bluuebunny/crossref_metadata_2025 huggingface dataset:
Excitation of Rayleigh and zero-group-velocity Lamb waves using air-borne N-waves focused by an ellipsoidal reflector
Authors: Dai Xiaowei , Haberman Michael R. , Tsai Yi-Te , Zhu Jinying
Published: May 2013

Air-coupled ultrasonic non-destructive testing (NDT) of elastic solids is a challenge due to the large impedance contrast between air and most materials used in industrial and structural applications. However, because air-coupled sensing offers many advantages such as rapid scanning of large areas and the elimination of part immersion for inspection, there remains strong incentive to find unique methods for air-coupled excitation of wave motion in elastic solids. This work presents experimental results of an in-air acoustic source that has been shown to excite wave motion in high impedance elastic solids. The source consists of a spark generator and an ellipsoidal reflector. The spark generator radiates a short-duration, high-amplitude acoustic signal as the result of an electrostatic discharge between two electrodes with high potential difference. Analogous to lithotripter, the spark is located at the near focus and generates an outgoing wave that is then focused at the far focus of the reflector which is co-located at the air-solid interface. Measurements of the air-borne acoustic wave in the free-field, the focused acoustic wave, and Rayleigh and zero-group-velocity (ZGV) Lamb waves generated in a concrete slab will be presented and analyzed. [Work supported by NIST Technology Innovation Program (TIP).]
Coupling of finite difference elastodynamic and semi-analytic Rayleigh integral codes for the modeling of ultrasound propagation at the hip
Authors: Cassereau Didier , Nauleau Pierre , Grimal Quentin , Minonzio Jean-Gabriel , Bendjoudi Aniss , Bossy Emmanuel , Laugier Pascal
Published: May 2013

Ultrasonic exploration of the femoral neck is of wide interest as it can provide some information about a potential fracture risk, particularly for osteoporosic patients. In vivo, the ultrasonic wave first propagates through soft tissues that can be idealized as a homogeneous fluid. Then, the ultrasonic wave interacts with the bone structure. Transmitted and back-propagated signals are then measured at receivers. A numerical model of this complete chain is useful to understand and control the various parameters involved in this process. The complexity of the bone structure is approached using the elastodynamic finite difference time domain (FDTD) code SimSonic. Due to the small size of the spatial grid needed by FDTD schemes, the propagation between the emitter and the femoral neck may be excessively time and resource consuming. We have developed a coupling between SimSonic and a direct and fast evaluation of the diffraction in homogeneous fluids, based on the numerical discretization of the Rayleigh integral. This approach allows to reduce drastically the total computation time for the complete simulation. Results obtained with this new system are presented, including computation times and computer resources. This approach is particularly useful to simulate experiments with phased arrays, which involve several emissions.
An infrasound-based avian navigational “map”
Authors: Hagstrum Jonathan T.
Published: May 2013

The “compasses” (solar, star, geomagnetic) that homing pigeons and other migratory birds use to orient during flight are generally understood, but the “map” sense they need to first determine their homeward direction is not. Atmospheric odor and geomagnetic gradients have been proposed as “map” cues, but are inadequate and remain controversial. Experiments with frosted lenses indicate that sight can also be ruled out. Laboratory tests, however, show that pigeons can detect infrasound (>0.05 Hz), and such signals travel with little attenuation for 1000s of kilometers through the atmosphere. Results from an acoustic ray-tracing program (HARPA) using daily atmospheric profiles are compared with pigeon release data for a number of sites in upstate NY. HARPA runs show that homeward infrasonic cues could have arrived at the sites from directions opposite pigeon departure bearings, especially when these bearings were unusual. Such signals possibly arise from ground-to-air coupling of microseisms or from scattering of microbaroms off terrain features (~0.2 Hz). Pigeons and other birds might use Doppler shifts to determine the directionality of homeward infrasonic cues while flying in circular or other patterns at constant velocity after release; they apparently have built-in airspeed indicators adapted from their olfactory and aural systems.
Modeling pulse-like lip vibrations in brass instruments
Authors: Kemp Jonathan A. , Smith Richard A.
Published: May 2013

During the starting transient of a note on a brass instrument it can take several cycles of lip vibration before acoustics reflections from the end of the instrument can influence the lip frequency. Under certain conditions, the lip may fail to oscillate at the pitch of the air column resulting in an unwanted pulse-like waveform with relatively low repetition rates (similar to the vocal fry register of phonation in the human voice). This is often observed in the playing of beginners if the lips are insufficiently tense or if the top and bottom lips overlap to a large extent. In this study, the reasons for this behavior will be investigated using modeling techniques with the aim of improving the agreement between physical models and measured transients by including the forces responsible for this effect.
Measurement setup for articulatory transient differences in woodwind performance
Authors: Hofmann Alex , Chatziioannou Vasileios , Weilguni Michael , Goebl Werner , Kausel Wilfried
Published: May 2013

