bluuebunny/crossref_metadata_2025 huggingface dataset:
Vowel normalization using stop consonant loci.
Authors: Koops Christian
Published: October 2009

Vowel normalization techniques used in sociophonetics aim to eliminate the influence of vocal tract length differences on resonant frequencies so that speaker-specific vowel configurations can be compared more precisely. In vowel-extrinsic methods, a speaker’s F1-F2 range is scaled up or down, and vowel spaces are aligned using the geometrical F1-F2 mean. However, a speaker’s F1-F2 mean may be systematically biased if his or her vowel configuration is affected by a set of parallel vowel shifts, e.g., fronting or raising of several vowels in a chain shift. To address this problem, an alternative method of deriving speaker-specific scaling factors was tested based on a vowel-independent acoustic measure: the locus (virtual origin) of the second formant transition following the /d/-release. Forty adult speakers of American English (20 male, 20 female) were recorded reading ten repetitions of 16 /dVd/-words. From these data, individual F1-F2 means and locus equations were calculated, including each speaker’s F2 locus frequency for /d/. Preliminary results based on the corner vowels /i, a, u/ show that F2 locus and F1-F2 mean are strongly correlated (r-squared=0.84). Statistical modeling is underway to determine whether scaling factors derived from locus analysis are superior to scaling factors derived from F1-F2 means.
Acoustic correlates of devoiced Japanese vowels.
Authors: Varden J. Kevin
Published: October 2009

The analysis of acoustic correlates of devoiced vowels produced by young native Japanese speakers is presented. In addition to gradient amounts of voicing seen in vowels undergoing devoicing, gradient amounts of fricativization can be observed, supporting an analysis of devoicing in Japanese as gestural overlap of both glottal gesture and oral stricture. Specifically, the surfacing of voiceless consonant-devoiced vowel sites as voiceless fricatives, evidenced by high-frequency frication in spectrograms and a significantly higher number of zero crossings, will be discussed, as will the interplay between devoicing and fricativization in the data set. Analysis of the formant frequency information at the vowel sites, preserved in the frication and allowing the perception of the original vowels even in the face of heavy loss of vocality, is also presented.
Articulatory variability within experimental sessions.
Authors: Lofqvist Anders
Published: October 2009

This paper examines articulatory and acoustic variability within recording sessions lasting about 25 min, with particular emphasis on overall drift during the session. Articulatory movements were recorded using a magnetometer system. The subjects were five speakers of Japanese, producing words with long and short consonants. Fifty tokens of several utterance types were obtained. For this analysis, ten tokens were binned in five different bins according to where they occurred during the recording session. Statistical analysis showed that for the five speakers, differences between the bins occurred for between 30% and 60% of the utterances and measurements. Most differences occurred for acoustic measures of segment duration and for a kinematic measure of tongue movement duration. During the course of the recording session, segment durations mostly decreased, but the patterns differed both within and across speakers. These results are useful for planning experiments, and for Power analysis of the number of tokens required for statistical analysis. [Work supported by NIH and the Swedish Research Council.]
High-resolution time delay estimation using a combined correlator-based and phase measurement approaches.
Authors: Assous Said , Hopper Clare , Lovell Mike , Gunn David , Jackson Peter , Rees John
Published: October 2009

Time delay estimation is a fundamental step in source localization or beamforming applications. It has attracted considerable research attention over the past few decades in many fields including radar, sonar, seismology, geophysics, and ultrasonics. Various techniques are reported in the literature, mainly the correlator-based methods. However, with these approaches, when the time delay is not an integral multiple of the sampling rate, it is necessary to either increase the sampling rate or use interpolation to get better resolution. In this paper, we present a new approach for time delay estimation based on combining the correlator-based technique and the phase information obtained from the received signal. A cross-correlation based approach is applied first to get an initial estimate then the DFT is applied to a window of the signal starting at this initial estimate to calculate a related phase. This phase will define an additional fractional inter-sample time delay, which is corrected to provide more accurate estimation. Simulated examples are given with synthetic signals. An underwater application for distance and speed of sound measurements is then demonstrated. Finally, the proposed method is shown to outperform the correlator-based approach and a high resolution is achievable without the need of oversampling or interpolation.
Active clutter reduction through fusion with passive sonar data.
Authors: Aughenbaugh Jason M. , Yocom Bryan , La Cour Brian R. , Yudichak Tom
Published: October 2009

