bluuebunny/crossref_metadata_2025 huggingface dataset:
Acoustic Scattering Cancellation: an Alternative to Coordinate Transformation Scattering Reduction
Authors: Martin Theodore P. , Rohde Charles A. , Guild Matthew D. , Naify Christina J. , Calvo David C. , Orris Gregory J.
Published: May 2016

Coordinate transformations have been extensively studied as a means to hide an object immersed in a fluid from an acoustic field. The coordinate transform approach relies on coatings with non-uniform and typically anisotropic material properties to guide an acoustic wave around the object. Acoustic scattering cancellation is an alternative approach that relies on modal cancellation within a desirable bandwidth to significantly reduce the acoustic scattering cross-section of an object. We present a direct comparison, using coatings with similar layer thicknesses and material properties, demonstrating that scattering cancellation achieves a similar level of scattering reduction compared to coordinate transforms, but with much thinner coatings. Furthermore, we present experimental results demonstrating significant reductions in the scattering cross-section of neutrally buoyant elastic cylinders suspended in an aqueous environment. Omnidirectional scattering reduction is obtained using a single isotropic coating through modal cancellation of the monopole and dipole modes of the cylinders. The modal cancellation is achieved by carefully tuning the coating’s material properties and thickness such that the combined cylindrical object and coating exhibit effective medium properties identical to water. The scattering cancellation is close to 15 dB over a broad bandwidth for all reduced frequencies below ka ≅ 1.
Synthesis of wave surface by frame loudspeaker array with limited spatial frequency band
Authors: Ando Akio , Yamabayashi Daisuke , Yamoto Takuya
Published: May 2016

We have studied a frame loudspeaker array for sound reproduction of three-dimensional television aiming at realizing clear localization and better sound depth control. Because such array has no secondary sources in its center region, the reproduced wave surface is sometimes deteriorated even if the least squares optimization will be used to reshape the surface. Over the past few decades, a considerable number of studies have been made on the sound field reproduction by loudspeaker array. Recently, higher order ambisonics (HOA) and near-field compensated HOA (NFC-HOA) attract attention of the researchers. It is well known that these methods can limit a spatial frequency band by controlling the degree of spherical function. If such limitation is introduced to the frame loudspeaker array, the deterioration of wave surface will be alleviated because the optimization will be easy to progress by matching only the low spatial frequency components of the reproduced wave surface with those of the reference. The computer simulation showed that the least squares optimization with the moderate restriction of spatial frequency band brings better wave surfaces than that without the restriction.
Acoustic time reversal in granular media
Authors: Harazi Maxime , Yang Yougu , Fink Mathias , Tourin Arnaud , Jia Xiaoping
Published: May 2016

In a non-dissipative medium, the wave equation is symmetric in time. Therefore, for every wave diverging from a pulsed source, there exists a wave that retraces all its original paths in a reverse order and converges at the original source. In the early nineties, M. Fink proposed a method for generating such a time-reversed wave. This method was first tested with ultrasound and then successfully extended to other types of waves such as microwaves, water waves, and even in optics. Several studies have shown that time reversal wave focusing is very robust to disorder. Here, we investigate time reversal (TR) of elastic waves propagating in fragile granular media consisting of glass beads under static compression. Pulsed elastic waves transmitted from a compression or a shear wave source are measured, time reversed, and back-propagated. The ability of the time-reversed wave to focus at the initial source is checked as a function of the source amplitude. We find that TR of the ballistic coherent wave is very robust to perturbations but provides poor resolution. By contrast, the short-wavelength scattered waves offer a finer TR focusing but are sensitive to rearrangements induced by the forward propagation wave itself: at large source amplitudes, time reversal focusing is broken, due to sound-induced rearrangements but without visible grain motion. Experimental results are confronted with predictions from a numerical model in which the propagation medium is modelled by a percolating network of masses interacting via linear springs.
The contribution of amplitude modulations from speech and competing noise sources for speech recognition
Authors: Fogerty Daniel
Published: May 2016

Speech is often heard in the presence of background noise during everyday listening conditions. Concurrent amplitude modulations of the speech and the interfering noise result in portions (i.e., “glimpses”) of the speech signal that remain preserved at favorable signal-to-noise ratios while other portions of speech may be highly degraded. The acoustic and phonetic properties preserved during these glimpses play a role in determining the overall sentence intelligibility. Under such conditions, vocalic cues are relatively more preserved than consonantal/obstruent cues due to average differences in intensity. Investigations examining these different acoustic cues suggest important perceptual contributions from preserved speech amplitude modulations and non-simultaneous interactions from competing noise amplitude modulations. Results from several studies suggest that conditions that maximize the preservation of the sentence temporal envelope (e.g., during long glimpses or for vocalic intervals) result in higher levels of speech recognition. Amplitude modulation during noise-dominated intervals also impacts speech recognition performance. Overall, results suggest that the relative preservation and interaction (e.g., modulation masking or perceptual restoration) of amplitude modulations from the speech and noise is crucial for predicting speech recognition during noise interruption. [Work supported by NIH and ASHA.]
Interrupted speech with competing talkers: Benefits of temporal envelope and periodicity cues for younger and older adults
Authors: Bologna William J. , Chatterjee Monita , Dubno Judy R.
Published: May 2016