To model transient differences caused by varying articulation techniques on single-reed woodwind instruments, human performances have to be measured and analyzed. In a previous study, we investigated differences between tongued and air-separated tones on a saxophone by monitoring inner mouthpiece pressure, mouth pressure, and reed bending during performance. Some of the observed effects (e.g., damping and displacement of the reed) were applied to a physical model. Although tip-opening is an essential parameter of lumped reed models, we were only able to directly compare our measurements to the model by inner mouthpiece air-pressure. In this study, we aim to relate measurable reed-bending to the resulting tip-opening and also determine the sensor-reed's stiffness. A micro-mechanical characterization test system records the static and dynamic mechanical data of the sensor-equipped reeds. Pull and tensile measurements are used to reveal the relation between reed bending and tip opening and will increase our understanding of quasi-static properties of the reed-mouthpiece system. This allows conclusions about the tip opening behavior at transient emergence in expressive single-reed woodwind performance.
Acoustical characteristics of Technology Educational Shops in British Columbia
Authors: Summan Ahmed , Hodgson Murray
Published: May 2013

Technology Educational Shops (TES) are designed to develop high school students’ technological literacy. Their acoustical conditions play a dominant role in the quality of these environments. TES are, at the same time, classrooms for learning and industrial workshops for making things. Each use has its own standards governing its acoustical characteristics: ANSI S12.60-2002 for classrooms and the Ondet & Sueur DL2 criteria for workshops. A major conflict could exist by using the same room for two different purposes. This study investigated this conflict by evaluating the acoustical characteristics of 20 unoccupied wood, metal and automotive shops. It conducted measurements of background noise level (BNL), reverberation time (RT), speech intelligibility index (SII), and DL2. Results showed that BNLs and RTs in most TES were higher than the acceptability criteria for unoccupied core learning spaces. SII values indicated bad/poor speech intelligibility for normal and raised voice levels and reasonable/good speech intelligibility for loud and shout voice levels. DL2 values were found acceptable in TES larger than 100 m2 in floor area. In general, these results indicate the poor acoustical conditions of TES as classrooms, and the need for special sound control measures.
Experimental verification of a wave-vector-frequency-domain nonlinear acoustic model
Authors: Jing Yun , Cannata Jon
Published: May 2013

In this paper, a recently developed wave-vector-frequency-domain method for nonlinear wave propagation is verified by underwater experiments. A specially designed focused transducer was used to generate short high intensity pulses. 2D scans were conducted at a pre-focal distance, which were later used as the input to the numerical model to predict the acoustic field at focal and post-focal zones. Adaptive attenuation was introduced to reduce the Gibbs effect. Graphic processing units (GPU) were also employed to speed up the computation. Good agreement was observed between the simulation and experiment.
Compressively excited acoustic wave propagation in a three-dimensional channel bifurcated by an elastic partition
Authors: Aho Katherine , Thompson Charles
Published: May 2013

Fluid motion resulting from compression of the exterior walls of a three-dimensional channel that is axially partitioned by a flexible membrane is examined. The axial variation in the acoustic impedance of the partition gives rise to a pressure gradient across the partition and generates an evanescent field at its surface. The effects of these evanescent modes and their impact on spatial wavenumber selectivity are of particular interest. The Green’s function for the pressure is given for the case of low fluid compressibility and channel high aspect ratio. In this limit, it is shown the evanescent modes may be modeled in terms of generalized functions. The complete solution of the pressure field inside the channel will be shown. Implication of the results on the development a MEMS transducer in which the displacement of membrane is used as the method of transduction is presented. [NSF Grant 0841392.]
Propagation of N-waves in a turbulent and refracting atmosphere with ground effects (laboratory-scale experiment)
Authors: Ollivier Sébastien , Salze Edouard , Blanc-Benon Philippe
Published: May 2013

Sound propagation in a refracting atmosphere leads to the formation of a geometrical shadow zone close to the ground, and of an illuminated zone above a limiting ray. Without turbulence, acoustic waves can be diffracted into the shadow zone close to the ground. When propagating through a turbulent atmosphere, it is known that waves can be distorted, scattered, or focused. In order to investigate how turbulence modifies the pressure field into the shadow zone, a well controlled laboratory-scale experiment has been performed. An electrical spark source is used to generate short duration (20 μs) and high pressure (1500 Pa) N-waves. A convex surface models the effect of an upward refracting atmosphere, and a heating grid generates thermal turbulence (1% fluctuations of refraction index). To compute statistics of wave parameters variation, seven 1/8 in. microphones have been used to record 2000 waves at each position after propagation through the turbulent field. Wave parameters (peak pressure, rise time) obtained with turbulence are compared to data obtained without turbulence. Results show that turbulence scatters sound into the shadow zone, which increases significantly the noise level.
The numerical simulation of propagation of intensive acoustic noise
Authors: Demin Igor , Sergey Gurbatov , Nikolay Pronchatov-Rubtsov
Published: May 2013