Active and passive sonar systems provide different types and qualities of data. For example, passive sonar typically provides bearing information and classification clues. Active systems can provide the complementary range and Doppler shift information. Similarly, the types of false alarms—passive interferes and active clutter—may be from different objects. The combination of information from both systems should yield a better estimate of the target state, but traditionally, such systems are used separately. A method for fusing active and passive sonar data is presented with example results. The fusion engine is a Bayesian tracker. The key step for fusing active and passive data is the formation of the likelihood function for each type of measurement. This process is described at a high level, but the emphasis on the results of fusing the data. The example problem involves simulated active and passive data, including the presence of passive interferers and active clutter objects. The results show the value of using active systems to refine passive estimates and more importantly suggest the potential power of using passive data to mitigate active clutter. [This work was supported by the Office of Naval Research Contract No. N00014-06-G-0218-01.]
Research on the applications of time reversal method for noise control by use of micro-electro-mechanical-system array microphones for cell phones.
Authors: Hsieh Yu-Hao , Lin Sheng-Che , Wu Bo-Hsien , Too Gee-Pinn , Chang Shyang-Jye
Published: October 2009

The purpose of this research is to construct a noise reduction system in 3G cell phones that combines time reversal method (TRM) and the micro-electro-mechanical-system (MEMS) microphone array. Time reversal method has advantages of compensate distortion due to path effect in propagation and signals focus at the original location. This research contains theoretical reasoning, TRM simulations, and experiments. The main point of this study is to estimate the impulse response function accurately to form a virtual path model, because desired signal needs to be obtained at listener’s location, not source location. The simulations present that the procedure is effective under different environmental conditions and parameters. Both normalized cross-correlation coefficient (NCCC) and signal-to-noise ratio (SNR) are used to estimate the simulation result. In the experiments, a MEMS microphone array is constructed and simulates the configuration as it is on the cell phone. The results demonstrate that the procedure can separate a specific source from a combination signal of multiple sources, and increase the NCCC and SNR of the specific source signal. Finally, the enhancement level of the result is discussed. [This study is supported by Center of Nanotechnology and Microsystems Engineering, National Cheng Kung University, with the Project no. 98C066.]
Improving mp3 capability of mobile phones by linking acoustic information with vibrations.
Authors: Kim Cheol Hong , Park Young Jin , Kim Jong-Myon
Published: October 2009

Consumers around the world have strong demand for mobile devices such as mobile phones and mp3 players. For this reason, the mp3-playing mobile phone has become one of the most popular mobile devices, especially for the younger generation. In this study, we propose an efficient method of utilizing small vibration motors embedded in the mobile phones to improve mp3 capability. The small vibration motors are generally embedded in the mobile phones to implement tangible user interface. The proposed method, which links acoustic information with areas on the mobile phone surface, utilizes the existing embedded small vibration motors to enhance mp3 capability. The use of additional small vibration motors can maximize mp3 capability but can cause additional power consumption and heat dissipation in the mobile phones. Therefore, the proposed method utilizes the pre-embedded vibration motors to generate vibrations according to the acoustic information from mp3. Experimental results, performed by using up-to-date mobile phone platform over 50 persons, demonstrate that the proposed method is suitable for mp3-playing mobile phones in case that the users want to feel vibrations with mp3. [Work supported by the MKE, Korea, under the ITRC supervised by the IITA [IITA-2009-(C1090-0903-0008)].]
Backscattering by a solid aluminum cylinder near a boundary: Experiment and finite element model comparisons.
Authors: La Follett Jon , Marston Philip L.
Published: October 2009

Modeling can be a useful tool for understanding the mechanisms that affect target backscattering. Comparing model predictions with experimental data serves to build credibility for the modeling approach and can give an estimation of accuracy. Measurements were made of backscattering by a solid aluminum cylinder, having a length to diameter ratio of 2. The target was suspended just below the surface of a water tank to simulate some aspects of backscattering when resting on the seabed. The target was rotated in a plane parallel to the free surface in both the free field and proud target configurations to obtain the spectral response (color) as a function of tilt. These measurements were compared with predictions made by a previously developed approximate finite element method [Williams et al., J. Acoust. Soc. Am. 125, 2701 (2009)] designed to model axisymmetric targets. Model predictions of acoustic color were able to accurately produce many of the bright features present in the tank data. However, for the proud configuration at broadside and larger values of wavenumber radius product, ka greater than 6, there are some discrepancies. [Research supported by ONR.]
Bistatic line-scan holography of scattering by metallic objects near a free surface.
Authors: La Follett Jon , Marston Philip L.
Published: October 2009

Bistatic scattering measurements were performed with metallic targets in water located near a free surface to simulate aspects of scattering by a proud target resting on ocean sediment. A water filled cylindrical aluminum shell and a solid aluminum cylinder were studied, both having a length to diameter ratio of 2. In addition a solid 45 deg aluminum cone was studied. Previously, an acoustic holographic method [Baik et al., J. Acoust. Soc. Am. 121, 3203 (2007)] was applied to bistatic scattering measurements where the source, free field target, and receiver scan line were all contained in a plane. That method was shown to be helpful for identifying contributions to the scattering. In the present work, this method is modified to analyze bistatic scattering by proud targets where both the source and receiver are at a grazing angles with respect to the free surface. The holographic images are compared with bistatic synthetic aperture sonar images for certain targets. Some of the image features were caused by the proximity of the target to the surface. [Work supported by the Office of Naval Research.]
Evolution of navy diver exposure standards for deterministic underwater sound in the 100–500 Hz band.
Authors: Pestorius Frederick M. , Fothergill David , Cudahy Edward
Published: October 2009

