bluuebunny/crossref_metadata_2025 huggingface dataset:
Female voice age classification based on acoustic characteristics of speech.
Authors: Lim Eun-Min , Choi Jeung-Yoon
Published: July 2011

As there are many changes in vocal apparatus caused by aging, many acoustic differences also can be found in speech. These differences show different tendencies for male and female speakers and are more pronounced for females. Therefore, this paper focuses on female speaker age analysis and classification. To analyze acoustic differences, global measurements related to the vocal source such as band energy, jitter, shimmer, open quotient, and spectral tilt are used. Additionally, local features that are extracted near vowel landmark regions are also used. Appropriate age boundaries are selected based on the results of analysis, and feature selection is carried out according to the results of analysis of variance (ANOVA) test for classification. The proposed age classification system is modeled using Gaussian mixture models (GMMs), and the sequential feature selection method is used. Experiments are performed on the aGender corpus provided in the INTERSPEECH 2010 Paralinguistic Challenge.
Dependence of long-term-average spectrum curves of voices as function of equivalent sound level.
Authors: Sundberg Johan , Granqvist Svante
Published: July 2011

Long-term-average spectrum (LTAS) is a quick and simple analysis method, which is frequently used in voice analysis. It typically reaches a stable curve after about 30 or 40 s of speech or singing and reflects both voice source and formant frequency characteristics of a voice. However, it is quite sensitive to changes of vocal loudness. In two previous investigations a set of recordings were analyzed in which 15 male and 16 female adults read a standard text aloud at seven different degrees of vocal loudness. The results showed that the LTAS curve changed linearly but in a frequency dependent manner with the equivalent sound level Leq. Furthermore, the alpha ratio between the summed energy below and above 1 kHz was found to be a quadratic function of Leq. In the present investigation we study the relation between Leq and (1) the overall LTAS slope between 0.75 and 4 kHz and (2) the average increase in LTAS level, in dB, in the same range. While slope varied considerably between speakers, average LTAS level increase could be accurately described by a linear function of Leq which is similar between speakers. Possible applications of these findings will be discussed.
Acoustic and aerodynamic analyses of transgender voice.
Authors: Hancock Adrienne B. , Harrington Rachael M. , Mahshie James , Lennon Alicia
Published: July 2011

Acoustic and aerodynamic data are needed to describe how the vocal mechanism of a transgender person operates to achieve desired gender perception. Early evidence-based practice for transgender voice therapy was limited to changing vocal pitch—at first aiming for the normative values of the desired gender and more recently aiming for more feasible targets of pitch within a “gender-neutral range” established by perceptual research. 20 years later, research is discovering that pitch may be a major influence in gender perception, but is almost certainly not the only influence. In this presentation, the physiology of 20 voices (male-to-female and female-to male) will be illuminated by reporting the glottal closure patterns (e.g., electroglottography), patterns of airflow through the vocal mechanism (e.g., subglottal pressure estimated from intraoral pressure and derived glottal airflow), and resonant qualities of the vocal tract (e.g., formants). Preliminary data suggest that transgender values are not equivalent to a simple average of female and male normative data. Once we learn how the transgender vocal tract can safely overcome constraints of biological anatomy and which parameters contribute most to listeners’ perceptions of gender, speech-language pathologists can better guide their clients to speak in an authentic voice.
Three-dimensional whole-field measurements of pulsatile glottal jets using synthetic aperture particle image velocimetry.
Authors: Daily David J. , Truscott Tadd , Thomson Scott L
Published: July 2011