The propagation of intensive acoustic noise is of fundamental interest in nonlinear acoustics. Some of the simplest models describing such phenomena are generalized Burgers’ equations for finite amplitude sound waves. An important problem in this field is to find the wave’s behavior far from the emitting source for stochastic initial waveforms. The method of numerical solution of generalized Burgers equation proposed is step-by-step calculation supported on using Fast Fourier Transform of the considered signal. The general idea is to keep only Fourier image of concerned signal and update it recursively (in space). For simulating the wave evolution we used 4096 (212) point realizations and took averaging over 1000 realizations. Also the object of the present study is a numerical analysis of the spectral and bispectral functions of the intense random signals propagating in nondispersive nonlinear media. The possibility of recovering the input spectrum from the measured spectrum and bispectrum at the output of the nonlinear medium is discusses. The analytical estimations are supported by numerical simulation. For two different types of primary spectrum evolution of jet noise were numerically simulates at a short distance and assayed bispectrum and a spectrum analysis of the signals.
Weak lateral coupling between stereocilia of mammalian cochlear hair cells requires new stimulus methods to study the biomechanics of hearing
Authors: Karavitaki K. Domenica , Niksch Paul D. , Corey David P.
Published: May 2013

The forces felt by different transduction channels in a bundle depend critically on how well stereocilia remain cohesive during deflection. In the bullfrog saccule, sliding adhesion mediated by horizontal top connectors (HTC) confers coherent motion to hair cell stereocilia and parallel gating to all transduction channels. In cochlear inner and outer hair cells (IHCs and OHCs), the mature complement of HTC is established by postnatal day 12; they extend between adjacent stereocilia of both rows and columns. Contrary to our expectation that bundle cohesion should be robust in all directions, our experiments on gerbil cochleas show that lateral coupling among stereocilia of the tallest row is weak. These findings suggest that the function and molecular composition of the HTC in the cochlea are different from those in bullfrog hair cells. They also raise concern for current stimulus methods, which involve glass probes that are often small compared to the bundle width. Our data suggest that only stereocilia in contact with the probe are stimulated, and delivery of the stimulus to the remaining stereocilia is weak and inhomogeneous. To mimic the OHC stimulus delivered by the overlying tectorial membrane in vivo, we are developing new stimulation technologies.
Head movement during horizontal and median sound localization experiments in which head-rotation is allowed
Authors: Morikawa Daisuke , Toyoda Yuki , Hirahara Tatsuya
Published: May 2013

We measured subjects’ head movements during horizontal and median sound localization experiments in which head-rotation was allowed in order to know how they move their heads to localize sound in a head rotation condition. The head movements in a head-rotation condition were measured while localizing 500-Hz low-pass noise, 12-kHz high-pass noise, and white noise. With regard to horizontal plane, sound localization became easier with head-rotation than head-still condition. All subjects turned their heads toward the presented sounds, yet they did not necessarily turn their heads to face the sound. The amount of head rotation was small for WN compared to that for LPN and HPN. Sound localization also became easier with head-rotation than head-still condition for the median plane. All subjects swung their heads right and left centering on 0°, no matter what the stimulus elevation angle was.
Falling stars: Acoustic influences on meteor detection
Authors: Edewaard Darlene , Gordon Michael S.
Published: May 2013

As particles enter the earth’s atmosphere they produce a burst of electromagnetic energy, including visible and radio-wave emissions. Consequently, just as meteors can be detected visually in the night sky they can be “heard” using radio telescopes. The current project investigated the potential influence of these audio signals on meteor detection. Anecdotally, and in related research, it has been found that auditory signals can enhance or even alter visual perception of objects. The current project examined the specific effects of accompanying auditory signals on the detection of meteors. Meteors present an interesting case of audiovisual integration in that detection paradigms often entail extended vigilance and extremely brief, yet brilliant astronomical events. Experiments specifically investigated how auditory signals that varied in spectra influenced changes in visual magnitude and duration judgments of meteors. In addition, research targeted how extraneous auditory cues during a vigilant meteor search might contribute to false judgments. Results are described in terms of audiovisual integration and the relation of perceptual mechanisms to meteor detection.
Time-to-arrival discrimination of multiple sound sources
Authors: Gordon Michael S. , Edewaard Darlene , Pacailler Matthew
Published: May 2013

Previous research in auditory detection of time-to-arrival (TTA) has tended to focus on single sound sources approaching a listener. However, visual studies of TTA have suggested that there may be different perceptual strategies employed with respect to single versus multiple objects on approach. The current research is designed to directly address the capacity and informational support of listeners to make determinations on the TTA of multiple sound sources. Initial experimentation evaluated the capacity for discrimination between a stationary and moving sound source. Additional studies evaluated multiple moving sound sources, with direct manipulation of their spectra to determine how various acoustic qualities contribute to TTA determinations. Results suggest the relative influence of intensity, frequency, and sound source dynamics as they facilitate competitive discriminations between approaching sound sources. Additional conclusions demonstrate changes in the use of spectral cues for lower and higher levels of auditory clutter as they might impact TTA.
Performance of a highly directional microphone array in a multi-talker reverberant environment
Authors: Favrot Sylvain , Mason Christine R. , Streeter Timothy M. , Desloge Joseph G. , Kidd Gerald
Published: May 2013

