familiar spectral structure by deleting even-numbered harmonics. This latter manipulation left only the odd harmonics, a spectral pattern that, while harmonic, does not normally occur in natural sounds. Results indicate that harmonic speech excitation helps listeners to correctly group speech signals, aiding their comprehension amid concurrent talkers. However, the benefit does not generalize to unfamiliar harmonic spectra, consistent with the use of familiar spectral templates rather than harmonic structure per se.
Vocal dosimeter devices and their uncertainty
Authors: Bottalico Pasquale , Ipsaro Passione Ivano , Hunter Eric J.
Published: May 2016

Vocal dosimeters may be used to characterize long-term voice production in typical daily activities. The reliability of four vocal dosimeter devices in the estimation of Sound Pressure Level (SPL) and fundamental frequency (Fo) was assessed. The devices evaluated were the Ambulatory Phonation Monitor (KayPENTAX), VoxLog (Sonvox), VocaLog (Griffin Laboratories), and Voice-Care (PR.O.VOICE). Twenty subjects were recorded in a sound booth by means of a head-mounted microphone while using each dosimeter. Subjects were asked to produce a sustained /a/ vowel and to read a text in three different voice styles (relaxed, normal, and raised). On the basis of the difference in the estimation of the parameters between the values acquired from the HMM and the dosimeters, the mean error, and the combined uncertainty were calculated. For the SPL, the dosimeter with the highest mean error was the APM, followed by the VoxLog, the VocaLog, and the Voice-Care. For Fo, the dosimeter with the highest mean error was the VoxLog, followed by the Voice-Care, and the APM (the VocaLog does not monitor Fo).
Acoustics and perception of charisma in bilingual English-Spanish 2016 United States presidential election candidates
Authors: Signorello Rosario
Published: May 2016

The present work studies the acoustic voice profiles and the perception of charismatic traits of two bilingual candidates to the 2016 United States presidential election: Marco Rubio and Jeb Bush. Marco Rubio is a native bilingual English-Spanish speaker. Jeb Bush’s native language is English and he is fluent in Spanish. The objective of the study is to investigate the differences and the similarities in speakers’ acoustic voice profiles of the speakers addressing the audience in English versus in Spanish. The study also aims to assess the differences and the similarities in listeners’ perception of speakers’ charismatic traits while speaking in English vs. in Spanish. Monolingual English, monolingual Spanish, and bilingual English and Spanish listeners participating to the study assessed charisma traits perception through speakers’ voice quality using the Multidimensional Adjective-based Scale of Charisma Perception (MASCharP, Signorello, 2014). Undergoing statistical analysis investigates the influence of individuals’ native language in speakers’ voice acoustic profiles and in listener’s perception of charismatic traits through voice quality. [Work supported by NIH grant DC01797.]
Acoustic detection and tracking of UAS targets
Authors: Reiff Christian G.
Published: May 2016

Acoustic detection using a Minimum Variance Distortionless Response (MVDR) method and tracking of Unmanned Aircraft System (UAS) targets with the MVDR results over time will be presented. The acoustic environment was complex with surf noise, wind, road traffic, air traffic, and aircraft hangar noise. The targets were gas and electric fixed wing UAS as well as electric multi-rotor quadcopters. Scenarios with the UAS target leaving, incoming, and with multiple simultaneous targets are addressed. Tracking results are evaluated through comparison with GPS based UAS ground truth tracks.
The effect of Doppler rate on small aperture bearing estimation
Authors: Swanson David C. , Culver Robert L.
Published: May 2016

Passive estimation of the angle of arrival using a small aperture array outdoors can be adversely effected by high rates of frequency chirping. This is caused by either rapid change in source frequency and/or rapid change in Doppler frequency due to the motion of the source. Consider two microphone signals simultaneously sampled and analyzed for phase difference via FFT. The time difference of arrival across the small aperture array is small compared to the FFT buffer size, so there are slightly different starting and stopping frequencies for each microphone during the FFT input buffer record. However, the chirp (rate of change in Doppler or source frequency) leads to a significant phase error between the two microphones. An upwards chirp moves the estimated bearing erroneously towards 0 degrees while a downward chirp moves the estimated toward 180 degrees. For tracking aircraft from the ground it is not unusual to experience high rates of change in Doppler frequency near the closest point of approach. This paper shows the extent of the error from practical Doppler and source chirps and how to correct for the bearing error.
Trains, transformers, and tinnitus: Case studies in hard-to-detect sounds
Authors: Bommer Arno S. , Olvera Edgar
Published: May 2016

It’s a phone call many acoustical consultants dread: someone driven crazy by an annoying sound. Sometimes, the source is unknown. Sometimes it’s a nearby industry or antagonistic neighbor. Some sounds are intermittent; some are steady. Some are high frequency; others are low and are described as vibration. Most consultants are greatly relieved if they can hear and measure the sound themselves. But the source can still be elusive, especially if the sound is very faint and if there are multiple potential sources. More often than not, the consultant can hear and measure nothing, which gets blamed on his poor hearing and shoddy equipment. A discussion of tinnitus is often received as an accusation of “hearing things.” But is tinnitus (or a similar condition) the only possible explanation? Why is the sound heard only at home? What triggered the person to start hearing it? The person may be especially sensitive, but to what? What advice is useful? The authors will present several case histories where community sounds were ultimately identified. For those sounds not identified, the author will discuss a number of possible explanations along with recommendations for the consultant and the person hearing the sound.
The singing parking garage: Wind induced Helmholtz resonation of perforated metal panels
Authors: Miller-Klein Erik
Published: May 2016

An above ground parking garage with perforated metal facade experienced high amplitude sound when northwest winds exceeded 10 mph around this building. The noise only occurred under very specific weather conditions, which required on-call noise and vibration measurements to identify the cause and feasible solution. The noise was traced to the Helmholtz resonator effect caused by the perforated metal panel holes and the air cavity associated with the headlight screens behind these perforated facade panels. Through our on-site investigation and analysis, the root cause was identified and a low cost remediation plan was designed and implemented.