The use of synthetic aperture particle image velocimetry (SAPIV) to measure the three-dimensional (3-D) jet velocity field generated by the human vocal folds during speech is described. SAPIV uses an array of cameras to acquire 3D flow velocity data throughout a volume of interest. In the present application, eight high-speed cameras are synchronized with a volume-illuminating pulsed laser to image the supraglottal jet exiting from self-oscillating synthetic vocal fold models. The images from the eight cameras are transformed and reconstructed to generate different image planes within the flow field, to which traditional particle image velocimetry (PIV) techniques are applied to calculate 3-D velocity vectors throughout the flow field. Two-dimensional PIV using thin laser sheets has been previously used to study the supraglottal jet, but 3-D measurements have been typically limited to either (1) quasisteady measurements as the laser sheet traverses the glottal jet volume, or (2) 3-D measurements over a narrow subvolume of the glottal jet. SAPIV enables 3-D data to be acquired simultaneously throughout the flow field and over a volume enclosing a large region of the jet. The SAPIV implementation is explained and 3-D data of the glottal jet are presented.
Effects of left-right asymmetry in vocal fold geometry and stiffness on phonation onset in a physical model.
Authors: Mendelsohn Abie H. , Zhang Zhaoyan
Published: July 2011

Vocal fold paralysis or paresis often leads to left-right asymmetry in vocal fold geometry and stiffness. An improved understanding of the influence of such left-right asymmetry on phonation would allow clinicians to better diagnose and treat these voice disorders. In this study, the effects of such left-right asymmetry on phonation were investigated using a two-layer physical vocal fold model. Specific vocal fold parameters investigated included body layer stiffness, cover layer stiffness, cover layer thickness, and cover layer depth. Results showed that a unilateral decrease in vocal fold body layer or cover layer stiffness reduced the phonation threshold pressure and onset frequency to a value that was closer to that of the less stiff fold in symmetric conditions. Unilateral thinning of the cover layer depth caused an increase in phonation threshold pressure and onset frequency, whereas unilateral shortening of cover layer thickness led to a threshold pressure close to the averaged threshold pressures of the two unique geometries in symmetric conditions. The clinical implications of the results were also discussed. [Work supported by the NIH.]
Matched signals: The beginnings of time reversal.
Authors: Clay Clarence S. , Anderson Brian E.
Published: July 2011

This paper discusses the original time reversal experiments as conducted by Parvulescu and Clay. The idea of conducting the first time reversal experiment came about from measurements of the reproducibility of sound transmissions in the ocean in the late 1950s and early 1960s. Then it was believed by many that long range acoustic signal transmissions in the ocean were not reproducible or stable. During a coffee hour in 1960, Parvulescu proposed a simple experiment to create an acoustic matched signal through the use of time reversal, in their coffee room/laboratory area. This represents the first known record of a time reversal experiment. This paper will then discuss a series of experiments that were conducted underwater to communicate between two ships through the use of this matched signal technique. Details are in the theoretical-experimental monograph of Tolstoy and Clay [Ocean Acoustics (McGraw-Hill, New York, (1966); J. Acoust. Soc. Am. (1987)]. This paper calls attention to these early experiments and theory as the first demonstrations of the now popular technique termed time reversal.
Toward a high power non-contact acoustic source using time reversal.
Authors: Ulrich T. J. , Le Bas Pierre-Yves , Anderson Brian , Esplin J. James
Published: July 2011

Over the last decades, nonlinear acoustic techniques have been developed to detect mechanical damage in solids. They have been proven to be far more sensitive to early damage than standard linear acoustic techniques such as C-scans, time of flight, etc. Unfortunately, nonlinear techniques often require high amplitude waves to propagate within the sample. To be practical in industrial applications, signals have to be generated without contact. Currently available non-contact transducers are usually not powerful enough to apply nonlinear techniques. Well describe a first step toward the creation of a high amplitude non-contact acoustic source. This source is based on the principle of focusing energy on the surface of a sample using time reversal in a hollow cavity. By using a laser vibrometer for the necessary calibration of the system, we are able to use a full non-contact process. This source is proven to be much more energetic than current off the shelf non-contact transducers. It is a broad band source with an adjustable standoff distance. This has the potential to open new perspectives for acoustic processes that require high amplitude, including nonlinear techniques for non-destructive evaluation.
Localization of a nonlinear source in the bulk of a solid.
Authors: Le Bas Pierre-Yves , Ulrich TJ , Johnson Paul , Anderson Brian
Published: July 2011

