bluuebunny/crossref_metadata_2025 huggingface dataset:
Using nonlinear ultrasound to measure microstructural changes due to radiation damage in steel
Authors: Matlack Kathryn H. , Kim Jin-Yeon , Wall James J. , Qu Jianmin , Jacobs Laurence J.
Published: May 2013

This work demonstrates how nonlinear ultrasound (NLU) can be used to monitor radiation damage in nuclear reactor pressure vessel (RPV) steels. Radiation damage is a crucial concern in the nuclear industry since many nuclear plants throughout the United States are entering a period of life extension, meaning the RPV will be exposed to higher levels of neutron radiation than it was originally designed to withstand. Currently, there is no nondestructive evaluation (NDE) method to unambiguously characterize radiation damage in RPV steels, the development of which would enable the assessment of the integrity of the vessel, allowing operators to determine if they can continue to safely operate. NLU is an NDE technique that is sensitive to microstructural features in metallic materials. The physical effect monitored by NLU is the generation of higher harmonic frequencies in an initially monochromatic ultrasonic wave, arising from the interaction of the ultrasonic wave with microstructural features. Recent research has demonstrated that NLU is sensitive to the same microstructural changes that are produced in radiation damage, such as precipitate formation and changes in dislocation density. Current experimental and modeling results are presented that relate the nonlinear ultrasonic parameter to the amount of radiation damage in RPV steel materials.
Statistical moments of broadband acoustic signals propagating in a refractive, turbulent atmosphere with spatial-temporal fluctuations in temperature and wind velocity
Authors: Ostashev Vladimir , Wilson D. Keith , Vecherin Sergey N. , Collier Sandra L.
Published: May 2013

Propagation of broadband acoustic signals in a refractive, turbulent atmosphere with spatial-temporal fluctuations in temperature and wind velocity is considered. Starting from a parabolic wave equation, and using the Markov approximation and the hypothesis of locally frozen turbulence, closed-form equations for the statistical moments of arbitrary order of the sound-pressure field are derived for both sound propagation above an impedance ground and line-of-sight propagation. These equations generalize those obtained previously [Wilson and Ostashev, J. Acoust. Soc. Am. 109, 1909–1922 (2001)], where the case of monochromatic sound waves and spatial fluctuations in temperature and wind velocity was considered. The general theory can be used for analysis of many statistical characteristics of broadband acoustic signals propagating in the atmosphere, e.g., temporal coherence, frequency decorrelation, and the pulse spread and wonder. Using this theory, the spatial-temporal coherence function of a broadband acoustic signal is calculated and analyzed. Knowledge of the theoretical coherence function is important for performing source localization with physics-based estimators, particularly maximum-likelihood estimators.
Prediction of sound levels from high-altitude, broadband sources: Is there a Lloyd’s mirror effect?
Authors: Wilson D. Keith , Pettit Chris L. , Ostashev Vladimir E. , Vecherin Sergey N.
Published: May 2013

For a high-altitude, narrowband sound source, propagation models predict the presence of pronounced rings of constructive and destructive interference for listeners near the ground. This is an example of the Lloyd’s mirror effect. But even when propagation predictions are made for realistic, broadband aircraft spectra, by partitioning the spectrum into octave or one-third octave bands, running the model at the center frequency of each band, and then summing, a residual Lloyd’s mirror effect can still be apparent. By varying the approach to frequency selection in the calculation, it is shown that the rings actually have random locations and generally disappear when enough frequencies are sampled, thus implying that they are numerical artifacts. This outcome unfortunately implies that coherent calculations at many individual frequencies are required to perform a broadband calculation. Some techniques for improving convergence, such as importance sampling and performing calculations incoherently, are discussed.
Numerical simulations of evolution of weak disturbances in vibrationally excited gas
Authors: Zavershinskii Igor , Makaryan Vlavimir , Molevich Nonna
Published: May 2013

We consider a model of gas with an exponential law of relaxation of vibrational states of molecules excited by the external energy source (Joule heating in discharges, exothermic chemical reactions, optical pumping, etc). In such a medium, the second (bulk) viscosity coefficient inversion can take place due to the positive feedback between the sound perturbation and the nonequilibrium heat release. Such a medium with the negative viscosity is acoustical active. The existence of stationary nonlinear acoustical structures that are different from the step- or saw -wise shock wave structures are discussed basing on the solutions of general nonlinear acoustical equation [Molevich, Klimov, Makaryan, Int. J. Aeroacoust. No. 3–4 (2005)]. Using the numerical simulation of full one -dimensional (1D) system of relaxing gas dynamics, we show that any weak localized acoustical disturbance transforms into the sequence of self-sustained solitary pulses. Collisions of such pulses lead to their full reconstruction after the initial stage of the nonlinear increase of summarized amplitude. Using the 2D-system of relaxing gas dynamics, we consider the evolution of the noise signal into the non-stationary quasi-regular system of colliding self-sustained solitary pulses.
Numerical prediction of the vibroacoustic of sandwich panels with add-on damping
Authors: Rzig Imen , Atalla Noureddine
Published: May 2013