The US Navy developed low-frequency active (LFA) sonar that had the potential for adverse exposure of divers to high levels of LFA sound. A number of studies into the effects on divers of deterministic waveforms in the frequency band 100–500 Hz were conducted. Animal, human diver, and modeling underwater sound exposure studies were completed across a broad array of organ systems. Human diver studies have investigated underwater hearing thresholds, aversion, wet suit attenuation, and psychophysical response. Tests have included both laboratory and field studies at sound pressure levels up to 160 dB re 1 μPa. Ultimately, over 500 exposures using over 100 different test subjects were conducted. These tests yielded a baseline set of guidelines for the exposure of Navy and recreational divers to deterministic sonar transmissions in the subject test band. A summary of the initial and subsequent testing and a tracing of the evolution of the current guidelines will be presented.
Detection and localization of a target in shallow water between two source-receive ultrasonic arrays: A small-scale experiment demonstration.
Authors: Marandet Christian , Roux Philippe , Nicolas Barbara
Published: October 2009

We experimentally demonstrate the detection and localization of a wavelength-size target in a shallow ultrasonic waveguide between two source-receive arrays. The waveguide represents a 1.5-km-long, 50-m-deep ocean acoustic channel at the 1/1000 scale, in which two coplanar arrays would record the broadband transfer matrix around 3 kHz. Invoking reciprocity principle, we perform a time-domain double beamforming algorithm simultaneously on the source and receive array. This array processing projects the multi-reverberated acoustic signals recorded between each source and each receiver into an equivalent set of eigenrays defined by their launch and receive angles. Comparison is made between the intensity of each eigenray with and without the target in the ultrasonic waveguide. When detection is achieved, localization is performed through the back projection of each eigenray into the acoustic waveguide weighted by the intensity fluctuation induced by the target. The use of the diffraction-based sensitivity kernel for each eigenray improves the localization of the target. Detection and localization experimental results will be shown in the presence of surface waves for ultrasonic arrays at 1 and 3 MHz.
Impact of target scattering properties on active waveguide invariance.
Authors: Campbell Richard L. , Zurk Lisa M.
Published: October 2009

The waveguide invariant is a parameter which quantifies time-frequency structure observed in broadband, time-varying sonar geometries. This structure is only weakly dependent on the specific properties of the ocean waveguide, and thus processing schemes utilizing this invariant relationship between range and frequency have the potential to be robust in unknown or poorly known environments. Recent work has extended the concept of waveguide invariance to active sonar, and experimental demonstrations in near-monostatic active geometries using a vertical plate scatterer have been presented (Quijano, 2008). In this study, scaled tank experiments are extended to true bistatic geometries using a variety of canonical scatterers (i.e., spheres, plates, and cylinders), in order to explore effects that are unique to the bistatic scenario. The resulting spectrograms indicate that the target scattering matrix can be the dominant influence on the time-frequency structure. Results from normal mode and raytracing simulations are presented to interpret the tank data and predict the striation structure in realistic ocean environments.
Podcasting demystified: From concept to production for every budget.
Authors: King Kathleen P.
Published: October 2009

Podcasting offers an inexpensive and yet powerful way to reach people around the globe with your message in audio or video format. Leveraging our global societies’ access of 24/7 web-based platforms to meet their information and learning needs provides a vibrant distribution engine waiting for people of all disciplines to create and post their work. This tutorial will provide an overview of the basic “how-to’s” of planning, producing, and hosting your podcast episodes and series. Podcasting possibilities may include redistributing content, and events, supplementing learning experiences, extending the traditional classroom, and integrating new opportunities. Intellectual property, copyright issues, sound production, equipment options for every budget, free software for recording, editing, and production, and resources will be discussed as well. The author hosted and produced over eight series, reached over 6 million people through her podcasting work, written two books, and presented many seminars on this, other digital media, and distance learning topics. Utilizing this experience, insight into the many different formats, audiences, hosting features, and applications of podcasting will be presented. As examples of applications, consider that podcasting can be thought of as teacher-created, student-created, or already existing content. We will discuss how you might use these different forms for different valuable purposes in teaching and learning settings.
Case study in classroom acoustics measurements.
Authors: Good Kenneth , Roy Kenneth
Published: October 2009