Time reversal (TR) has the potential to become a powerful tool in non-destructive evaluation. Coupled with nonlinear properties of cracks in a group of techniques known as Time reverse nonlinear elastic wave spectroscopy, it provides the means to detect and image mechanical damage in complex solids. In this prototype experimental study, we show that we can excite a buried nonlinear feature applying TR. The feature scatters energy that is detected on the sample perimeter. The time signals are filtered about the nonlinear-generated components and are broadcast back, focusing on their source, the buried nonlinear feature. The current challenge is introducing sufficient energy in order to excite the buried feature and produce nonlinear scattering that can be detected on the edges of the sample. This was achieved using a 30 channel system. Two features were localized: a part of the interface between two blocks submitted to a high amplitude signal, and a defect on the same interface (a cured drop of glue creating hard contact between the two solids). Results are visualized using the energy flux quantity.
Ultrasonic defect localization in pipes using time reversal.
Authors: O’Donoughue Nicholas A. , Harley Joel , Moura José M. F. , Shi Jun
Published: July 2011

Time reversal mirrors have been shown to effectively focus acoustic waves for structural health monitoring in thin plates. This approach achieves a high degree of accuracy, despite the complex propagation environment inherent to thin plates. We consider ultrasonic defect localization via time reversal in thin-walled pipes. The different boundary conditions in pipes give rise to new wave modes and dispersive effects. We use an array of six transducers arranged around the circumference of a 10-cm-diameter steel pipe to excite guided ultrasonic wave modes, and apply iterative time reversal to focus these waves on a defect. We apply maximum likelihood to detect the presence of a defect, and then correlate the response from the defect with a library of known signatures to locate the defect. Results are provided from a simulation based on experimental data collected from a stainless steel laboratory pipe specimen 11.5 cm in diameter and 1.5 m in length. We simulated our defect with a grease-coupled commercial transducer and measured waves from two arrays of six transducers affixed to the pipe. Our initial excitation is a wideband sinc with 400 kHz bandwidth.
Compressing the computational burden of matched-field processing for sound source localization.
Authors: Sabra Karim G. , Mantzel William , Romberg Justin
Published: July 2011

Sound source localization in shallow water environment is done using matched field processing (MFP), also referred to as time-reversal imaging or backprojection method. Standard MFP is usually implemented by matching (using a cross-correlation operation) the received acoustic data from the sought source with modeled data (or replicas) for point source located at multiple test locations over the a priori search area. Consequently, a direct implementation of MFP (i.e., brute force search) over a large search area can be highly computationally demanding especially when attempting to locate repeatedly several stationary or moving unknown sources in a complex environment. We formulated instead a compressive MFP approach allowing for significant computation time savings with a computational cost that scales with the logarithm of the number of independent-or uncorrelated- test locations. This approach leverages the key concepts behind group testing and compressed sensing by computing instead the expected value of multiple ambiguity surface realizations obtained by backprojection of orthogonal random signals from the receiver locations. Thus compressive MFP allows to estimate the actual ambiguity function using few replica computations by evaluating hypothetical random groups of test locations instead of individual test locations (as done in conventional MFP).
Experimental and numerical investigations on decay parameter distributions in coupled-volume systems.
Authors: Ostrowski Angela , Escolano Jose , Xiang Ning , Braasch Jonas
Published: July 2011

The current interest of couple-volume systems in performing art spaces has motivated research to better understand and predict multiple-sloped sound energy decays. This work investigates sound energy decay characteristics of coupled spaces using scale model measurements and diffusion-equation modeling results. The experimental and numerical study is aimed to reveal possible ranges of decay parameters when investigating source and receiver positions in the primary room and aperture sizes. The receiver positions over large areas in the primary room are investigated to identify the presence of a spatial dependence and variations of decay parameters on the source position as it changes with respect to the aperture. Both scale model measurements and diffusion-equation model results are analyzed using Bayesian probability inference in order to estimate decay orders (number of multiple slopes) and to quantify multiple-slope decay parameters including individual decay levels, level differences, and decay times of individual slopes. This paper will also discuss dependence on the upper limit of backward integration on signal-to-noise ratios achieved from experimental measurements.
Using a spherical microphone array to analyze stage acoustics.
Authors: Guthrie Anne E. , Caulkins Terence J. , Clapp Samuel , Braasch Jonas
Published: July 2011