Recognition of interrupted speech requires connecting speech fragments over time and across gaps of missing information. Intelligibility improves when silent intervals are filled with noise; the effect is enhanced when noise contains rudimentary speech information, such as the temporal envelope of the missing speech. In multiple-talker environments, recognition of interrupted speech is more difficult, particularly for older adults. In these cases, temporal envelope cues may provide an important scaffold for auditory object formation. Other basic speech cues, such as F0-related periodicity, may help listeners segregate multiple voices and provide additional benefit. The relative, and potentially additive, benefit of temporal envelope and periodicity cues, and their use by older adults, remain unclear. To address these questions, younger and older adults with normal hearing listened to sentences in quiet and competing talker backgrounds. Sentences were periodically interrupted with (1) silence, (2) envelope-modulated noise, (3) steady-state pulse trains, which contained periodicity information from the missing speech, or (4) envelope-modulated pulse trains, which provided both envelope and periodicity information. Results are discussed in terms of contributions of temporal envelope and periodicity cues to perceptual organization for younger and older adults listening in complex environments. [Work supported by NIH/NIDCD and a AAA Student Investigator Research Grant.]
Effects of adult aging on perception of alternated speech
Authors: Wingfield Arthur
Published: May 2016

In 1954, Cherry and Taylor found that when presentation of a continuous speech message was rapidly alternated between the two ears, intelligibility progressively declined as the rate of alternation was increased up to 3–4 switching cycles/s (167–125 ms per ear). Intelligibility then improved as switching rates were increased beyond this point. This V-shaped function might occur if a finite time is required to switch attention from one ear to the other, during which time no usable information is available from either ear. Hence, the more frequently the speech is alternated between ears, the greater will be the cumulative loss of acoustic information. At alternation rates beyond 125 ms per ear the listener begins to adopt a strategy of attending to the interrupted signal from only one ear, relying on the redundancy of the large number of small speech segments to reconstruct the message (Miller and Licklider, 1950). We report data supporting the alternative view that the critical dimension underlying the point of minimal intelligibility is speech content per ear rather than time per ear. We then use this phenomenon as a test of whether adult aging results in a slowing of attentional shifts in audition.
Effect of language experience on the intelligibility of interrupted speech
Authors: Smiljanic Rajka , Sheft Stanley , Chandrasekaran Bharath , Shafiro Valeriy
Published: May 2016

Signal distortions are more detrimental to speech perception by native than non-native listeners. We investigated how speech clarity and semantic context influence the perception of interrupted speech. Native and non-native American English listeners heard semantically meaningful or anomalous sentences produced as conversational or “clear” speech, gated at different rates (0.5–16 Hz). Results showed that both semantic context and speech clarity had a significant rate-dependent impact on the intelligibility of interrupted speech. In general, intelligibility was higher for native than non-native listeners. However, the magnitude of the clear-speech benefit varied across the two listener groups. The clear-speech benefit was obtained for gating rates of 2 Hz and above, except for meaningful sentences with native listeners where the benefit begin at 1 Hz. Both listener groups were able to use contextual information but native listeners derived more benefit at lower gating rates, indicating greater ability to access semantic information with limited acoustic-phonetic input. The results suggest that the non-native speech-perception deficit in adverse listening conditions may in part be due to their less efficient use of compensatory information at higher levels of language processing.
Neuronal oscillations in decoding time-compressed speech
Authors: Ghitza Oded
Published: May 2016

At the core of oscillation-based models of speech perception is the notion that decoding is guided by parsing. In these models, parsing is executed by setting a time-varying, hierarchical window structure synchronized to the input. Syllabic parsing is into speech fragments that are multi-phone in duration, and it is realized by a theta oscillator capable of tracking the input syllabic rhythm, with the theta cycles aligned with intervocalic speech fragments termed theta-syllables. Prosodic parsing is into fragments that are multi-word in duration, and it is realized by a delta oscillator capable of tracking phrase-level prosodic information, with the delta cycles aligned with chunks. Intelligibility remains high as long as the oscillators are in sync with the input, and it sharply deteriorates once they are out of sync. In the pre-lexical layers, decoding is realized by a cascade of neuronal oscillators in the theta, beta, and gamma frequency bands, with theta as “master.” This talk reviews a model that utilizes this cortical computation principle, capable of explaining counterintuitive data on the intelligibility of time-compressed speech hard to explain with conventional models of speech perception. [Work supported by AFOSR.]
Collaboration and results in research on speech motor control
Authors: Perkell Joseph S.
Published: May 2016