This paper discusses the numerical modeling of the vibroacoustic response of sandwich-composite panels with added-on damping, under mechanical and acoustical excitations. The studied damping is in the form of a viscolaetsic layer located within the panel. A modal synthesis approach is used for the calculation of the structural response and the Rayleigh’s integral is used for the acoustic response (the panel is assumed flat and baffled). Since the panel has a viscoelastic core, a methodology is presented to handle efficiently the modeling of the frequency depended properties of the viscoelastic layer. A direct frequency response is used to validate the proposed approach. Next, a parameters study on the effect of the viscoelastic layer location is presented, In particular, three locations are compared: within the Honeycomb core, within the skins and added to the skin with a constraining layer. The effects of the excitation type on the vibration and acoustic response are also discussed.
A dynamic response of a laminated windshield with viscoelastic core—Numerical vs experimental results
Authors: Bouayed Kaiss , Hamdi Mohamed-Ali
Published: May 2013

The dynamic response of a laminated windshield with a viscoelastic core is computed using a simplified modal method combined with a quadratic sandwich finite element. The method is based on a modal expansion of the displacement field using a constant young modulus of the core layer. The frequency dependence of the complex modulus of the core is taken into account using the residual dynamic stiffness matrix. The method is applied to predict the frequency response of two types of laminated windshield using a standard and acoustic PVB cores. Numerical results are compared in a first step with those obtained using the direct solver of Nastran software, and in a second step with experimental results obtained by a laser vibrometer. Comparisons show a very good agreement between experimental and numerical results and demonstrate the efficiency of the simplified modal solving method and the developed parabolic sandwich element. The method will be applied to compute the coupled vibro-acoustic frequency response of a full vehicle body integrating a laminated windshield and glass surfaces.
Computer-based English /r/-/l/ perceptual training for Japanese children
Authors: Shinohara Yasuaki , Iverson Paul
Published: May 2013

Computer-based perceptual training has proven successful for improving English /r/-/l/ perception by Japanese adults, but this has not been tested with younger age groups, who presumably have greater perceptual plasticity. The present study examined phonetic training for children 6–8 years old. The training program included identification and discrimination tasks with word-initial English /r/-/l/ minimal pairs (e.g., rock–lock), with each participant completing ten sessions. The results demonstrated that children improved their English /r/-/l/ identification, although identification in untrained positions such as medial and consonant clusters did not improve as much as in the trained word-initial position. In addition, older children in this age range improved more than did younger children, suggesting that the ability to use this kind of program may improve with age, even though perceptual plasticity for speech presumably declines with age.
Acoustic characteristics of Danish infant directed speech
Authors: Bohn Ocke-Schwen
Published: May 2013

Danish presents several challenges for language learners, such as a very densely packed upper portion of the acoustic vowel space, and a sibilant contrast that is acoustically less distinct than in, e.g., English. The present study examined whether Danish caregivers enhance Danish contrasts when speaking to their 18 month old children (infant directed speech—IDS) as opposed to an adult (adult directed speech—ADS). Caregivers were recorded talking about toy animals in conversations with their child and with an adult interlocutor. The toy names were designed to elicit Danish contrasts differing in voice onset time and in place of articulation for sibilants, and vowels which are close neighbors in the crowded Danish vowel space. The dependent variables for the comparison of IDS to ADS were as follows: VOT differences for homorganic stop consonants, the frequency at the amplitude peak for the sibilants, the Euclidean F1/F2 differences between vowels, F0 of the stressed (first) syllable in the toy name, as well as the duration of the stressed syllable, the vowels, and the fricatives. Results of the acoustic differences between ADS and IDS were compared to the results of parents’ reports on the children’s productive and receptive vocabulary knowledge. [Work supported by the Danish National Research Foundation –Danmarks Grundforskningsfond.]
Perception of speaker age in children’s voices
Authors: Assmann Peter F. , Barreda Santiago , Nearey Terrance M.
Published: May 2013

To study the perception of speaker age in children’s voices, adult listeners were presented with vowels in /hVd/ syllables either in isolation or in a carrier sentence, and used a graphical slider to register their estimate of the speaker’s age. The data showed a moderate correlation of perceived age and chronological age. Age estimation accuracy was fairly constant across age up to about age 11, but there was a systematic tendency for listeners to underestimate the ages of older girls. This tendency was actually enhanced when listeners were informed of the speaker’s sex. Age estimation accuracy was higher for syllables embedded in a carrier sentence. Linear regression analyses were conducted using acoustic measurements of the stimuli to predict perceived age. These analyses indicated significant contributions of fundamental frequency, duration, vowel category, formant frequencies, as well as certain measures related to the voicing source. The persistent underestimation of age for older girls, and the effect knowledge of speaker sex has on this underestimation suggest that acoustic information is combined with expectations regarding speakers of a given sex in arriving at an estimate of speaker age.
Vowel production in sighted adults and blind adults: A study of speech adaptation strategies in high-intensity background noise
Authors: Trudeau-Fisette Pamela , Turgeon Christine , Côté Dominique
Published: May 2013