A visual guided hearing aid (VGHA) has recently been developed, which uses an eye tracker to steer the “acoustic look direction” (ALD) of a beamforming microphone array. The current study evaluates the performance of this highly directional microphone in providing spatial release from masking (SRM) under acoustically dry and reverberant conditions. Four normal-hearing subjects participated in a speech intelligibility test with collocated and spatially separated speech maskers when listening either through the microphone array or through KEMAR to simulate “natural” binaural conditions. The results indicated that near-normal SRM was achieved by listening through the VGHA in both environments. In the acoustically dry condition, SRM was similar to the measured signal-to-noise ratio (SNR) gain from the microphone array. However, in the reverberant condition, subjects showed significantly greater SRM than predicted from the measured SNR gain from the array. This is consistent with the measured improvement in SNR for the early part of the room impulse response for the target but not for the spatially separated maskers. This indicates that in some reverberant conditions the microphone array provides substantial source selection benefits.
Head tracking and source localization in reverberant cocktail party scenarios
Authors: Parks Anthony , Braasch Jonas
Published: May 2013

From experience and observation, it is known that listeners actively explore their auditory environment through a variety of head movement strategies. Beyond the resolution of front-back confusions, little is known about the mechanisms by which head movement enables listeners perform a broad range of auditory scene analysis tasks. In this experiment, an attempt was made to look at one of these tasks: how well listeners can track a source by head movement. A three-talker paradigm was utilized in a headphone-based head-tracked virtual environment spatialized with HRTFs. The task involves a target that moves from trial to trial with two stationary interferers (male target, two female interferers, all speaking phonetically balanced sentences). The listener is prompted to move her head such that she believes she is facing the target. The two independent conditions are three levels of reverberation (anechoic, strong early reflections, strong late reflections) and seven target azimuth angles (from +45 to −45 degrees in steps of 15 degrees), while the measured responses include facing accuracy as a percentage and number of times the playback button is hit before each trial is completed (to indicate task difficulty). Results are discussed within the context of both room acoustics and perception.
Sound source localization: Bandwidth and envelope
Authors: Yost William , Zhong Xuan
Published: May 2013

Human listeners were asked to locate six sound sources separated by 15o in the right quarter field. Sound sources were located in a sound-deadened room (reverberation time <90 ms) at the height of the listener’s pinna 1.67 meters from the listener. In experiment 1, eight, 200-ms pure tones covering the frequency range from 250 to 7011 Hz were presented. In experiment 2, 200-ms noise bursts with different bandwidths (1/6, 1/3/, 1, and 2 octaves) at three center frequencies (250, 2000, and 4000 Hz) were presented. In experiment 3, 200-ms, 4000-Hz tones were presented with transposed envelopes with rates of 50, 100, 150, and 250 Hz. Several indicators of sound source localization performance were measured including root-mean-square (rms) error in degrees. RMS error decreased with increasing bandwidth from approximately 20 degrees for pure tones to approximately 6 degrees for 2-octave wide noises. RMS error depended on center frequency much more for narrow bandwidths than for broader bandwidths. RMS error decreased slightly from 50-Hz rate of modulation to 250-Hz rate of modulation. The data suggest that stimulus bandwidth is the primary variable effecting sound source localization performance in the free-field. [Research supported by an AFOSR Grant.]
The Haas Effect—Lateral extent and perceptual weighting of localization cues
Authors: Pastore M. Torben , Braasch Jonas
Published: May 2013

The Haas effect is a well-known manifestation of the precedence effect. Originally, Haas measured the echo threshold as a function of the primary auditory event and its single reflection being equally loud. What is not well known is the lateral position of the lead/lag pair as a function of inter-stimulus interval and level difference between lead and lag. How robust the Haas effect is in the localization dominance region was investigated, adjusting the level difference between lead and lag, and using 200 ms band-passed noise presented dichotically over headphones. In addition, the onset and offset cues were removed for half the trials and left intact for the other half to investigate the roles of onset and offset cues versus ongoing cues. Lateral displacement of the auditory event was recorded with an acoustic pointer. Analysis of these results help to reveal the perceptual weighting of localization cues in the Haas effect.
Factors influencing target detectability in realistic listening scenarios
Authors: Weller Tobias , Best Virginia , Buchholz Jörg M.
Published: May 2013