Research on speech motor control was conducted in the Speech Communication Group in R.L.E. at M.I.T. between 1965 and 2012. Virtually all of this work was collaborative—with participation of group members and colleagues from other institutions in the United States and abroad. The work dealt with various aspects of speech production, including properties of the production and perception mechanisms and their influences on observed behaviors and control strategies. Examples will be presented that illustrate the indispensable roles of collaborators in studies of motor equivalence, relations between production and perception and the roles of feedback and feedforward control. Collaborators’ contributions were based on their expertise in a number of areas, including: electrical and biomedical engineering, experimental psychology, acoustic phonetics, speech perception, statistical analysis, neuro-computational modeling, and design of sophisticated hardware and software. These examples demonstrate how such collaborations were essential to a productive output of research findings and theoretical advances for over four decades. [Research supported by NINCDS and NIDCD, NIH.]
Bayesian model selection for a broadband coprime array with unknown number of broadband sources
Authors: Bush Dane R. , Xiang Ning
Published: May 2016

Coprime microphone arrays use sparse sensing to achieve O(MN) degrees of freedom using only O(M + N) elements, where M and N are coprime integers. The benefit is a narrow beam at frequencies higher than the spatial Nyquist limit allows, with residual side lobes arising from aliasing. These side lobes can be mitigated when observing broadband sources [D. Bush and N. Xiang, J. Acoust. Soc. Am., 138, 447–456 (2015)]. Peak positions indicate directions of arrival in this case; however, uncertainties on number of concurrent sound sources in practical applications challenge classical approaches to direction-of-arrival estimations. One has to first resolve the uncertainty on how many sources are present. In this work, Bayesian inference is used to first select which model the data prefer from competing models before estimating model parameters, including the particular directions of arrival. The model is a linear combination of modified Laplacian distributions (one per sound source). The posterior probability function is explored over the entire parameter space by nested sampling in order to evaluate Bayesian evidence for each model. Bayesian evidence is crucial for resolving the uncertainties regarding number of sources, preferring simpler models while penalizing unnecessarily complicated models in an inherent implementation of Occam’s razor.
Design and tests of an acoustoelectric logging tool
Authors: Lu Junqiang , JU Xiaodong , Zhao Honglin , Men Baiyong , Duan Wenxing , Qiao Wenxiao
Published: May 2016

The acoustoelectric logging is a developing method, and the logging method can interpret the seepage properties of the pore formation especially, such as permeability and so on. A new acoustoelectric logging tool is presented. The tool is composed of two acoustic transmitting transducers, three acoustic receiving transducers, two excitation electrodes, and four receiving electrodes. The transmitting transducers can work in a phased mode, and more radiant energy can be effectively generated. The acoustic radiant energy is a very important factor for the acoustoelectric logging tool, and measurements are conducted in an anechoic tank with dimensions of 5.0 m × 5.0 m × 4.0 m. The pulse width of excitation signals is 70μs, and the peak value is above 3000 V. The main frequency of receiving waves is 9.52 kHz, sound pressure is 47.2 kPa, and transmitting voltage response level is 147.5 dB. The tool is tested in the experimental wells and exploration wells. While the acoustic transmitting transducers radiate acoustic excitation signals, acoustoelectric signals from interfaces and concomitant acoustoelectric signals with acoustic waves are all acquired in clastic rock formations.
Practical estimates of the coherent leakage of sound from a mixed layer surface duct
Authors: Jones Adrian D. , Duncan Alec J. , Zhang Zhi Y.
Published: May 2016

In a mixed-layer surface duct, acoustic transmission at frequencies below that required for duct trapping, as well as at frequencies moderately higher, is subject to coherent leakage of energy to the thermocline. The rate of coherent leakage with range is related to the imaginary part of the horizontal wavenumber of a mode, for which an iterative technique is commonly used for evaluations. In order to obtain alternative direct analytic expressions of the leakage rate, the analysis of Furry, described, for example, by Pederson and Gordon (JASA 47, 304–326, 1970), has been revisited. Consideration has been given to both very large and very small ratios between sound speed gradients in the duct and below the duct, and ratios of sound speed gradients typical of those encountered as sea. The leakage rate for the first mode was found to well approximate the total signal leakage, hence the proposed analytical expressions relate to the first mode. Leakage values based on these expressions are compared against results from both a modal model and a wavenumber integration model, and a pre-existing approximate expression for leakage, for a number of surface ducted scenarios for frequencies relevant to the onset of duct trapping.
Examination of rough surface scattering from water/air interface using hybrid parabolic equation model
Authors: Aslan Mustafa , Smith Kevin B.
Published: May 2016