A spherical microphone array, capable of beamforming, has been constructed to measure sound fields using second-order ambisonics. The design of the microphone is based on the work by Duraiswami et al. using equations for scattering off a rigid sphere. The 16-channel array was used to measure nine different concert-hall stages around the state of New York, with multiple positions recorded to analyze their acoustical behavior. Additional measurements with an omnidirectional microphone were obtained to determine A.C. Gade’s stage support parameter for comparison purposes. Based on the measurements, alternative methods will be discussed to obtain better descriptors to quantify stage acoustics based on the additional spatial information the spherical microphone provides. Additionally, the work by Dammerud in developing geometric parameters based on comparisons between omnidirectional measurements and architectural dimensions will be utilized. A comparison between parameters by Dammerud and information extracted from a beamforming analysis of the ambisonic stage measurements will be shown. In particular, the comparison will examine the spatial response over time to understand how this is affected by the dimensions of the stage.
Explorations of room acoustics using a second-order ambisonic microphone.
Authors: Clapp Sam , Guthrie Anne , Braasch Jonas , Xiang Ning
Published: July 2011

Most room acoustic parameters are calculated with data from omni-directional or figure-of-eight microphones. Using an ambisonic microphone to record room impulse responses can open up several new areas of inquiry. It can yield much more information about the spatial characteristics of the sound field at the points of interest, including the diffuseness of the sound field and the directions of individual reflections. In this research, a 16-channel, second-order ambisonic microphone is designed, built, and tested with both simulations and simple, controlled sound events. Room impulse responses are then measured for a wide variety of different concert halls located throughout New York state. The results are analyzed using beamforming techniques to determine spatial information about the sound field. This method has allowed for determining the directions of individual early reflections and to characterize the spatial evolution of the sound field over the course of the impulse response. This new dimension of data can also help in understanding the differences between halls that may yield similar values in traditional metrics (such as reverberation time and clarity), but which are judged very differently by listeners.
Just-noticeable-difference of clarity index (C80) using real-time switching between test signals.
Authors: Wells Adam P. , Ormsbee Caitlin I. , Celmer Robert D. , Vigeant Michelle C.
Published: July 2011

The purpose of this study was to investigate the just noticeable different (JND), or the smallest detectable difference, for the clarity index for music (C80). This project is the third in a series of investigations on this topic. The first study showed that C80 JND is not close to 1 dB, as work from other researchers had previously found. The second study examined the effect of two different test methods. Test-method1 required the subjects to listen to all of signal A and all of signal B, while for Test-method2 subjects could switch in real-time between the two signals. The results of the second study, which were implemented in this current study, suggested adding an extensive training period with Test-method1 before using test-method2 for the actual test. Two C80 base-cases were used, with base-case1 having C80 at a 1 kHz set to −3 dB with a reverberation time (T30) of 1.9 s and base-case2 with a C80 of +1 dB with T30 of 1.5 s. Highly trained musicians were used as test subjects. The C80 JNDs for each base case, along with a comparison to the results from the first two studies, will be discussed. [Work is supported by the Paul S. Veneklasen Research Foundation.]
Rating of the loudest college basketball arenas for Entertainment Sports Network (ESPN) magazine.
Authors: Shepherd Micah R. , Hambric Stephen A. , Evans Neal U. , Domme Dan J. , Christian Andrew W. , Poulain Kieran , Gardner Michael D. , Orr Andrew J. , Cranage Bryan P.
Published: July 2011

A recent ESPN magazine article [“These go to 11,” ESPN the Magazine 15 Nov (2010)] ranked the top collegiate basketball arenas according to “noise potential.” The top five arenas were listed as Kansas University, Duke University, New Mexico University, Kentucky University, and Florida University. The rankings were established by a team of Penn State Acoustics students using the theory for sound buildup in large rooms, since actual measurements were infeasible. Both diffuse field and direct field contributions of the sound pressure were estimated at center court for octave band frequencies from 125 Hz to 4 kHz. Seating geometries, materials, and other relevant information were collected for each arena and used with estimated absorption coefficients to determine the room constant and critical distance. The diffuse field contributions were then combined with approximate sources terms based on the seating capacity of the arena, the proximity of the fans to the court, and whether they were students. The sound pressures were then combined into a total A-weighted sound pressure level and used to determine the ranking. The direct and reverberant contributions of each arena were then compared to establish the positive and negative aspects of each arena in terms of noise potential. This comparison reveals how Kansas and Duke reached the top of the ranking, despite having drastically different arena geometry and capacity.
Are Blainville’s beaked whales echo-locating without a clock?
Authors: Fischer Mark
Published: July 2011