In psychoacoustics, there is an increasing demand for more realistic testing environments that better capture the real-world abilities of listeners and their hearing devices. However, there are significant challenges involved in controlling the detectability of relevant target signals in realistic environments. We conducted an extensive detection study in a simulated real-world environment to understand some of the important dimensions influencing detection. A multi-talker cafeteria scene was generated using Room simulation software and played back by means of a 3-D loudspeaker array. Detection thresholds for the target word “two” were measured adaptively for eight different target directions in the horizontal plane. Performance was then measured for fixed signal-to-noise ratios around these thresholds to obtain a psychometric function for each direction. To examine the effect of target-location uncertainty, psychometric functions were also measured with randomized target directions. Detection thresholds depended on the target direction, consistent with changes in signal-to-noise ratio caused by the head shadow. Target-location uncertainty increased thresholds globally by a small amount. These findings provide a framework for controlling the detectability of target sounds in future experiments aimed at measuring localization, identification, awareness, etc. in realistic listening environments.
Simulation of the head-related transfer functions using cloud computing
Authors: Huttunen Tomi , Tuppurainen Kimmo , Vanne Antti , Ylä-Oijala Pasi , Järvenpää Seppo , Kärkkäinen Asta , Kärkkäinen Leo
Published: May 2013

Due to the complexity of measurements for obtaining individual head-related transfer functions (HRTFs), numerical simulations offer an attractive alternative for generating large HRTF data bases. In this study, HRTFs are simulated using a fast multipole boundary element method (BEM). The BEM is well suited for the HRTF simulations. Namely, only the surface of the model geometry is discretized which simplifies the pre-processing compared to other full-wave simulation methods (such as finite element and finite difference methods). The BEM is formulated in frequency domain and the model is solved separately for each frequency. Since a large number of frequencies is needed in wide-band HRTF simulations, the BEM simulation greatly benefits from distributed (or parallel) computing. That is, a single computing unit takes care of a single frequency. In this study, a distributed BEM using cloud computing is introduced. Simulations are computed in a public cloud (Amazon EC2) using a realistic head and torso geometry (3D laser scanned geometry of Bruel & Kjaer HATS 4128 mannequin). The frequency range of the simulations is from 20 to 20000 Hz. The feasibility of cloud computing for simulating HRTFs is examined and first analysis results for the simulated HRTFs are shown.
Simplification of head-related impulse response in early reflection simulation
Authors: Zhang Liang , Zhong Xiao-li
Published: May 2013

In virtual auditory environment, early reflections are usually simulated by the image-source method, and binaural signals are synthesized by convolving the input stimulus with corresponding head-related impulse responses (HRIRs). Considering the limited resolution of the human hearing, this work investigates the minimal length of HRIRs needed in early reflection simulation via a simple model consisting of a single direct sound and a reflection. The direct sound is synthesized using 512-point HRIR and fixed at the position directly in front of the subject, while the reflection is synthesized by HRIRs at various directions with four time-domain lengths (512, 256, 128, and 64 points, at a sampling frequency of 44.1 kHz) as well as five time delays relative to the direct sound (from 10 to 50 ms at intervals of 10 ms). A three-interval, two-alternative forced-choice paradigm is employed in this work. Results indicate that for most spatial directions the HRIR with a length of 64-point is perceptually adequate in early reflection simulation. [Work supported by State Key Laboratory of Subtropical Building Science, South China University of Technology, Grant No. 2013KB23.]
Estimation of spectral notch frequencies of the individual head-related transfer function from anthropometry of listener's pinna
Authors: Ishii Yohji , Iida Kazuhiro
Published: May 2013

Listener’s own head-related transfer functions (HRTFs) are necessary for accurate sound image reproduction. The HRTFs of other listeners often cause the front-back confusion and the errors in elevation perception. It is, however, impractical to measure the HRTFs of any listener for any sound source direction because the measurement requires special apparatus and much time. On the other hand, the estimation of the entire spectrum information of listener’s own HRTF still remains as an unsolved difficult issue. One of the authors has shown that the simplified HRTFs, which is recomposed only of the first spectral peak around 4 kHz (P1) and the lowest two spectral notches (N1 and N2) above P1, extracted from the listener’s measured HRTFs in the median plane, provide almost the same localization accuracy as the measured HRTFs. While the frequency of P1 is almost constant independent of the sound source elevation and the listener, those of N1 and N2 are highly dependent on both the elevation and the listener. The present study proposes a method, which estimates the frequencies of N1 and N2 in the median plane for the individual listener from the anthropometry of the listener's pinna, and examines the validity of the method.
Numerical modeling of the radiation by a submerged fluid-filled cylindrical shell: Observation of the S0, A0, and A waves
Authors: Iakovlev Serguei , Santos Hugo A. F. A. , Schulman Benjamin , Williston Kyle
Published: May 2013