Many ocean acoustic propagation models assume an idealized pressure-release boundary at the ocean surface. This is easily accomplished numerically using finite element, finite difference, and split-step Fourier techniques. For models based on the split-step Fourier algorithm, rough surfaces can also be treated, but require additional complexity in the definitions of the field and propagator functions through a field transformation technique. For this reason, it may be advantageous to model a rough water/air interface in a manner analogous to the bottom treatment by extending the calculation into the air medium, thereby simplifying the definitions of the propagator functions. However, standard approaches to treat density discontinuities in split-step Fourier algorithms invoke smoothing functions, which have been shown to introduce phase errors in range. In this work, the hybrid split-step/finite-difference approach introduced by Yevick and Thomson (1996) is implemented in the Monterey-Miami Parabolic Equation (MMPE) model for both the water/sediment and water/air interfaces. Particular attention is paid to comparisons between the rough surface scattering results from the field transformation technique (applied to a pressure release surface) and the hybrid split-step/finite-difference approach (applied to the water/air interface).
Potential impacts of climate change on acoustic propagation in the Arctic
Authors: Duda Timothy F. , Freitag Lee E. , Adornato Lori R. , Byrne Robert H.
Published: May 2016

Some forecasts show surface pH in the Arctic dropping from 8.1 to 7.6 over the next 100 years. This substantial decrease may cause changes in acoustic transmission at frequencies where the pH-dependent borate absorption plays a role, below about 5 kHz. In many cases, upward refraction of sound in the near-isothermal Arctic waters causes ice or surface scattering effects to dominate transmission. However, recent observations in the Canada Basin show that 700 and 900-Hz sound can be fully ducted beneath the Pacific Summer Water, with no ice interaction, and be detectable over distances of a few hundred kilometers. In this situation, the received signal level is controlled largely by cylindrical spreading and absorption. Here, for a wide band of frequencies, the effects of probable pH reductions and reduced absorption are investigated using a few models of pH depth profiles. There is potential for increased signal levels of 5 dB or more for 200-km propagation if the duct waters have significantly reduced pH.
Comparison of interior noise levels produced by rain impinging on several commercial roof constructions
Authors: Pippitt Logan D. , Huey Michelle L. , Coffeen Robert C.
Published: May 2016

Noise caused by rain on commercial roofs is an ever-occurring problem. Due to climate changes, rainfall seems to have become more sporadic and intense, making the effects of rain noise on the interior environment an acoustical issue of intensifying importance. Noise produced within architectural spaces by rain on roofs is difficult to quantify due to varying rain intensity, water droplet size, droplet velocity, roof construction, and interior acoustical characteristics. However, it is possible to compare interior rain noise levels when rain conditions and the interior space conditions are constant. This paper compares (1) rain noise levels for several commercial roof constructions with and without suspended ceilings beneath the roof, (2) rain noise levels with roofs using typical rigid foam insulation versus mineral wool insulation. Noise level measurements and rain generation are in general conformance with ISO 140-18 “Laboratory measurement of sound generated by rainfall on building elements.” The rain noise levels are presented along with similar rain noise measurements made in the same test facility that were the subject of a 2007 ASA paper. This research was conducted through the School of Architecture, Design and Planning at the University of Kansas.
A method to evaluate nonideal effects of anechoic chambers on multiple-angle measurements
Authors: Denison Michael H. , Bodon K. J. , Leishman Timothy W.
Published: May 2016

Anechoic chambers are typically qualified by comparing sound pressures at several radial distances from a sound source and verifying that they follow the spherical spreading law within specified tolerances. While this technique is useful, it may not sufficiently characterize free-field variations at fixed radial distances and numerous angular positions, as are commonly used for directivity, sound power, and other important acoustical measurements. This paper discusses a technique to detect angular field deviations in anechoic chambers. It incorporates a loudspeaker in an altazimuth mount, an adjustable-radius boom arm, and a precision microphone. The boom arm and microphone remain in line with the loudspeaker driver axis at a fixed radius while the system rotates to specified polar or azimuthal angle increments. In an ideal free-field environment, the frequency response function from the loudspeaker input to the microphone output should remain consistent—regardless of the system orientation. However, in typical anechoic chambers, they vary. Standard deviation calculations over many angles reveal frequency-dependent departures from the ideal, especially for narrow-band data. The results show the impact of these discrepancies for multiple-angle measurements and how they change with radial distance from the source.
Acoustical evaluation of residential laneways and laneway housing
Authors: Lin Rosa , Connelly Maureen
Published: May 2016

Laneway housing (LWH), a free-standing, small wood-frame house (1 to 1.5 stories high and less than 900sf in floor area), is an increasingly popular product of Vancouver B.C.’s high-density urban growth policy. LWH is subject to multiple noise concerns due to its small architectural form and siting in laneways typically designed for garage access and utilities. (Laneways are approximately 16 ft wide and 12 to 18 ft high.) Case study investigations examined residential laneways as potential urban canyons and the acoustical performance of LWH envelope. Methods employed include ASTM field measurements, software models (Odeon and AFMG Soundflow), and the National Research Council of Canada traffic noise model (comparable to the FHWA model). Metrics evaluated include rate of attenuation over propagation distance through a laneway, façade transmission loss, and room absorption. Results show that certain laneways function as urban canyons, measuring nearly 20 dBA higher in SPL at the LWH façade than in laneways without urban canyon characteristics. Indoor SPL due to traffic noise transmitting through the building envelope were above 45 dBA. At least half of the case studies investigated did not meet noise criteria for residential health. These investigations emphasize the need for acoustical specifications in design and construction guidelines for LWH.
A hybrid method combining the edge source integral equation and the boundary element method for scattering problems
Authors: Martín Román Sara R. , Svensson U. Peter , Šlechta Jan , Smith Julius O.
Published: May 2016