Recent studies have shown that congenitally blind speakers have greater auditory discrimination acuity than sighted speakers [Ménard, Dupont, Baum, and Aubin, J. Acoust. Soc. Am. 126, 1404–1414 (2009)]. At the production level, however, blind speakers produce smaller displacements of the lips (visible articulator) than their sighted peers. In order to further investigate the impact of visual experience on the articulatory gestures used to produce intelligible speech, adaptation strategies in background noise was studied in blind and sighted speakers. Ten sighted and 10 congenitally blind adult French participants were recorded during the production of the vowels /i/, /y/, /u/, /a/ in a CVC context. Two conditions were elicited: with high-intensity noise heard through headphones and without noise. Synchronous acoustic and articulatory data were recorded using the Carstens AG500 Electromagnetic Articulograph system. Formant measures and movements of the lips and tongue were analyzed. Results reveal that blind speakers produced smaller ranges of lip movement than sighted speakers in the noisy condition, suggesting that the blind subjects made less use of visible articulators to improve intelligibility. Results are discussed in light of multimodal production-perception relationships in speech.
Token-to-token variability and anticipatory coarticulation as indicators of maturity of speech motor control in 4-year-old children
Authors: Barbier Guillaume , Perrier Pascal , Ménard Lucie , Tiede Mark , Perkell Joseph S.
Published: May 2013

Children’s gestures do not appear to be executed with the same dexterity as adults’. Studies of arm movements have shown that young children’s gestures are less accurate, more variable and slower than those of adults. This difference in behavior can be explained by a lack of experience with the sensory consequences of motor acts and still-developing forward models for the control of those acts. The hypothesis of immature and incomplete sensori-motor representations for speech in 4-year-old native speakers of Canadian French is addressed here through the analysis of ultrasound recordings of tongue contour kinematics and the speech signal from a corpus of isolated vowels and vowel-consonant-vowel sequences. Special attention is devoted to the analysis of vowel variability from two perspectives. Variability across repetitions in a single context provides information about the accuracy of the control. Contextual variability provides insights into the planning process as reflected in anticipatory coarticulation. Analysis of the observed lingual gestures will lead to improved understanding of the development of speech motor control and refinement of sensori-motor representations of speech. [Work supported by FQRNT project N° 147877 and ANR project ANR-08-BLAN-0272.]
Toddlers' comprehension of noise-vocoded speech and sine-wave analogs to speech
Authors: Newman Rochelle S. , Chatterjee Monita , Morini Giovanna , Nasuta Molly
Published: May 2013

A great deal of research has investigated listeners’ ability to compensate for degraded speech signals such as noise-vocoded speech (a signal with reduced spectral structure but intact amplitude envelope information) and sine-wave analogs to speech (a signal that maintains the global dynamic spectral structure of the signal at the expense of amplitude envelope information). Nittrouer and colleagues found developmental changes in the ability to comprehend such signals, reporting that while adults perform more accurately with sine-wave analogs than with noise-vocoded speech, school-aged children show the opposite pattern [e.g., Nittrouer Lowenstein and Packer (2009)]. In a series of studies, we tested toddler’s comprehension of these degraded signals. Twenty-seven-month-old children saw two images on each trial (e.g., cat, dog), and heard a voice instructing them which image to look at (“Find the cat!”). Sentences were presented either in full speech or were degraded. Toddlers (n = 24 per condition) looked at the appropriate object equally long with vocoded speech of 24 channels (60.2%) or 8 channels (62.4%) as with full speech (62.6%), but performed barely above chance with 4 channels (53.6%) and at chance for 2 channels (49.8%). Preliminary results suggest that performance with sine-wave analogs is poorer than 8-channel vocoded speech (56.1%), but testing is ongoing.
Segment imitation in unfamiliar language varieties
Authors: Forsberg Julia
Published: May 2013

Phonetic imitation has been explored in various research: in studies on impersonation [Zetterholm (2001); spontaneous phonetic imitation [Babel (2012)]; and as part of studies investigating overall imitation [Papcun (1988), Babel (2011)]. There are fewer studies focusing on specific phonemes, and the context required for successful imitation. Babel [186 (2012)] (Am. English) and Zetterholm [281 (1997)] (Swedish) found spontaneous imitation was more common in the open vowels than in close. A case study by Zetterholm [275 (1997)] of one impersonator shows formant measurements as closer to the target voice than that of the impersonator. This paper presents a comparative acoustic analysis of imitation of unfamiliar phonemes by untrained imitators, based on a small study containing recordings from SWEDIA and SUF. Vowel tokens are edited to three lengths with varying degrees of context. Listeners are asked to imitate the sound, and attempt to use it in a word. Acoustic analysis of F0, 1, 2, and 3 will be compared to the original recordings as well as the speakers’ own speech. The conclusion and discussion includes indications of how easy it is to produce phonemes when they are not native to a speaker’s own variant, as well as its relation to the forensic context.
Imitability of contextual vowel nasalization and interactions with lexical neighborhood density
Authors: Zellou Georgia , Scarborough Rebecca , Nielsen Kuniko
Published: May 2013