We consider a submerged fluid-filled cylindrical shell subjected to an external acoustic pulse and analyze the structure of the field radiated by the shell into the fluids, both external and internal. We first propose a computationally efficient semi-analytical model of the interaction based on the Reissner-Mindlin shell theory combining some of the classical methods of mathematical physics with the finite-difference methodology, and then use the model to simulate the interaction. We demonstrate that the model accurately reproduces the wave structure of the radiated fields seen in the experiments for submerged evacuated shells, namely, both the symmetric Lamb waves S0 and the pseudo-Rayleigh waves A0. It is further observed that the internal and external wave patterns associated with the A0 waves exhibit the same alternation of the equiphase lines as the one seen in the experiments for a plate loaded by the fluid on both sides, a result that seems to be particularly relevant in the context of very limited number of experimental images of the radiated field for shells loaded by fluid from both inside and outside. Not less interestingly, we also demonstrate that the Scholte-Stoneley, or A, wave is also reproduced by the model, and we offer some insights into the non-observability of this wave for certain types of cylindrical shells reported in earlier experimental studies.
Localization and identification of three-dimensional sound source with beamforming based acoustic tomography
Authors: Ding Hao , Lu Huancai , Li Chunxiao , Jing Jiangming , Mei Dongting , Chai Guozhong
Published: May 2013

Beamforming based commercial planar microphone array could only localize and identify the sound source when the distance between source and array is known. This paper presents a beamforming based acoustic tomography (BBAT) method to locate and identify the source in 3D space, say, the BBAT method can not only locate the source in X-Y plane that is parallel to the array, but also the depth Z of the source. In this method, the sound field is reconstructed on the virtual planes at different distances along depth direction (Z direction). The maximum response of sound field on every virtual reconstruction plane is tracked, where the largest value among those maximum responses appears at Z direction is the depth of the source. The location of source at X and Y directions can then be easily identified based on beamforming principle, which is utilized by commercial planar array. The BBAT method is evaluated theoretically by simulation of monopole source, the experimental evaluation is done as well in anechoic chamber. The results from both simulation and experiment indicate that this method is capable to locate and identify sound source in 3D space. However, it cannot recognize the sound source located in front of the planar array or behind because of the genetic limitation of 2D planar array in identification of source depth.
Intonation perception in English: Effects of stimulus amplitude and listeners’ language background
Authors: Morrow Katherine , Liu Chang
Published: May 2013

The contour of fundamental frequency (F0) of the final word is the primary acoustic cue for intonation production and perception in English utterances. On the other hand, speakers of Mandarin Chinese may have to use other acoustic cues such as amplitude and duration rather than F0 contours to differentiate intonation contrasts since F0 contours carry lexical meaning in Mandarin Chinese. The goal of this study was to examine the role of the final word amplitude in intonation perception of English sentences. The final word amplitude was manipulated at three levels relative to the carrier sentence: −6, 0, and +6 dB. F0 contours of the final word were controlled continuously from falling to rising patterns. Listeners’ task was to identify the sentence intonation: question or statement. Preliminary results showed the intonation boundary shifted from slightly falling F0 contours to slightly rising F0 contours as the final word amplitude decreased for Chinese listeners, but the boundary did not change for the three amplitudes for English listeners. These results imply that Chinese listeners may use the final word amplitude as a secondary cue to perceive intonation contrasts in English, while English listeners may primarily rely on F0 contours for intonation perception.
Effects of supraglottic compressions on the aerodynamics and acoustics of excised canine larynges
Authors: Alipour Fariborz , Finnegan Eileen
Published: May 2013

The purpose of this study was to examine the aerodynamic and acoustic effects due to supraglottic compressions, which may be seen in some dysphonic patients. Canine larynges were prepared and mounted and vocal fold oscillations were generated and controlled by the flow of air through the glottis. Glottal adduction was accomplished by rotating the arytenoids with a suture passed behind the vocal folds to simulate lateral cricoarytenoid muscle action. Supraglottic medial and anterior-posterior compressions were accomplished by manual squeezing at the arytenoid level and alternating between the rest and compressed conditions. The raw data, including EGG, subglottal pressure, flowrate, and microphone signals, were recorded on a DAT tape and later digitized and processed with Matlab. A video image of the superior aspect of the larynx was recorded using a stroboscopic light during the whole experiment. Results indicated that the excised larynges oscillated better and easier without the false vocal folds, but generated louder sound with false vocal folds. Medial compression always resulted in increased subglottal pressure, decreased flow rate and most often increased the sound intensity, but decreased EGG closed quotient. Both of these compressions had negative effects on the amplitude of EGG signal, suggesting disruption of vocal fold contact.
Effect of source viewing on the unmasking of speech
Authors: Chorubczyk Alan , Sato Shin-Ichi , Cardozo Maira
Published: May 2013

In this work, it is analyzed if the subject visual awareness of the speaker presence would result in an enhancement of the speech understanding capability. For this purpose, a pair comparison test with different scales of speech intelligibility conditions with and without source viewing was conducted.
Detection of obstructive sleep apnea by estimation of oral and nasal cavity cross-section areas from acoustic recordings of snore
Authors: Huang Hsu-Kang , Liu Yi-Wen , Chiang Rayleigh Ping-Ying
Published: May 2013