A hybrid method for acoustic scattering problems is studied in this paper. The boundary element method is combined with a recently developed edge diffraction based method [J. Acoust. Soc. Am. 133, pp. 3681–3691, 2013]. Although the edge diffraction method has been shown to provide accurate results for convex, rigid objects at a very attractive computational cost, it has some numerical challenges for certain radiation directions. The hybrid method suggested here has a similar structure as the boundary element method (BEM): in a first step, the sound pressure is calculated on the surface of the scattering object, and in a second step, the scattered sound is obtained at any external receiver point. In this method, the edge diffraction based method is used for the first step, and then, the calculation of the scattered sound is performed à la BEM by means of the Kirkhoff–Helmholtz Integral equation. Several benchmark cases are studied, and the results are compared with different reference methods.
Resonant mode characteristics of a cajón drum and its effect on sound directivity
Authors: Denison Michael H. , Bodon K. J. , Gee Kent L.
Published: May 2016

The cajón is a hand percussion instrument originally from Peru. Spanish for “large box” or “crate,” the cajón is typically a box enclosure with a thin wooden plate at the front that serves as the playing surface. Many resonance modes are excited when a player strikes the front plate, as recently studied by Pehmoeller and Ludwidgsen [J. Acoust. Soc. Am. 138, 1935 (2015)]. In this study, scanning laser Doppler vibrometry is coupled with high-resolution far-field directivity measurements to examine source vibration and the resultant sound field. Ties between the plate modal vibration and far-field sound radiation spatial patterns are shown. In particular, the non-uniform front plate normal velocities at higher-order resonances result in distinct nulls in the far-field directivity. Avoidance of these nulls yields recommended microphone locations for cajón sound amplification.
Instrument identification and blending in vinyl records during the transition period from jazz to rock music
Authors: Braasch Jonas
Published: May 2016

Arnold Schering and others have partitioned music genres into traditions where the identification of individual instruments (Spaltklang) is the goal and others where instruments are blended to a homogeneous sound (Verschmelzung). Baroque music is often cited to emphasize the identifiability of individual instruments, whereas the fusion of orchestral instruments became one of the major goals in the romantic period. Historical vinyl record releases were used to investigate the extent to which the theory of identification and blending can be applied to 20th century popular music based on the fundamental principles of Auditory Scene Analysis. The identifiability of instruments was one of the major concerns for jazz combos, for example, in the Miles Davis sextet of the late fifties. Blending became more important in many types of rock music to create a huge sound, for example, in Phil Spector’s Wall of Sound of the 1960s. Artificial reverberation, dynamic range compression, and other effects were now used to create loud and fused sound fields. Based on a psychophysical loudness model, it will be discussed how both approaches lend themselves to create loud sounding mixes to maximize the achieved perceived loudness for a given medium-dependent maximum signal level.
Predicting the acoustical properties of 3d printed resonators using a matrix of impulse responses and mode interpolation
Authors: Michon Romain , Granzow John
Published: May 2016

Accurately predicting acoustical properties of 3D printed models is of interest to instrument designers who explore novel geometries. We introduce a technique to carry out these estimates using a database of impulse responses and mode interpolation. 3D models are organized as a function of their physical characteristics and placed into a multidimensional space/matrix. The models at the boundaries of this space define the limits of our prediction algorithm and they are produced using 3D printing. Impulse responses of these models are measured, and modal information is extracted from each object. Mode parameters are interpolated within the matrix to predict the frequency response of unprinted models that fall within the geometrical space of the test matrix. A physical model using modal synthesis also allows us to listen to the resulting resonator.
Review study of main sources of noise generation at wheel-rail interaction
Authors: Li Qinan , Alemi Mohammad Mehdi
Published: May 2016

Railways are very economical and efficient long-distance transportation systems in many high-density populated countries. However, in terms of environmental safety, different countries announced an increasing traffic noise near residential area. The relative importance of wheel-rail noise to overall train noise plays a crucial role in developing railway transportation. Generally, there are three types of noise at wheel-rail interaction: rolling noise, impact noise, and squeal noise. Rolling noise refers to vertical vibration excitation on straight track due to the undulations of the wheel and rail surfaces. TWINS as the most advanced model up to date can provide total noise prediction within 2 dB compared to full-scale experiments. The impact noise happens when wheel is running on the rail surface discontinuities. Since linear noise generation assumption is employed at contact patch and non-linear contact element is ignored, impact noise due to large roughness are rarely developed and they need to be extended to the reasonable higher frequencies (5 kHz). The squeal noise is usually due to lateral excitation mechanism and always occurs on sharp radius curves. Due to the complexity of wheel-rail noise generation mechanisms, the almost all existing prediction models either have not been validated or they are bounded to predict noise under limited conditions and certain frequency ranges. In this paper, the differences between three main sources of noise generation at wheel-rail interaction will be reviewed.
Using helium as the working fluid in thermoacoustic engines
Authors: Wells Nathaniel , Andersen Bonnie
Published: May 2016