This study investigates the imitability of contextual vowel nasalization in English. Unlike other phonetic features reported to be imitable [e.g., vowel formants (Babel, 2012), VOT (Nielsen, 2011)], vowel nasality is non-contrastive in English. Nasality is, however, systematically variable: words from dense lexical neighborhoods (high-ND words) are produced with greater nasality than words from sparse neighborhoods [Scarborough (2004), (2012)]. Two experiments were conducted to test (1) whether (experimentally manipulated) nasality can be imitated in a shadowing task, and (2) whether direction of manipulation (more or less nasality, enhancing or countering natural neighborhood-conditioned patterns) affects shadowing behavior. Subjects shadowed 16 high-ND words (which are naturally more nasal) containing a vowel-nasal sequence and modified by spectral mixing to exhibit either greater-than-natural (experiment 1) or less-than-natural (experiment 2) nasality. Both the increase and the decrease in nasality were imitated (though not overall degree of nasality, as our imitation model was more nasal in both conditions than any of our subjects). This change persisted into a post-shadowing task for just the less-nasal condition. These results indicate that speakers are sensitive to non-contrastive phonetic detail in nasality, affecting their subsequent production. Further, the naturalness of nasality (reflecting neighborhood-conditioned variation) may affect the pattern of imitation.
The vowel spaces of Southern Californian English and Mexican Spanish as produced by monolinguals and bilinguals
Authors: Grijalva Criccely , Piccinini Page E. , Arvaniti Amalia
Published: May 2013

The vowel spaces of Southern Californian English and Mexican Spanish were investigated using three groups of speakers: 11 English monolinguals (8 females), 11 Spanish monolinguals (9 females), and 10 Spanish-English bilinguals (7 females). Speakers produced six repetitions of the ten American English vowels [ i, ɪ, ɛ, æ, ɑ, ɔ, ʊ, u, ʌ, and ɜr] and six repetitions of the five Spanish vowels [i, e, a, u, and o]. Monolinguals produced vowels in one language; bilinguals produced vowels in both languages. Preliminary analysis shows Southern Californian English back vowels were less fronted compared to the results of Hagiwara (1997) from Southern Californian English, but more fronted than those of Hillenbrand et al. (1995) on General American English. Mexican Spanish back vowels [u] and [o] were substantially fronted compared to Castilian Spanish vowels [Bradlow (1995)], while [i] was lower and less fronted. In general, Mexican Spanish vowels were produced higher and more backed than Southern Californian English vowels in monolingual productions. Bilinguals produced their two vowel spaces closer together but with less dispersion than monolinguals, showing how bilinguals keep both language categories distinct.
Longitudinal changes of formant values and vowel space in two Mandarin-speaking children before 9 years of age
Authors: Chen Li-mei , Cheng Fan-Yin , Hsu Wei Chen
Published: May 2013

Vowel productions of two Mandarin-speaking children were audio recorded in their homes with picture naming tasks once every 3 months, from birth to 9 years old. The present study is the ninth year of a longitudinal observation. Major findings in this stage are as follows: (1) The trend of decrease in formant values was continuously observed in the boy subject. As for the girl subject, it is not until 9 years old, the obvious decrease in formant values was found, especially in F1; (2) F1 values are more stable than F2 values in both subjects. They appeared to acquire jaw movement sooner than tongue movement; (3) Throughout these 9 years, the variability of F1 is around 200–300 Hz, and the variability of F2 is 500–700 Hz in both subjects. No trend of decrease was found; (4) The trend of shrinkage in F1-F2 vowel area continues from 7 to 9 years old for the boy subject, but not for the girl subject; (5) There is a clear decline in fundamental frequencies at 8-9 years of age in the boy subject. Longitudinal data of vowel formant values from the same group of subjects provide important references for assessment and treatment of articulation disorders in children.
Exploring production-perception relationships in normal hearing and cochlear implant adults: A lip-tube perturbation study
Authors: Turgeon Christine , Prémont Amélie
Published: May 2013