Obstructive sleep apnea (OSA) refers to the condition in which a person’s breathing is paused while asleep, or the airflow is decreased, due to obstruction in the upper respiratory airway. In severe cases, OSA can cause complete arousals and deprive the patient from normal sleep. Surgical intervention is sometimes recommended, but accurate identification of the site of obstruction can be difficult. In the present study, we devised signal processing methods to estimate the site and the severity of airflow obstruction from recordings of sounds of snore. The vocal tract, the oral and the nasal cavity are modeled as three branches joining at the pharynx. Each branch consists of cylindrical segments whose cross-section areas can vary during snoring. Estimation of these cross-section areas consists of two steps: First, an auto-regressive moving-average method is applied to find the linear coefficients of a pole-zero model that optimally accounts for the recorded sound. Then, the Levinson-Durbin algorithm is applied to convert the coefficients to ratios of cross-section areas between adjacent segments. The present method is applied to a set of recorded snore samples during clinically confirmed apnea episodes, and results are compared with those of simple snore. Effectiveness of the method is analyzed statistically.
Study of unvoiced fricative speech production: Influence of initial conditions on flow development
Authors: Fujiso Yo , Van Hirtum Annemie , Nozaki Kazunori , Wada Shigeo
Published: May 2013

Human unvoiced fricative speech sounds such as [s] and [f] are produced by a complex fluid-structure interaction. Indeed, a moderate Reynolds number (100 ≤ Re ≤ 10000) turbulent jet is issued from a constriction somewhere in the vocal tract formed between the hard palate and an articulator such as tongue, teeth, or lips. By using simplified in-vitro replicas representing parts of the human vocal tract, some physical phenomena relevant to the unvoiced fricative speech production can be reproduced and more easily understood. The current study focuses on the influence of initial conditions on flow development by performing flow measurements and Large Eddy Simulation on a rectangular channel containing a tooth-shaped obstacle.
Frequency-sum beamforming in an inhomogeneous environment
Authors: Abadi Shima H. , VanOverloop Matthew J. , Dowling David R.
Published: May 2013

The arrival directions of ray paths between a sound source and a receiving array can be determined by beamforming the array-recorded signals. And, when the array and the signal are well matched, directional resolution increases with increasing signal frequency. However, when the environment between the source and the receivers is inhomogeneous, the recorded signal may be distorted and beamforming results may be increasingly degraded with increasing signal frequency. However, this sensitivity to inhomogeneities may be altered through use of an unconventional beamforming technique that manufactures higher frequency information by summing frequencies from lower-frequency signal components via a quadratic (or higher) product of complex signal amplitudes. This presentation will describe frequency-sum beamforming, and then illustrate it with simulation results and near-field acoustic experiments made with and without a thin plastic barrier between the source and the receiving array. The experiments were conducted in a 1.0-meter-deep and 1.07-m-diameter cylindrical water tank using a single sound projector, a receiving array of 16 hydrophones, and 100 micro-second signal pulses having nominal center frequencies from 30 to 120 kHz. The results from frequency-sum beamforming will be compared to the output of conventional delay-and-sum beamforming for different center frequencies. [Work sponsored by ONR and NAVSEA.]
Underwater source localization using a hydrophone-equipped glider
Authors: Jiang Yong Min , Osler John
Published: May 2013

Buoyancy-driven underwater gliders are autonomous underwater vehicles that were originally developed to collect oceanographic data. CMRE is studying the use of this technology for the characterization of denied areas, including alternate sensor payloads and applications. During the Rapid Environmental Assessment phase of the Noble Mariner 2012 NATO exercise, Conducted in Gulf of Lions in September 2012, an omnidirectional hydrophone was mounted on a shallow water glider to sample the spatial distribution of the acoustic and oceanographic fields at different ranges and depths. This paper presents a study of the potential to localize acoustic sources by using the acoustic and environmental data collected by the glider. During the experiment, a bottom moored acoustic source was deployed in an area with benign bathymetry. Continuous wave and frequency modulated pulses were broadcast for approximately 6 h. The glider was flying along predefined tracks and the distances from the source were typically from 5 to 9 km. A Ray tracing model is used to evaluate the arrival structures of the acoustic signal, and to estimate the source location. The impact of the range dependent water column sound speed profile on the uncertainty of the source localization is also discussed.
Three-dimensional acoustic propagation under a rough sea surface
Authors: Ballard Megan S.
Published: May 2013

A three-dimensional propagation model using stepwise coupled modes is applied to calculate the acoustic field under a rough sea surface. The model is formulated in a cylindrical coordinate system and the solution for the three-dimensional acoustic field is approximated by accounting for mode coupling in the radial direction and including horizontal refraction in the azimuthal direction. The atmosphere above the sea surface is modeled as an acoustic half space having the properties of air and sea surface height is allowed to vary arbitrarily as a function of range and azimuth. For the sea surfaces presented in this work, the amplitude spectrum of the surface waves is modeled according to the JONSWAP spectrum and the directionality is included by assuming cosine-squared spreading. The acoustic field is calculated for sea surfaces determined for varying levels of wind intensity and fetch. A modal decomposition of the acoustic field is used to provide insight into the effects of the rough sea surface on the predicted transmission loss. The importance of using three-dimensional versus two-dimensional models for acoustic propagation under rough sea surfaces is investigated. [Work supported by ONR.]
Acoustic interface treatment with an adjoint operator for linear range-dependent ocean index of refraction inversions
Authors: Richards Edward , Potty Gopu
Published: May 2013