The efficiency of thermoacoustic engines can be improved by using helium as the working fluid because it reduces viscous losses, has a higher thermal conductivity and speed of sound. The engine in this study has a bottle-shaped resonator. The neck consists of a brass cylinder, closed at the top end and a copper cylinder, open at both ends, with copper mesh screens heat exchangers between them (ID of 1.91 cm and total length of 5.24 cm). A small amount of steel wool (20 mg) functions as the stack. The neck opens into an aluminum cavity (10 cm long with an ID of 4.13 cm). A combination of two types of heat-shrink tubing and Teflon were used to connect the brass and copper pieces. The engine was evacuated of air and backfilled with helium as much as the setup would tolerate. Using an input power of 14.8 W over intervals of 0.5–3 h, it was observed that the frequency decreased in time, indicating that the helium was leaking out slowly. From the frequency data, the volume fraction of helium was calculated, indicating that the engine was able to achieve 64% volume fraction of helium and decreased to 6%. The intensities of the sound over this range of volume fractions averaged at 155 W/m2 compared with air at an average of 118 W/m2.
Quasi-emulsion between water and cavitation bubble cloud in an ultrasonic field
Authors: Bai Lixin , Lin Weijun , Deng Jingjun , Li Chao , Xu Delong , Wu Pengfei , Chen Lishuo
Published: May 2016

The cavitation bubble distribution (cavitation structure) is spatially inhomogeneous in ultrasonic field. A quasi-emulsion phenomenon of cavitation structure is reported in this paper. The inception process of cavitation bubble cloud in a thin liquid layer (the thin liquid layer is trapped between a radiating surface and a hard reflector) was investigated experimentally with high-speed photography. It is revealed that cavitation bubble cloud can be considered as a uniform fluid (cloud), and water without cavitation can be considered as another uniform fluid (water). The conversion from cloud-in-water emulsion to water-in-cloud emulsion occurs with the increase in the number of bubbles. The formation and stability of cloud-water emulsions is analyzed theoretically. It is found that surface tension of cavitation bubble cloud played a leading role. Findings of this research proved that cavitation bubble clusters can be considered and investigated as a whole. [This work was supported by the National Natural Science Foundation of China (Grant No. 11174315).]
Listener head motion can degrade spatial selective auditory attention
Authors: Macpherson Ewan A. , Ellis Blair K.
Published: May 2016

Listeners can use spatial selective auditory attention (SSAA) to focus on one talker in a complex acoustic scene. Although the dynamic binaural cues generated by listener head motion improve sound localization, and might therefore enhance the perceptual differences between separated targets and distractors, the effect of head motion on SSAA is unknown. We measured listeners’ ability to attend, with and without head motion, to a frontal target in the presence of two symmetrically separated distractors. During stationary trials, listeners visually fixated and oriented toward a target loudspeaker. On head-motion trials, listeners oscillated their heads at ~0.5 Hz with an amplitude of ~±40° while continuously directing their gaze toward the target. On each trial, three equal-intensity sequences of four spoken digits were presented simultaneously as target and distractors. Listeners reported the target sequence heard. With distractors at ±22.5°, 86% of target digits were reported correctly without motion, but only 72% were reported correctly with head motion. Correspondingly, the percentage of distractor digits reported as targets increased from 11% to 23%. For widely separated distractors, there was no performance penalty for head motion. These results suggest that listeners cannot rapidly update the focus of their SSAA to compensate for head motion.
The perception of reverberation is constrained by environmental statistics
Authors: Traer James , McDermott Josh H.
Published: May 2016

Human sound recognition is robust to reverberation. We explored the hypothesis that this robustness is rooted in the ability to separate the contributions of a sound’s source from that of reverberation. As the separation of source and filter from their convolution is inherently ill-posed, any such capacity should depend on prior assumptions about the nature of filter and/or source. We measured the distribution of real-world environmental impulse responses (IRs) and tested whether it constrains the ability of listeners to estimate source and filter from reverberant audio. We surveyed volunteers about the spaces they encountered during daily life, and measured IRs at each location. We found that the tails of real-world IRs decay exponentially, with decay rates consistently slower at low frequencies. We then synthesized IRs that were either faithful to, or deviated from, the observed distribution of real-world IRs. We assessed (a) the sense of reverberation conveyed by IRs, (b) discrimination of novel sound sources in reverberation, and (c) discrimination of IRs given only their convolution with sound sources. We found that human listeners can separately estimate the source and filter in reverberant conditions, but are strongly constrained by whether the filter conforms to the naturally occurring distribution.
Interaction of pitch and phonation in Louma Oeshi
Authors: Gruber James , Lew Sigrid
Published: May 2016