Recent work was done related to exploring variations and interpretations of measurement techniques in a classroom environment. The primary focus of the work was related to quantifying room to room isolation; however, minimal background noise and reverberation time measurements were included. This presentation will discuss the differences in the measurements methods, results, and when each may be appropriate.
Acoustics in a high school gymnasium.
Authors: Hoeman Stephanie , Birney Jon , Schultheis Hannah , Kanter Shane , Coffeen Bob
Published: October 2009

A high school gymnasium constructed with unintentionally interesting acoustics was recently scheduled for a new sound reinforcement system. It was determined that the sound reinforcement system would be of little consequence until the architectural acoustics of the space were addressed. The gymnasium was constructed of entirely concave shapes which resulted in severe sound focusing. The space was extremely reverberant and exhibited many distinct, distracting reflections. Impulse responses were made in the space and computer models were made. The goal was to find a solution to reduce unwanted reflections and shorten the reverberation time. Since this project was for a public school with limited funds, utmost importance was placed on a solution that would be economical and uncomplicated to install. Acoustical analysis and auralizations of the space in its current condition and the projected performance of recommendations were compared to find an effective, cost-efficient solution the school board could implement to improve the architectural acoustics before the sound reinforcement system was added.
Optimizing the signal-to-noise ratio in speech rooms using passsive acoustics.
Authors: D’Antonio Peter
Published: October 2009

Adults with normal hearing require a roughly 0-dB signal-to-noise ratio for good speech intelligibility in classrooms and lecture halls. However, significantly higher values may be needed to compensate for neurological immaturity, sensorineural and conductive hearing losses, language proficiency, and excessive reverberation. ANSI 12.60 addresses ways to lower the noise interference due to background levels and reverberation time. However, it is also possible to increase the signal, by reflecting or diffusing early reflection. While speech power is delivered in the vowels which are predominately in the 250–500-Hz frequency range, speech intelligibility is delivered in the consonants, which occur in the 2–4-kHz frequency range. Therefore, effective core learning designs can incorporate scattering surfaces, rather than surfaces that absorb in the 2–4-kHz region, on the front wall, lower side walls, and central ceiling areas, to increase the speech signal. The decay time can be controlled with broadband absorption on the perimeter of the ceiling and upper wall surfaces. A computer model analysis of various speech environments will be presented.
The acoustic treatment in classroom refurbishment: A double blind experimental study examining the acoustic and auditory environment of the cellular classroom.
Authors: Canning David C.
Published: October 2009

This paper will present data from the Essex Study of mainstream schools designed for inclusion of children with special hearing difficulties. It will present data from the experimental study which was set up to guide the specification of classroom acoustic performance standards. The UK Special Educational Needs Legislation requires education authorities to meet the special educational needs of all children. The major need considered in the provision of children with special hearing requirements, including deaf and hard of hearing children, is their ability to function when there is competing sound. Control of competing sound is in part the job of the teacher, as the dominant source of sound is created by the occupants, but the acoustician has a significant role to play. The acoustic performance of a classroom has considerable impact on every aspect of teaching and learning and ultimately on whether a child’s special educational needs can be met within an inclusive educational setting. The findings of the study have implications for the acoustical performance of all schools, given that every school might reasonably be expected to provide for children with special hearing requirements. [The support of Sweyne Park School, Essex County Council, The National Deaf Children’s Society, and the Federation of Property Services is acknowledged.]
Auralizing adult-child listening differences.
Authors: Nelson Peggy B. , Sacks Jonah , Hinckley Jennifer
Published: October 2009

Substantial data from previous research show that children and adults require different acoustical conditions for good understanding. For example, adults can understand the majority of speech when the audibility of the speech is reduced to 40% or 20%, but young children need 80% or 60% audibility for the same level of understanding. Also, while adults need 4 to 6 bands of vocoded speech to reach good performance levels, children need 8–12 bands. While adults experience a release from masking when signals and background noise arise from different angles, children do not gain the same benefit, and in fact may experience masking from background noises coming from any direction. J.H. and J.S. from Acentech prepared auralizations that demonstrate these differences between children and adults. Those auralizations will be presented and discussed as possible educational tools.
High-performance acoustic ceilings make quiet classrooms quieter.
Authors: Lubman David , Sutherland Louis C.
Published: October 2009

Highly beneficial noise level reductions of 5–10 dB are reported in occupied classrooms equipped with highly sound absorbing ceilings. The incremental cost for such ceilings is nominal. These “Lombard effect” benefits apply to small classrooms with very low reverberation times (0.5 s or less) which is less than the 0.6-s maximum specified in ANSI S12.60-2002. The amount of noise reduction depends on teaching/learning style. In the typical noise reduction lecture classrooms (single talker), the typical noise reduction benefit is 5 dB in lecture classrooms (single talker) and group learning classrooms (multiple talkers). Unfortunately, these benefits require unoccupied background noise levels (BNLs) of about 35 dBA. They are not expected in typical noisy American classrooms (BNL ∼>45 dBA). This underscores the importance of compliance with ANSI’s 35-dBA BNL limit rather than the lenient and unsupported 45-dBA limit permitted in recent LEED and California CHPS guidelines. Details are reported in the outstanding study “Acoustic Ergonomics of School” by Oberdorster and Tiesler [University of Bremen, Germany (2006)]. Similar benefits were reported earlier in a case study by Sutherland [“The role of soundscape in children's learning,” J. Acoust. Soc. Am. 112, 2412–2413 (2002)].
Time domain geoacoustic inversion using back-propagation on an L-shaped array.
Authors: Park Cheolsoo , Gerstoft Peter , Seong Woojae , Hodgkiss William S.
Published: October 2009