Recent work using wavelets in the service of detection and classification of Ziphiid clicks indicates that a novel method of echo-location exists in the clicks of Blainville’s beaked whales /Mesoplodon densirostris/. Results will be presented that support the hypothesis that, instead of timing, they rely on the creation of a unique waveform that allows them to perceive round-trip distance based on changes in the wavelet power spectrum.
Real-time detection and tracking of beaked whales using a towed hydrophone array.
Authors: Yack Tina M. , Barlow Jay , Calambokidis John
Published: July 2011

Beaked whales spend the majority of their time at depth and are inconspicuous when they surface. Therefore, they are difficult to detect using only standard visual survey methods. However, recent advancements in acoustic detection have made passive acoustic monitoring from a towed array of hydrophones a viable alternative to visual survey methods for beaked whales. Beaked whales can be discriminated from other cetaceans by the unique characteristics of their echolocation clicks (duration >175 μs, center frequencies between 30 and 40 kHz, inter-click intervals between 0.2 and 0.4 s and frequency upsweeps). These unique characteristics make it possible to use passive acoustic monitoring (PAM) to detect and track beaked whales in real-time. In 2009 and 2010, we used PAM to identify areas of beaked whale habitat use in the Southern California Bight (SCB). This area is characterized by deep basins separated by islands and shallow ridges. We developed effective methods for detecting and tracking beaked whales in real-time and identified previously unknown areas of beaked whale habitat use in the SCB. This work will allow for beaked whale habitat relationships to be explored on a relatively small scale and will contribute to efforts for effective management and conservation of beaked whales in the SCB.
Complementary techniques for robust acoustic identification of individual sperm and beaked whales.
Authors: Ioup George E. , Ioup Juliette W. , Tiemann Christopher O. , Sidorovskaia Natalia A.
Published: July 2011

Properties of individual clicks and characteristics of click trains provide clues for the passive acoustic identification of individual sperm whales via several complementary techniques. For example, recordings of individual clicks can be clustered according to their shape or spectrum, or entire trains of clicks can be exploited by monitoring the slow evolution of click shapes over time. Cadence analysis studies differences in click-train rhythms of simultaneously clicking whales. Localization uses bearing, and if possible range, to differentiate individuals. Manual click-train identification employs both individual click properties and sequence analysis to identify individual whales. When combined, these methods can provide robust identification of individual sperm whales. The application of these techniques to beaked whales is difficult because, compared to sperm whales, beaked whales have lower source level, a narrower beam, and a faster rate of turning. Consequently, click trains on a single sensor have frequent drop-outs. The above techniques require more sophistication and benefit from closely spaced sensors to fill in drop-outs. Previous Littoral Acoustic Demonstration Center (LADC) multi-mooring Environmental Acoustic Recording System (EARS) data are analyzed for beaked whales and an experiment is proposed with greater mooring density to improve individual beaked whale identification. [Research supported by SPAWAR and ONR.]
Noise reduction for better detection of beaked whale clicks.
Authors: Lu Yang , Klinck Holger , Mellinger Dave
Published: July 2011