Hurskey et al. [J. Acoust. Soc. Am. 115(2), 607–619 (2004)] introduced the adjoint method and incorporated local sound-speed measurements into range-dependent ocean sound-speed inversion. They left two practical issues for the implementation of this method unresolved in this paper: the collection of appropriate environmental measurements and the implementation of bottom boundary conditions. The first of these issues was considered in a simulation study that used an oceanographic glider to collect range-dependent sound-speed measurements [Richards, CMRE memorandum (2012)]. A covariance matrix was constructed from the changes observed in the range-dependent sound-speed field. The adjoint inversion was performed in a reduced element subset of the empirical orthogonal functions (EOF) base of the covariance matrix. The second issue is the focus of this paper, which describes a bottom boundary condition with a defined adjoint operator. A horizontal fluid-bottom interface is implemented using the implicit finite difference form of the parabolic equation introduced by McDaniel and Lee [J. Acoust. Soc. Am. 71(4), 855–858 (1982)]. Combined with local range-dependent sound-speed statistics gathered with gliders, this development may provide a method of near real-time acoustic measurement of ocean sound-speed variations between an acoustic source and vertical hydrophone array.
The differences and though the equivalence in the detection methods of particle, ray, and beam tracing
Authors: Stephenson Uwe M.
Published: May 2013

Within the numerical methods used in room acoustics, the geometrical and energetic methods of sound particle, ray and beam tracing are often confused. This rather tutorial paper does not treat the tracing algorithms but rather aims to explain the differences in the physical models and the corresponding detection and evaluation methods. While ray tracing needs spherical detectors as receivers to count rays, the particle model is based on a weighting of the energies of the particles with their inner crossing distances to compute the local sound energy densities. For beam tracing, receiver points are sufficient. In its core, this paper shows the convergence of the evaluated intensities computed from immitted sound particle energies to those predicted by the well-known 1/R<+>2<+>-distance law for the free field—as applied with the mirror image source method and beam tracing as its efficient implementation. Finally, the geometrical methods are classified depending on their efficiency with higher orders of reflection and their extensibility by scattering and diffraction.
Comparison of sound field measurements and predictions in coupled volumes between numerical methods and scale model measurements
Authors: Luizard Paul , Otani Makoto , Botts Jonathan , Savioja Lauri , Katz Brian F.
Published: May 2013

Prediction of sound fields in closed spaces can be achieved by various methods, either physical or numerical, based on different theoretical features. While the benefits and limitations of many methods have been examined for single volume spaces, there has been little effort in examining these effects for coupled volume situations. The present study presents a case study comparing theoretical, experimentally physical measurements on a scale model, and various numerical methods, namely boundary element method (BEM), finite-difference time-domain (FDTD), and ray-tracing through the commercial software CATT-Acoustic and ODEON. Although these numerical methods all use 3D numerical models of the architecture, each is different. Ray-tracing is more suitable to geometries with larger planes; BEM requires a more regular finer surface mesh; and FDTD requires a volumetric mesh of the propagation medium. A simple common geometry based on the scale model is used as a basis to compare these different approaches. Application to coupled spaces raises issues linked to later parts in the decay due to multi-slope decay rates, as well as diffraction phenomenon due to acoustic energy traveling between coupling surfaces from one volume to another. The ability of these numerical methods to adequately model these effects is the question under study.
Acoustic propagation modeled by the curvilinear Fourier pseudospectral time-domain method
Authors: Hornikx Maarten , Steeghs Daan
Published: May 2013

The Fourier pseudospectral time-domain method is an efficient domain-discretization wave-based method to model sound propagation in inhomogeneous bounded media. The method was successfully applied to model atmospheric sound propagation and acoustics in urban environments. One of the limitations of the method is its restriction to a Cartesian grid, confining it to staircase-like geometries. When applying a transform from the Cartesian coordinate system to the curvilinear coordinate system, more arbitrary geometries may be solved by the method. In free field, the frequency dependent accuracy of the curvilinear Fourier pseudospectral time-domain method is investigated as a function of the deformation angle of the grid. Further, the performance of the pseudospectral method with a curvilinear grid as well as a Cartesian grid for scattering of elementary objects as an inclined plate and a cylinder is studied. Finally, sound propagation in a room with non-parallel boundaries and over a building with gabled roof is computed with the pseudospectral method with a curvilinear grid and compared with results obtained from the boundary element method. All computed results are in 2D.