This paper presents inversion of Shallow Water 06 experimental data measured on an L-shaped array (SWAMI 32). The array was deployed in 70-m water depth and consists of an equally spaced 10-element vertical line array (VLA) and a 256-m-long bottom moored 20-element horizontal line array (HLA). A mid-frequency (1100–2900-Hz) chirp source was towed at 35-m depth along a circular track around the VLA with radius 190 m. Time domain inversions using VLA, HLA, and both arrays were carried out and results compared. For the inversions, a multi-step optimization scheme is applied to the data using very fast simulated reannealing. The objective function is defined by the power of the back-propagated signal from the array to the source. At each step, water column sound speed profile, experimental geometry, and geoacoustic parameters are inverted successively. Accurate HLA positions were essential for the HLA and the HLA+VLA inversions. Finally, the inversion results were compared with other results near the site. [Work supported by ONR.]
Ocean tomography with acoustic daylight: A case study.
Authors: Godin Oleg A. , Zabotin Nikolay A. , Goncharov Valery V.
Published: October 2009

Ambient and shipping noise in the ocean provides acoustic illumination, which can be used, akin to daylight in the atmosphere, to visualize objects and characterize the environment [Buckingham et al., Nature (London) 356, 327–329 (1992)]. It has been shown theoretically [O. A. Godin, Phys. Rev. Lett. 97, 054301 (2006)] that, under rather general conditions, deterministic travel times between any two points in an inhomogeneous, moving or motionless, time-independent medium can be retrieved from the cross-correlation function of non-diffused acoustic noise recorded at the two points, without a detailed knowledge of the noise field’s sources or properties. Using the data obtained during the 1998–1999 Billboard Array Experiment [Worcester et al., J. Acoust. Soc. Am. 117, 1499–1510 (2005)], this paper demonstrates the feasibility of a tomographic reconstruction of the sound speed field from cross-correlation of acoustic noise recorded on a pair of vertical line arrays (VLAs) in deep water. Limitations of the noise data inversion associated with the ocean temporal variability, the VLA horizontal separation, recording bandwidth, and the noise directionality are analyzed. Prospects of long-range water column tomography with acoustic daylight are discussed. [Work supported by ONR.]
Two-point coherence of acoustic noise recorded by the North Pacific Acoustic Laboratory billboard array.
Authors: Godin Oleg A. , Zabotin Nikolay A.
Published: October 2009

Acoustic noise in the ocean contains extensive information about the noise sources and the propagation environment. In particular, two-point correlation function of noise (NCCF) is known to have peaks which correspond to acoustic travel times between the two points provided the noise field is sufficiently diffuse. In this paper, measurements of acoustic noise performed during the 1998–1999 Billboard Array Experiment [Baggeroer et al., J. Acoust. Soc. Am. 117, 1643–1665 (2005)] are re-examined with the goal of extracting environmental information. NCCF is evaluated by averaging time series of noise recorded on various vertical line arrays that comprise the Billboard Array. For any two hydrophones, NCCF is found to have a number of robust peaks. Possible generation and propagation mechanism responsible for various features of the NCCF are discussed. Statistical distributions of noise are utilized to differentiate between the NCCF peaks of different origins and to identify those peaks, which can be used to retrieve information about the sound speed without detailed knowledge of the noise sources. [Work supported by ONR.]
Estimating bubble density from attenuation measurements through an underwater explosion.
Authors: Holt Fred D. , Culver R. Lee
Published: October 2009

Underwater explosions have been studied intensively in the United States since 1941 [e.g., Cole (1945)]. Research to date primarily focuses on the initial shock and subsequent pressure waves caused by the oscillations of a “gas globe” that is the result of a charge detonation. These phenomena have relatively short timescales (typically less than 2 s). However, as the gas globe rises in the water column and breaks the surface, it leaves behind a residual bubble cloud which has been markedly less studied. A recent experiment measured the spatial and time-dependent acoustic response of the bubble cloud resulting from a charge detonated at 50-ft depth. A directional projector was used to propagate a linear FM (5–65-kHz) pulse through the bubble cloud to an array of hydrophones placed on the opposite side of the charge in order to measure attenuation. This talk will focus on the methods used to estimate bubble density size spectra from the attenuation measurements, those of Commander and McDonald (1991), Caruthers (1999), and Czerski (2009). [Work sponsored by the Office of Naval Research, Code 333.]
Sediment shear as a perturbation in geoacoustic inversions and an explanation of the anomalous frequency dependence of the attenuation.
Authors: Pierce Allan D. , Carey William M.
Published: October 2009