A least mean square (LMS)-based filter with a forgetting factor has been designed to improve the signal-to-noise ratio (SNR) of clicks recorded from Blainville’s beaked whales (Mesoplodon densirostris). A suboptimal step-size of the LMS filter is obtained through searching. The trained filter is proven to be environmentally robust; i.e., it can be used in beaked whale recordings from other areas, which is an advantage for applications such as autonomous gliders for which accurate click detection in untested locations is beneficial. It can also be used to improve the SNR of other marine mammal acoustic signals. A detector using filter output is designed to detect beaked whale calls. Through simulations, the proposed detector is shown to be capable of detecting most of the desired clicks, but is not able to differentiate other coexisting species such as Risso’s dolphins and pilot whales. By combining the proposed detector with the energy ratio mapping algorithm (ERMA) [Klinck and Mellinger (in preparation)], which measures energy differences between different species, higher detection accuracy for beaked whale clicks can be achieved.
Passive-acoustic monitoring of odontocetes using a Seaglider: First results of a field test in Hawaiian waters.
Authors: Klinck Holger , Mellinger David K. , Roch Marie A. , Klinck Karolin , Bogue Neil M. , Luby Jim C. , Jump William A. , Pyle John M. , Shilling Geoff B. , Litchendorf Trina , Wood Angela S.
Published: July 2011

In fall 2009 the University of Washington, Applied Physics Laboratory conducted in collaboration with the Oregon State University, a comprehensive field test of a passive-acoustic Seaglider along the western shelf-break of the island of Hawaii. During the 3 week mission, a total of approximately 170 h of broadband acoustic data [194 kHz sampling rate] were collected. The recordings were manually analyzed by an experienced analyst for beaked whale (Ziphiidae), dolphin (Delphinidae), and sperm whale (Physeter macrocephalus) echolocation clicks as well as echo sounder pings emitted by boats in the area. Here we present and discuss first results of these data analysis, which revealed that more than 50% of the recorded files (each of 1-minute duration) contain bioacoustic signals. Furthermore the recorded data and the results of the manual analysis are used to validate and optimize an automated classifier for odontocete echolocation clicks, which was developed in a collaborative effort with San Diego State University. The algorithm is intended to be implemented on the Seaglider to enable species identification by classifying detected echolocation clicks in (near) real-time during sea trials. [This work is funded by the Office of Naval Research and the U.S. Navy’s Environmental Readiness Division.]
Killer whale scouting: Listen live for troops J, K, and L.
Authors: Veirs Scott R. , Veirs Val R. , Wood Jason D.
Published: July 2011

The southern resident killer whales are highly vocal, sharing almost 40 distinct calls. The endangered population of about 87 orcas live in matriarchal groups (like troops) known as J, K, and L pods. Each pod has a favorite call that is used about half the time pod members vocalize. If you hear a favorite call you can guess (or infer) which pod is present. A new way to listen for orcas is with hydrophones (underwater microphones) that are connected to the internet. The Northeast Pacific Hydrophone Network—http://orcasound.net—includes five hydrophones located near Seattle, WA within the summertime range of the orcas. You can listen live through iTunes to discover which pods are calling or to monitor the environment for noise pollution that could be a problem for the whales. This hands-on activity teaches you how to listen, identify the three pods based on their favorite calls, and share your observations with other listeners. High school students can use free software to record orca calls, whistles, and clicks and analyze them, guided by a suite of citizen science projects.
Zombie classification demonstration.
Authors: Cooper Jennifer L.
Published: July 2011

Many tasks in the real world call for identifying or grouping things by the sounds they make. Some examples include identifying species of whale or even individual animals using recordings of their clicking sounds or finding the cause of an engine failure by the sound it makes. The same methods can be used to determine whether a sound is human or animal in nature or even human vs zombie. In this demonstration, students’ voices are recorded and subsequently classified as human or zombie using a neural network. Using a graph called a spectrogram, we can look at the spectral (time and pitch) features of each voice and discuss how those features were used to identify it as human or zombie.
Listen up and perceive speech: Demonstrating the flexibility of speech perception.
Authors: Miller Rachel M.
Published: July 2011