Louma Oeshi is an Akoid (Tibeto-Burman) language of Laos for which acoustic characteristics are undocumented with the exception of preliminary work by the present authors. This study focuses on the phonetic properties associated with Oeshi's three tones (high, mid, and low) and two registers (Tense, Lax), which fully intersect to yield a six-way suprasegmental contrast. Eight speakers were recorded in Phongsali Province, Lao PDR. Each spoke a 100 token word list, for which 30 tokens were repeated in a carrier sentence placing the token between high and mid-toned lax words. Decile measures of F0 and an array of measures reflecting phonation types (H1-H2, H1-A1, H1-A3, HNR, and SHR) were performed in Voicesauce (Shue et al. 2011) to capture dynamic values over the syllable duration. Our findings show a reliable three-way F0 contrast between tones and a single interaction with register such that High Tense words consistently fall (tones are otherwise level). Acoustic correlates of the Tense~Lax distinction are less clear. The picture that emerges is one where Tense register has variable phonetic manifestations — preglottalization of onsets, vocalic creaky voice, or a glottal stop coda — and the seemingly inconsistent acoustic results reflect these variable articulatory timing strategies.
Long-term average spectrum of te reo Māori
Authors: Sinex Donal , Maclagan Margaret , King Jeanette
Published: May 2016

Te reo Māori, the language of the indigenous people of New Zealand, lacks some consonants common in English, notably the fricative /s/ and voiceless plosives. However, Māori pronunciation has adapted over time and exposure to English. Thus, there is reason to expect that the spectrum of spoken Māori differs from that of New Zealand English, but that the magnitude of the difference may be decreasing. We measured the long-term average spectrum (LTAS) of the speech of Māori-English bilinguals, using recordings from a longitudinal study of Māori speech. The oldest talkers in the database were born in the late 19th century. Female and male, talkers, and both native and non-native speakers of Māori were included. The LTAS was determined for each talker and each language. For individual talkers, the average spectrum had consistently higher amplitude for English than Māori, for frequencies above approximately 2 kHz. Across talkers, the mean difference at 8 kHz was approximately 8 dB, for all but one group. The exception was younger female Māori L2 talkers, for whom the difference was only 3 dB. These observations are consistent with the consonant inventory of te reo Māori, and they also provide potentially useful information about ongoing changes in the language.
A preliminary analysis of stop codas in Kandze Khams Tibetan
Authors: Leongue Vitor
Published: May 2016

Research on Tibetan linguistics has been scarce. There are very few systematic accounts of the sounds in the language, and acoustic-phonetic analyses, in particular, are very difficult to come by. Furthermore, much of the existing literature has focused on the Lhasa dialect; many other varieties are not as adequately described. The current study investigates the phonetic realization of coda consonants in the under-documented Kandze dialect of Khams Tibetan. Historically, Tibetan allows up to two consonants in coda position, as reflected in the written language and in some western varieties. In most modern Tibetan dialects, however, many of the codas have been lost or simplified in some way. Preliminary data from one male native speaker indicate that codas that were historically voiced oral stops are realized in Kandze as glottal stops word-finally. When occurring word-internally, these codas tend to undergo deletion, which may then trigger lengthening of the preceding vowel. In both final and internal positions, some of the codas may also trigger vowel fronting. Acoustic data are presented to illustrate the alternations exhibited by the codas and the vowels that precede them.
The status of voiceless nasals in Miyako Ryukyuan
Authors: Ford Catherine , Tucker Benjamin V. , Ono Tsuyoshi
Published: May 2016

The current study investigates voiceless nasals in Miyako, a language spoken on remote Japanese islands near Taiwan, focusing on the dialect Ikema. Voiceless nasals are rare cross-linguistically. Hayashi (2013) suggested that there is phonemic contrast between voiced /n, m/ and voiceless /n̥, m̥/. Pilot data from one speaker both confirmed and refuted Hayashi’s claims, analysis indicating that these voiceless nasals occur both as voiceless and breathy voiced. Thus, the glottal state of this phoneme may depend on the phonetic environment. The current study further explores these findings based on data collected from four male and two female speakers. Participants were asked to produce target and minimal pair words in isolation and within novel sentences. Voiceless and voiced nasal segments were compared and analyzed for duration, cepstral peak prominence, and amplitude measures of H1 and H2 to determine whether breathy voicing state might differentiate target phonemes from /n/ and /m/. The results of this analysis are used to investigate the nature of this phonemic contrast and how Ikema is situated cross-linguistically in its use of voicing to distinguish nasal phonemes.
On the presence of voiceless nasalization in apparently effaced Somali Chizigula prenasalized stops
Authors: Temkin Martinez Michal , Boone Haley K.
Published: May 2016