The depth dependent shear wave speed in marine sediments is much less than both the compressional and water column sound speeds. The neglect of shear in geoacoustic inversions is usually justifiable, but at frequencies less than 300 Hz, the loss of acoustic energy from the water column because of nonreturning radiation of shear waves into the bottom dominates the loss due to the intrinsic attenuation of the sediment. To account for this in a simple manner, a perturbation theory that takes advantage of the small shear speed has been devised. The canonical problem addressed takes the disturbance as having a horizontal trace velocity ω/k in the x direction, this being somewhat less than the compressional speed in the bottom but substantially larger than the depth-dependent shear wave speed. To lowest order the stress components σzz and σxx satisfy the same reduced Helmholtz–Bergmann equation as if the bottom were an inhomogeneous fluid. The shear stress σxz satisfies an approximately uncoupled equation governing a shear wave propagating vertically downward. Coupling occurs at the interface, and the power carried off by the shear wave can be approximately determined. Results succinctly and quantitatively explain why geoacoustic inversions yield attenuation frequency exponents less than 2.
Analysis models for the Underwater Sound Reference Division low-frequency acoustic calibration systems.
Authors: Huang Dehua , Paolero Anthony
Published: October 2009

The state of an art for calibrating acoustic transducers at very low frequencies is by way of a confined and well understood environment. The Underwater Sound Reference Division (USRD) has three such calibration systems, called systems K, J, and L, respectively. Each system has a cylindrical tube, of certain length and diameter, that determine a cutoff calibration frequency. System K operates in a standing wave mode condition. Systems J and L both operate in traveling wave mode conditions, where plane waves propagate from one end of the tube to the other. Optimally locating a calibrated transducer and an unknown within the tube provides the proper configuration for calibration. This paper demonstrates the simulation tools to predict performance of aforementioned low-frequency calibration systems. The mathematical model, the GUI coding, simulation, predicted, and test results will be presented.
Electroacoustic transducer and array modeling tools.
Authors: Kuntsal Ender
Published: October 2009

The availability and the accuracy of transducer and array design tools are becoming even more important with the changing economical climate. The engineers would like to evaluate their designs more quickly in place of the prototyping, which requires additional manufacturing and testing time and expense. The software models must be user friendly, provide results that can be relied on, and also be affordable. The strength of a transducer and array design engineering team is still extensively based on experience and background, but these tools make the process more efficient. There are many transducer and design software programs used in industry and academia. Some are based on analytical solutions and some use more complicated numerical methods such as finite element models, which have the capability of modeling piezoelectric materials and acoustical parameters. Among these are ATILA, PAFEC, MAVART, NASTRAN, PZFlex, and other powerful programs such as CHIEF and TRN. Some of these programs can be used to design both resonators and arrays while taking baffling and/or acoustic interaction effects into account. In this presentation, a summary of commonly available software packages and their features will be described.
Broadband characteristics of piezoelectric transducer bonded to a thick plate resonator.
Authors: Akiyama Iwaki , Yoshizumi Natsuki , Saito Shigemi , Ohira Katsumi , Takahashi Osamu , Nakamura Kentaro
Published: October 2009

The piezoelectric transducer bonded to a thick plate resonator wholly vibrates as a single transducer. Since the distribution of stress in the thickness direction is asymmetric, the transducer resonates at even order harmonic frequencies as well as odd order ones. The back of the resonator is bonded to a quarter-wavelength matching layer to transmit backward the fundamental frequency wave through the layer. The front of piezoelectric transducer is bonded to another matching layer to transmit forward the ultrasonic wave of higher order frequencies. If both the piezoelectric transducer and the resonator are made of low Q materials, the admittance curve including troughs among resonance peaks is flattened and shows broadband characteristics. Such broadband characteristic transducers made of one to three composite piezoelectric materials were experimentally studied. The thickness of the piezoelectric transducer and the resonator were designed for resonance frequencies of 7 and 1 MHz, respectively. The back of matching layer was bonded to the absorbing material not to reflect forward the ultrasonic waves of 1 MHz. As a result, the ultrasound of center frequency of 5 MHz and fractional bandwidth of 100% were transmitted from the transducer driven by an impulsive signal.
A radial propagator for computing axisymmetric pressure fields using the angular spectrum method.
Authors: Pees Edward H.
Published: October 2009