It has been claimed that auditory feedback mechanisms enable monitoring and calibration of feedforward commands in speech production. Therefore, lack of auditory feedback may interfere with adequate compensation strategies in perturbed situations. This study investigates the effect of hearing status and a lip tube perturbation on vowel production. Eleven normal-hearing controls, and seventeen cochlear implant (CI) users (7 prelingually, 10 postlingually) were recorded during the production of the vowel /u/. Acoustic analyses were conducted with and without a 15-mm-diam tube inserted between the lips. Recording sessions were also made before and after the perturbation, with and without auditory feedback. Deaf participants’ auditory feedback was provided by the CI and interrupted by switching off their implant devices. Separate analyses were conducted on the first (F1), the second formant (F2), and the fundamental frequency (F0). Results revealed a main effect of group and an interaction between condition and hearing status. Together, results suggest that auditory feedback plays an important role in speech compensation.
Spherical array processing with binaural sound reproduction for improved speech intelligibility
Authors: Shabtai Noam R. , Rafaely Boaz
Published: May 2013

In telecommunication applications, interfering sounds and reverberation can have a detrimental effect on speech intelligibility. For this reason, microphone arrays have been recently employed in telecommunication systems for natural environments. Currently applied array processing methods typically aim to produce array output, which is optimal on signal-based measures, e.g., signal-to-noise ratio (SNR). These measures may be particularly appropriate when the receiver is a machine. However, in order to enhance speech intelligibility when the receiver is another human, it may be desired to trigger spatial hearing capabilities of the human auditory system, such as the cocktail party effect. In particular, spatial-release from masking has been investigated. This work presents a spherical array signal processing framework in which array output is generated binaurally using the head-related transfer function. In this framework both target direction is enhanced and spatial information of all sources are perceived by the listener. The performance of the proposed binaural beamformer is compared to the performance of a non-binaural maximum directivity beamformer based on a spatial reproduction listening tests. The average percentage correct decision is calculated over five subjects and is shown to be higher when the binaural beamformer is used for every tested SNR.
Sound field planarity characterized by superdirective beamforming
Authors: Jackson Philip J. , Jacobsen Finn , Coleman Philip D. , Abildgaard Pedersen Jan
Published: May 2013

The ability to replicate a plane wave represents an essential element of spatial sound field reproduction. In sound field synthesis, the desired field is often formulated as a plane wave and the error minimized; for other sound field control methods, the energy density or energy ratio is maximized. In all cases and further to the reproduction error, it is informative to characterize how planar the resultant sound field is. This paper presents a method for quantifying a region's acoustic planarity by superdirective beamforming with an array of microphones, which analyzes the azimuthal distribution of impinging waves and hence derives the planarity. Estimates are obtained for a variety of simulated sound field types, tested with respect to array orientation, wavenumber, and number of microphones. A range of microphone configurations is examined. Results are compared with delay-and-sum beamforming, which is equivalent to spatial Fourier decomposition. The superdirective beamformer provides better characterization of sound fields and is effective with a moderate number of omni-directional microphones over a broad frequency range. Practical investigation of planarity estimation in real sound fields is needed to demonstrate its validity as a physical sound field evaluation measure.
Comparison of statistics of controlled source tones and single ship noise in the deep ocean
Authors: Baxa Brianne M. , D'Spain Gerald L. , Worcester Peter F. , Dzieciuch Matthew A. , Heaney Kevin D. , Mercer Jim A. , Baggeroer Art B.
Published: May 2013

The deep ocean experiment, PhilSea09, was conducted April-May, 2009, in the central part of the northern Philippine Sea (22d N, 126d E). During one period in the experiment, the R/V Melville was station-keeping 35 km from the Distributed Vertical Line Array (DVLA) while seven tones, from 79 Hz to 535 Hz, were transmitted from a controlled source suspended below the ship. Recordings on the 1000-m section of the DVLA centered on the surface conjugate depth at 5026 m were dominated by the noise of this ship except at the controlled source tone frequencies. Using non-parametric statistical tests, the statistics of the spectral envelope at the tone frequencies are compared to the statistics of those for the nearby ship-noise-dominated frequency bins. When a high tone-signal to ship-noise ratio exists, the statistics of the tones differ from those of the ship noise at the 5% level of significance by the Kolmogorov-Smirnov two-sample test. Tone statistics are seen to be Gaussian distributed at frequency bands of low tone-signal to ship-noise ratios, whereas at high signal to noise ratios, the controlled source tones are non-Gaussian. When the signal to noise ratio is high, the statistics of the tone and of the noise are from different distributions. Both the tones and shipping noise travel approximately the same path to the DVLA, so these differences in the received field statistics represent differences in the statistical properties of these two acoustic sources themselves, not of the environment.
Probability distribution for energy of saturated broadband ocean acoustic transmission: Results from Gulf of Maine 2006 experiment
Authors: Tran Duong D. , Ratilal Purnima
Published: May 2013