Somali Chizigula (G311; xma; also Mushungulu) is an endangered language spoken in Somalia and by Somali-Bantu refugees abroad. We report on an acoustic and aerodynamic study of N + stop sequences in Somali Chizigula. Our findings illustrate that while the only acoustic cue for the nasal in voiceless prenasalized stops is aspiration of the oral stop, appearing to have undergone total effacement, aerodynamic data show robust nasal airflow prior to the oral stop. Although nasal devoicing is not rare in Bantu languages (e.g., Bura and Bondei; Maddieson and Ladefoged 1993), the lack of acoustic cues during the nasal is. The current state of the devoiced nasal in Somali Chizigula could lead to a reanalysis of effacement as [nt → ntʰ → (n̥tʰ)→ tʰ]. In addition to providing acoustic and aerodynamic data, we propose an articulatory phonology (Browman and Goldstein 1993) account for the voiceless prenasalized stop as it compares to other nasal + stop sequences in the language.
Word length is (in part) predicted by phoneme inventory size and syllable structure
Authors: Maddieson Ian
Published: May 2016

Based on anecdotal data from a few languages, it has been proposed that typical word length in a language is correlated with the size of the phoneme inventory: more phonemes predicts shorter words. This hypothesis has now been examined in some detail using a substantial sample of languages. A standardized text widely employed for phonetic illustrations is used, namely, the fable of “The North Wind and the Sun.” Illustrations published by the IPA and others have been collected for over 100 languages. These illustrations are arguably well-matched in style and vocabulary and reflect natural inflected forms embedded in text. Average word-length in these texts is calculated and correlated with the size of the language’s consonant and vowel inventories, as well as an index reflecting the permitted complexity of syllable structure. Vowel inventory is treated in two ways; basic vowels (just the distinctions on the basic height, backness and rounding parameters), and total vowels (including other distinctions such as nasalization, length, and voice quality). Mean word length is indeed negatively correlated with size of consonant and vowel inventories, most strongly with the number of basic vowels. More complex syllable structures also correlate with a shorter mean word length.
What the ‘L?: An ultrasound study of the acoustic and articulatory characteristics of laterals in Brazilian Portuguese
Authors: Charles Sherman D.
Published: May 2016

Recent technological and computational advances have enabled researchers to investigate vocal tract articulation and acoustics with an ever-increasing degree of nuance and specificity. This is particularly enlightening in the case of articulations which are subject to tremendous cross-linguistic and interspeaker variability, like laterals (Ladefoged and Maddieson 1996). The current study uses simultaneous 3D ultrasonography and audio recordings to investigate vocal tract configurations and their acoustic consequences, and to provide a comprehensive descriptive analysis of alveolar and palatal lateral speech sounds produced by one Brazilian Portuguese speaker. The results confirm previous accounts of multiple articulatory constrictions and low F1 and F2 frequencies for alveolar laterals (Barbarena et al. 2014; Johnson 2012; Ladefoged 2003; Ladefoged and Maddieson 1996). Novel contributions include (1) identification of a dynamic tongue gesture that, based on a Perturbation Theory analysis (Stevens 1996), appears to directly reflect formant transitions from a preceding vowel to the alveolar lateral consonant, and (2) three-dimensional articulatory descriptions of both alveolar and palatal laterals in Brazilian Portuguese.
Complete tonal neutralization in Taiwan southern Min
Authors: Chen Mao-Hsu
Published: May 2016

This study reported the result of a production experiment on the tonal neutralization of context unchecked 21 tone and checked 21 tone with a glottal stop coda in Taiwan Southern Min, which are said to be realized the same as a high falling sandhi tone 51 on surface when occurring in context positions. Speakers of two age groups, younger and older, were recruited for the production experiments to examine whether the neutralization is complete. The comparison between the f0 contours was conducted with smoothing spline analysis of variance, and it showed a case of complete neutralization for both age groups, where the two f0 contours were simply overlapped. As reported by linear mixed effects modeling, the durations of the target syllables with underlying checked and unchecked 21 tones were not significantly different from each other.
A study on drinking judgment using phonation characteristics in speech signal
Authors: Lee Won-Hee , Bae Myung-Jin
Published: May 2016

In this study, speech characteristics from before and after alcohol intoxication have been comparatively analyzed through speech analysis to obtain the degree of intoxication along with its parameters. The speech characteristics of an alcohol intoxicated person, such as pitch and formant and changing of sound levels, have been studied, but the speech characteristics before and after alcohol intoxication cannot be analyzed because of the sensitiveness of environmental changes. Therefore, we need to find more precise parameters, especially Phonation threshold pressure, the high levels of which result in dehydration of the laryngeal mucous membrane after alcohol intoxication. Speech after alcohol intoxication is with low lung pressure and decreasing lung capacity. Thus, we study the speech characteristics from before and after alcohol intoxication using speech-rate in speech signal and lung capacity.