The notion of a propagator is central to the angular spectrum of plane wave formulation of diffraction theory, which expresses the pressure field diffracted by a two-dimensional aperture as a superposition of a continuum of plane waves. In the conventional form, an exponential term, known as a propagator, is multiplied by the wavenumber spectrum obtained from a two-dimensional spatial Fourier transform of the aperture boundary condition, to obtain the wavenumber spectrum in a plane parallel to the boundary, offset by some distance specified in the propagator. By repeated use of this propagator and Fourier inversion, one can completely reconstruct the homogeneous part of the pressure field beyond the aperture boundary. In this presentation, we draw upon earlier work relating the boundary condition to the axial pressure and show that when the aperture is axially symmetric, an alternative type of propagator can be derived that propagates an axial wavenumber spectrum away from the axis of the aperture. Use of this radial propagator can be computationally advantageous since it allows for field reconstruction using one-dimensional Fourier transforms instead of Hankel transforms or two-dimensional Fourier transforms.
Microphone array techniques using cross-correlations.
Authors: Rhudy Matthew B. , Bucci Brian A.
Published: October 2009

Civilian noise complaints and damage claims have created a need to establish a detailed record of impulse noise generated at military training facilities. Wind noise is causing false positive impulse detections in the current noise monitoring systems. Multiple channel data methods were investigated in order to distinguish the characteristics of noise events. A microphone array was used to collect four simultaneous channels of military impulse and wind noise data. Cross-correlation functions were then used to characterize the input waveforms. Three different analyses of microphone array data were developed. A new value, the min peak correlation coefficient, is defined from the minimum value of peaks of the cross-correlation coefficient functions among the different channels. This value is a measure of the likelihood that a given waveform originated from a correlated noise source. The angle of incidence of the noise source is calculated using a sound source localization technique based on the geometry of the array. A weighted averaging method was also developed to synthesize multiple channels of data into one single channel. This method preserves the correlated part of the overall signal, while reducing the effects of uncorrelated noise, such as wind.
Blown-closed free reeds with and without pipe resonators.
Authors: Cottingham James P.
Published: October 2009

The Asian mouth-blown free reed instruments are of ancient origin and use a symmetric free reed coupled to a pipe resonator. The reed behaves as a blown-open or outward striking reed, with playing frequency below both the resonant frequency of the pipe and the natural frequency of the reed. Although these instruments were known in Europe when the Western free reed family originated about 200 years ago, it does not appear that the mechanism used in the Western instruments was copied from them. In Western free reed instruments the reed tongue is offset from the opening in the frame, permitting operation on only one direction of air flow. Pipe resonators are not required and generally not used. If one of these reeds is coupled with a pipe, the sounding frequency can, within certain limits, be pulled to match the pipe frequency, and it behaves as a blown-closed or inward striking reed, with playing frequency below both the resonant frequency of the pipe and the natural frequency of the reed. This paper summarizes recent experimental research on the blown-closed free reed, with or without coupling to a pipe resonator, with emphasis on significant differences between the two situations.
Free reeds coupling with either a vocal tract or rather small pipe.
Authors: Millot Laurent
Published: October 2009

Free reed model used is based on a precise description of flow passing throw, the section defined between the shallot and the bended reed, and on the modelling of reed as a clamped-free cantilever beam. Classical description of loading pipe is based on wave reflection function, not designed to be excited by a flow but backward- and forward-traveling plane waves, so the free reed coupling with such a pipe model is physically irrelevant. Then, it is shown that an acoustical flow description is needed, and simple, for at least rather short pipe, giving more realistic sound simulations. It is then explained why a vocal tract description, physically relevant, is possible using only acoustical flow modeling for each element constituting the vocal tract model. In fact, flow acoustical modeling for elements is linked to electroacoustical analogies derived not as a low-pass approximations of input impedance but using the approximation of flow behavior for each element, valid for the whole audio range rather than only the low-pass register. Theoretical explanations should be given to explain why wave paradigma may be irrelevant and sound numerical simulations would be given for each coupling situation considered.
Nonlinear acoustic wave propagation in inhomogeneous moving media.
Authors: Blanc-Benon Philippe , Averiyanov Mikhail V. , Cleveland Robin O. , Khokhlova Vera A.
Published: October 2009

Extensive theoretical analysis, numerical studies, and both large-scale and laboratory-scale experiments have been dedicated to the problem of shock wave propagation in air during recent years. The current interest is motivated by supersonic civil transport which is necessarily affected by problems of sonic boom propagation in the atmosphere. The high-amplitude shock wave generated by a supersonic aircraft propagates through the atmosphere toward the ground and generates an acoustic field with non-uniform pressure distribution. Temporal characteristics and spatial structure of the sonic boom are influenced by aircraft trajectory, nonlinear effects, and diffraction and scattering by inhomogeneities. We review recent results from various teams based on a generalized KZK-type equation that includes the effects of a moving inhomogeneous media. Statistical analysis of the numerical solutions is performed, and the results are compared to experimental data obtained in the controlled laboratory-scale experiments conducted in the Ecole Centrale de Lyon anechoic wind tunnel.
Shocking stones with the Khokhlov–Zabolotskaya–Kuznetsov equation.
Authors: Cleveland Robin O.
Published: October 2009