The probability distribution of ocean-acoustic broadband signal energy after saturated multipath propagation is derived using coherence theory. The frequency components obtained from Fourier decomposition of a broadband signal are each assumed to be fully saturated with energy spectral density that obey the exponential distribution with 5.6 dB standard deviation and unity scintillation index. When the signal bandwidth and measurement time are larger than the correlation bandwidth and correlation time respectively of its energy spectral density components, the broadband signal energy obtained by integrating the energy spectral density across the signal bandwidth then follows the Gamma distribution with standard deviation smaller than 5.6 dB and scintillation index less than unity. The theory is verified with broadband transmissions in the Gulf of Maine shallow water waveguide in the 300–1200 Hz frequency range. The standard deviations of received broadband signal energies range from 2.7 to 4.6 dB for effective bandwidths up to 42 Hz, while the standard deviations of individual energy spectral density components are roughly 5.6 dB. The energy spectral density correlation bandwidths of the received broadband signals are found to be larger for signals with higher center frequencies and are roughly 10% of each center frequency.
Analysis of horizontal wave number statistical characteristics to the problem of sound propagation in two-dimensional-fluctuating shallow sea with losses
Authors: Gulin Oleg , Yaroshchuk Igor
Published: May 2013

On the basis of the approach previously proposed by authors, a problem of the middle-frequency sound propagation in 2D-fluctuating shallow sea with losses is considered. Statistical characteristics of horizontal wave numbers corresponding to modal ones are studied. Fluctuations of horizontal wave numbers determine statistical features of wave field in random sea medium if wave field is sought by modal expansion. Within the framework of adiabatic approximation, we present calculations of sound field statistical moments, which demonstrate an effect of transmission loss attenuation along the horizontal distance. There are no references in acoustic literature for this fact. Some estimation has been carried out to explain new effect associated with two reasons that are the medium losses and sound speed fluctuations. They together influence wave numbers of modes in such a way attenuating losses.
Amplified music and measurement results regarding inflatable membrane absorber technology
Authors: Adelman-Larsen Niels W.
Published: May 2013

Previous studies and experience have shown that what distinguishes the best from the less well liked venues for pop and rock music is a shorter reverberation time in the 63–250 Hz octave bands. Since a longer reverberation time in these bands is needed in order to obtain warmth at classical music concerts, variable acoustics must address these frequencies in order to provide the best results in multipurpose halls. This paper will expand on research on recommendable acoustics for amplified music. Certified measurements from reverberation chambers and installed systems on a patented, inflatable, on/off absorption technology are presented. Since the technology can be used in the entire ceiling area, the T30 of a hall can be lowered by almost 50% in the important octave bands for pop and rock music. Absorption coefficients are almost constant across the frequency bands from 63 to 1000 Hz. The technology, which is the only passive solution to enable variability also at the important lower frequencies, is meant to be used in any hall where both classical as well as amplified music is being played, such as in music schools and performing arts centers.
The role of high-frequency cues for spatial hearing in rooms
Authors: Bharadwaj Hari , Masud Salwa , Shinn-Cunningham Barbara
Published: May 2013

The ability to attend to a sound source of interest while ignoring competing sounds is vital to navigating everyday acoustic scenes. Commonly, this ability depends on the ability to focus on a sound source using acoustic spatial cues, particularly interaural time differences (ITDs) and interaural level differences (ILDs). Based on past studies of localization in anechoic settings, low-frequency ITDs have been thought to dominate perception of source location. However, reverberant environments differentially degrade ITDs and ILDs, which may affect their relative influence on localization. Moreover, a recent study suggests that ILDs play a bigger role on spatial perception in reverberant settings than in anechoic settings. Here, in a series of localization and spatial attention experiments using high-pass, low-pass and broadband sounds, we tested the hypothesis that high-frequency ILD and envelope ITD cues are important for spatial judgments in reverberant rooms. We also measured the brainstem frequency following responses (FFRs) of individual subjects in response to click trains and spoken syllables. Results suggest that compared to in anechoic space, in reverberant settings, high-frequency cues are more reliable and influential on perception and that the strength of FFR phase locking to stimulus envelope predicts how well individual listeners can direct spatial attention.
Internet rooms from internet audio
Authors: Chafe Chris , Granzow John
Published: May 2013

Music rehearsal and concert performance at a distance over long-haul optical fiber is a reality because of expanding network capacity to support low-latency, uncompressed audio streaming. Multichannel sound exchanged across the globe in real time creates “rooms” for synchronous performance. Nearby connections work well and musicians feel like they are playing together in the same room. Larger, continental-size, distances remain a challenge because of transmission delay and seemingly subtle but perceptually important cues which are in conflict with qualities expected of natural rooms. Establishing plausible, room-like reverberation between the endpoints helps mitigate these difficulties and expand the distance across which remotely located musicians perform together comfortably. The paper presents a working implementation for distributed reverberation and qualitative evaluations of reverberated versus non-reverberated conditions over the same long-haul connection.
Maa Dah-You and the design of reverberant rooms for determination of sound power
Authors: Maling George C.
Published: May 2013

Reverberation rooms are useful tools for the determination of sound power emitted by machines and other sound sources. One critical factor in the design of such rooms is the mode spacing at low frequencies. A seminal contribution to our understanding of mode spacing was made by Maa in 1939, and his paper was published in the Journal of the Acoustical Society of America. Richard Bolt also made a contribution at about the same time. In this paper, we begin with Maa’s work and trace the development of mode spacing statistics through the work of Richard H. Bolt and Ludwig Sepmeyer. A computer study by the author of mode spacing statistics in 200 cubic meter rooms is described. This work led to recommendations for room dimensions to be included in an international standard on determination of sound power in reverberation rooms.
Active noise control in rooms
Authors: Tichy Jiri
Published: May 2013

Maa published his ideas and achievements in the early days of the new branch of noise control based on using secondary sources to reduce the sound level of the primary sources by negative interference. He concentrated on active control of the reverberant sound in a room by a loudspeaker placed in one of the room corners and a microphone in its close vicinity. He examined the role of the complete solution of the wave equation consisting of a direct wave radiated from the sources and the reverberant field created by room modes. His work resulted in deriving a general formula for possible noise reduction that is independent of the shape, size, and the content of the room. He has shown noise reduction of 8 dB for a noise band centered at 100 Hz.
Bugs and bats: Neural analysis of behaviorally relevant sounds in crickets
Authors: Pollack Gerald
Published: May 2013

Hearing in crickets is specialized to serve particular behavioral functions, namely intraspecific communication and predator avoidance. Male crickets produce species-specific acoustic signals (songs) that attract distant females, promote copulation, and contribute to agonistic interactions with rivals. Crickets also hear the echolocation calls of aerially hunting bats, which evoke avoidance responses. These clear behavioral functions of hearing, combined with the relative simplicity of the cricket's nervous system, make it possible to address questions about how behaviorally relevant sensory signals are analyzed at the level of single, uniquely identifiable nerve cells. Cricket songs and bat calls differ both in rhythm and in spectrum, and neurons throughout the auditory processing chain are specialized for processing these two sorts of signal. I will focus on specializations that are evident at early stages of auditory processing, i.e., primary sensory neurons and the first-order interneurons with which they interact.
New observations and modeling of an unusual spatiotemporal pattern of fish chorusing off the southern California coast
Authors: D'Spain Gerald L. , Batchelor Heidi H. , Helble Tyler A.
Published: May 2013

The purpose of this paper is to present new results on an unusual spatiotemporal pattern of fish chorusing off the southern California coast. Characteristics of this fish chorus have been reported previously; it occurs at night in the late spring and summer months in shallow, sandy bottom regions just outside the surf zone. The background sound levels increase by up to 30 dB and cycle in level with a period of 30–35 s all night long. In this paper, recent results from measurements made by a set of high spatial resolution sensor systems spanning a 50-km stretch of coastline out to 20 km offshore over a 2-month time period are presented. These data allow the spatial dependence and long-term temporal variability of the chorus to be examined at high spatial resolution. Refinements to a numerical model that predicts this chorusing behavior are required to account for some aspects of these new observations. [Work supported by the Office of Naval Research, Code 322-MMB.]
Electromagnetic-acoustic imaging of stiffness and dielectric properties in gels
Authors: Zhang Ning , Cleveland Robin , Edwards David J.
Published: May 2013

Here we present a novel multi-modal imaging method, electromagnetic-acoustics (EMA) that combines acoustic radiation force and electromagnetic scattering. Experiments were carried out in gels using a focused megahertz ultrasound source, amplitude modulated at 160 Hz, to create an oscillating radiation force resulting in vibration of the gel in the focal region. At the same time an EM signal was transmitted into the gel and the backscattered EM signal recorded. The tissue that was in motion resulted in a Doppler shift of the EM signal which manifested itself as frequency modulation of the EM wave. The modulated component was detected by means of a demodulator and lock-in amplifier and the amplitude of the modulated signal we call the EMA signal. By steering the ultrasound beam through the sample, an EMA image can be created which has spatial resolution of the ultrasound but is sensitive to shear and dielectric properties. We show that the EMA signal is sensitive to changes in elasticity and conductivity in a gel. EMA may have utility in biomedical imaging by detecting diseases, which have contrast in dielectric properties without the cost and complexity of an magnetic resonance imaging system.
New planar nano-gauge detection microphone: Analytical and numerical acoustic modeling
Authors: Guianvarc'h Cécile , Verdot Thierry , Czarny Jaroslaw , Redon Emmanuel , Ege Kerem , Guyader Jean-Louis , Walther Arnaud , Robert Philippe
Published: May 2013