In shock wave lithotripsy (SWL) high-amplitude acoustic waves generated outside the body are focused onto kidney stones in order to fragment them into pieces that are small enough to pass naturally. In order to investigate mechanisms of fragmentation and collateral damage to the soft tissue it is necessary to understand the acoustic field delivered to the tissue in and around the kidney. This is not easily accomplished experimentally and motivates the development of numerical models. The application of the Khokhlov–Zabolotskaya–Kuznetsov (KZK) equation to the SWL problem is discussed as there are two underlying assumptions that are challenged: the peak pressures are on the order of 50 MPa resulting in an appreciable acoustic Mach number of 0.02, and large focusing gains are employed which violate the paraxial approximation. Predictions using the KZK equation, with a layered model of the tissue path to the kidney, demonstrate that the waveform shape, in particular, the risetime, is strongly affected by the tissue path. However, once the waveform enters the urine in the collecting space of the kidney, the waveform heals and a sharp shock results within about 5 mm of propagation. [Work supported in part by NIH DK-43881.]
Signal-to-noise-ratio loss in hearing-impairment and feature enhancement.
Authors: Han Woojae , Singh Riya , Allen Jont
Published: October 2009

The primary purpose of this study is to investigate detailed consequences of hearing loss and to scrutinize if individual hearing impaired (HI) listeners exhibit individual patterns in their consonant confusion scores. 16 English nonsense CV syllables, by 18 various talkers in five speech-weighted noise (−12, −6, 0, +6, and +12-dB signal-to-noise-ratios) and quiet conditions, were randomly presented to 11 HI subjects (18 HI ears) having sensorineural hearing losses at two different amplified levels: with and without NAL-R (similar to fitting hearing aids). The HI ears have sensorineural hearing loss with flat (mild-to-moderate-to-severe), ski-slop at low- and/or high-frequency regions, and cookie-bite configurations. To characterize the consonant loss, the performance of each HI ear was compared to average normal-hearing listeners in masking noise. Preliminary analysis shows that each HI ear has an individual profile, which is a characteristic of the impaired ear. Once the noise-sensitive consonants are identified, the aim is to use signal processing techniques to selectively enhance features to make them audible in noise. [Work supported by the NIH Grant RDC009277A.]
Comodulation masking release to detect cochlear dead regions in hearing impaired ears.
Authors: Singh Riya , Han Woojae , Allen Jont B.
Published: October 2009

The purpose of this study is to investigate if comodulation masking release experiments with hearing impaired (HI) individuals can be used as a reliable method to detect dead regions in the cochlea, if any. The experiment involves detection of a pure tone in a narrow band of noise (the on-signal band) in presence of four other narrow bands of noise that fall outside the auditory bandwidth at the center frequency of the on-signal band. These flanking bands are either all co-modulated so as to have the same amplitude envelope as the on-signal band or are randomly modulated and are presented monaurally to an impaired ear. Each band is level adjusted to be audible to the HI ear. Data are being collected for both these cases from normal-hearing listeners and HI listeners with no dead regions. Every HI individual also undergoes psychophysical tuning curve and threshold-equalizing noise tests which are currently used tests to detect cochlear dead regions. Each HI individual also takes speech perception tests with CV syllables under various noise conditions. The consonant loss profiles from the perception experiments help make a reasonable estimate of the possibility of a dead region. [Work supported by NIH.]
Enduring non-verbal memory for spectral timbre.
Authors: McKeown Denis , Mercer Tom
Published: October 2009

It is often assumed that auditory sensory memory disappears within a few seconds or is so smudged as not to permit fine discriminations between successive sounds beyond a few seconds. But studies have been confounded by proactive interference, task difficulty, and verbal labeling. In Experiment 1, participants heard two complex tones separated by a silent retention interval of 2, 4, 8, 16, or 32 s. Tones varied widely in pitch from trial to trial (so no standard memory could be formed) and were non-speech-like (difficult to verbalize), and very long inter-trial intervals were used (32 s) to reduce interference from prior sounds. After the tone-pair presentation, listeners made a same-different response based on slight differences in spectral timbre. Discrimination performance expressed as d′, at each retention interval revealed that the persistence of memory for spectral timbre surpassed the usually quoted lifetime of some few seconds and for certain participants it endured for 32 s. In Experiment 2, overall stimulus levels were roved within trials to rule out loudness cues in the discrimination. It is proposed that reasonably robust non-verbal memory traces for spectral timbre persist for several seconds allowing listeners to make fine discriminations of changes to a spectrum.