The miniaturization of microphones is of great interest for several fields, such as medical applications (audio implants), or consumer electronics (cell phones). Almost all existing miniature microphones rely on electrostatic transduction and offer good performances (sensitivity, frequency bandwidth). However, their sensitivity, proportional to the membranes area, would be dramatically reduced in case of extreme miniaturization. A new concept of microphones developed by CEA-LETI, which uses membranes moving in the plane of the substrate and inducing strain on piezoresistive Si nano-gauges (M&NEMS technology), seems promising for its miniaturization potential without significant decrease of sensitivity. The design and optimization of such planar piezo-resistive microphone require a deep understanding of its acoustic and vibroacoustic behavior. Regarding the small dimensions of the slits (1–100µm) and the sharp discontinuities in the microphones structure, viscous and thermal effects in the boundary layers and turbulent perturbations are of great importance, and must then be taken into account with high accuracy in device modeling. The aim of the present work is to provide accurate analytical and numerical (FEM) models able to gather all these effects in a consistent manner, and to suggest an experimental method to check their validity.
Miniaturized electrostatic receiver with small-sized backing electrode
Authors: Podkovskiy Alexey , Honzík Petr , Durand Stéphane , Joly Nicolas , Bruneau Michel
Published: May 2013

A miniaturized electrostatic receiver design, having a central cylindrical backing electrode of small radius surrounded by a flat annular cavity behind the circular membrane, can lead to both a higher sensitivity and a larger frequency bandwidth compared to the ones achieved with other designs, while bringing a geometrical simplicity which is advantageous from the point of view of microfabrication. An appropriate computational method, relying on a specific 2-D axisymmetrical simulation using an adaptive mesh and accounting for both viscous and thermal boundary layer effects, provides results against which analytical results can be tested. An analytical approach, which leads to solutions based on the eigenmode expansion of the membrane displacement, the acoustic pressure field depending on the radial coordinate in the central fluid gap but being assumed quasi-uniform in the annular cavity, is much faster in terms of running time and appears to be sufficiently accurate to achieve final optimization of this kind of devices.
How a hearing aid transducer works
Authors: Kim Noori , Allen Jont B.
Published: May 2013

The oldest magnetic earphone, the balanced armature receiver (BAR), is the most widely used receivers in modern hearing-aid instruments, where the efficiency of the power (battery life) and the size of the device, as well as the larger frequency bandwidth, are critical parameters. Since these miniature loudspeakers remain one of the most expensive components of the hearing-aids, a detailed studying of them is a cornerstone of understanding the hearing-aid system, and we believe that the appropriate and rigorous analysis of this transducer is critical. The motivation of this study started from the modeling of a widely used commercial hearing-aid receiver ED series, manufactured by Knowles Electronics, Inc. Our proposed model includes a semi-inductor and a gyrator along with the two-port network glue which enables us an intuitive design of the electromagnetic transducer. Based on the BAR model, we will investigate and discuss the roles of each physical component in the BAR such as a coil, magnets, an armature, a diaphragm, and the rear volume of the receiver. Ultimately, this work will deliver a fundamental and innate answer for the question, “How does a hearing-aid transducer work?”
Development and performance evaluation of virtual auditory display system to synthesize sound from multiple sound sources using graphics processing unit
Authors: Watanabe Kanji , Oikawa Yusuke , Sato Sojun , Takane Shouichi , Abe Koji
Published: May 2013

Head-related transfer function (HRTF) is characterized as sound transmission from sound source to listener's eardrum. When a listener hears a sound that is filtered with the HRTFs, the listener can localize a virtual target (sound image) as if the sound had come from the position corresponding to that at which the HRTFs were measured. A moving sound image can be generated to switch HRTFs of successive direction in real-time. While many virtual auditory displays (VADs) based on synthesis of HRTFs have been proposed, most of them can synthesize only a few sound images due to lack of computation power. In this article, the VAD system implemented based on graphics processing unit (GPU) was introduced. In our system, the convolution of HRTFs is parallelized on GPU to realize a high-speed processing. In addition, the multiple HRTFs each of which is corresponding to sound sources at different position are processed in parallel to control multiple sound image simultaneously. In this article, the performance of our system was evaluated not only objectively but also subjectively. The results showed that our current system can present at least 40 sound images simultaneously in real-time.
Measurement of acoustic transmission properties of a handset with a piezoelectric vibrator using a head and torso simulator
Authors: Horiuchi Toshiharu , Kato Tsuneo
Published: May 2013

This paper presents equalizing an acoustic transmission property from a handset with a piezoelectric vibrator at a pinna to an eardrum with that of a normal headphone by hearing the measured sound through a head and torso simulator (HATS). Recently, a piezoelectric vibrator that vibrates a pinna to produce sounds was adopted as a receiver of smartphones to improve perceived quality within noisy environments. The HATS, used for handset testings in accordance with ITU-T recommendations, has a silicon-rubber pinna simulator to reproduce realistic acoustic properties with its human-like shape and stiffness. However, the handset with the built-in piezoelectric vibrator is beyond the scope and was not tested on the HATS before. This paper clarifies the difference of frequency responses between the pinna simulator and the real pinna based on a subjective assessment that adjusts the loudness of pure tones through the pinna simulator to be equalized auditorily to those through the real pinna. We used B&K HATS Type 4128-D. The results indicated a flat response for the pinna simulator, while a low-pass-like response for the real pinna with a cutoff at 1.5 kHz. Thereby, the actual sound can be simulated from the sound measured by the HATS with the responses.