Individuals encounter many different types of speech information on a daily basis. Some of this information comes in forms that are not even auditory in nature (e.g., visual speech). Other types of information contain variations that should make them difficult to understand (e.g., speech-in-noise and accented speech). However, individuals are able to perceive all sorts of speech quite easily. The present demonstrations will show the flexibility of speech perception by providing girl scouts with examples of various types of speech. The demonstrations will be (1) a lipreading choice task that presents silent videos of talkers producing words, (2) silent videos of talkers producing sentences, (3) silent, point-light videos that show how speech can be perceived simply by seeing the movement of talkers‘ faces, (4) sentences produced in three levels of noise, and (5) words produced by native Chinese and Spanish talkers.
How musical instruments make sounds: An exploration of standing waves and resonance frequencies.
Authors: Neilsen Tracianne B. , Morgan Jessica
Published: July 2011

A basic understanding of the production of sound waves can be obtained by studying standing waves and resonance frequencies. A vibrating string settles into a steady, standing wave pattern when it is driven at one of its resonance frequencies. The resonance frequencies depend on the string’s length and the tension applied to the string. Similarly the resonance frequencies of tubes are determined by the tube’s length and whether the ends are open or closed. In a series of hands-on demos, we explore the factors that influence the creation of standing waves by exciting the resonances of strings, tubes, rods, a metal plate, slinky, and a wine glass. These simple models provide insight into how musical instruments produce sound.
Building a sound and breaking it down.
Authors: Ross Tetjana
Published: July 2011

The sounds we hear everyday are made up of acoustic waves of many different frequencies. When a sound is dominated by waves of a particular frequency, it is referred to as having a certain pitch. However, even single musical notes are often made up of various different frequencies (usually harmonics of the first). This is part of what gives different musical instruments their distinctive sounds. Our ears analyze sounds, with different nerve endings being activated by waves of different frequencies. This breaks sounds down so that we can distinguish different pitches or even chords. This breaking down of sound, showing its loudness as a function of frequency, is called finding its spectrum. In two hands-on experiments, we will use a computer to visualize the building and breaking down of sounds. First, we will use a keyboard as a wave generator to incrementally build up a sound from waves of different frequencies, all the while using a computer to show us the spectrum of the resultant sound. Second, we will record you singing an “ah” or “oo” sound and examine its spectrum to show all the frequencies involved.
Longitudinal waves demonstrations.
Authors: Swearingen Michelle E.
Published: July 2011

In a wave, something always moves back and forth. A longitudinal wave is one that travels along the direction of propagation. In this demonstration, two different devices will be used to illustrate longitudinal propagation. An aluminum bar is used to produce transverse waves. A slinky will be used to visually show longitudinal propagation along the slinky. Finally, the aluminum bar is again used to show differences in longitudinal and transverse wave propagation.
Discover sounds in the sea.
Authors: Vigness-Raposa Kathleen J. , Scowcroft Gail , Knowlton Christopher , Morin Holly , Worcester Peter F. , Miller James H. , Ketten Darlene R.
Published: July 2011

Investigate the various sounds that exist in our oceans in the Name That Sound interactive game. Sounds are presented and contestants are challenged to select the correct source of the sound from four possible options. This hands-on activity is based on the Audio Gallery, with a wide range of sounds produced by animals, people, and natural processes like waves and rain, at the “Discovery of Sound in the Sea” website (http://www.dosits.org). The DOSITS website explains the physical science of underwater sound and how people and animals use sound for many purposes, like communication and exploration. It also has three major resource sections: media, teacher, and student. The media resources include a Facts & Myths quiz, Frequently Asked Questions, and PDF downloads of a tri-fold pamphlet and 12-page educational brochure. The teacher and student resources include structured tutorials, classroom activities, and educational games, including the Name That Sound game. The DOSITS team recently launched a significantly redesigned website. The look and feel has been refreshed without losing functionality and content. The redesign includes an interactive front page and Audio Gallery, and a redesigned Scientist Gallery. Participants at ASA will receive free CD-ROMS. Come explore the science of underwater sound with DOSITS!
Hot topics in underwater acoustics.
Authors: Zurk Lisa
Published: July 2011

Research in underwater acoustics continues to evolve with the development of more capable acoustic systems, the introduction of new applications for underwater sensing, and with the emergence of integrated physics-based processing approaches, which provide robustness in variable ocean environments. This talk provides an overview and examples of some of the active research areas in underwater acoustics, with particular emphasis on three topics. The first topic is the characterization and exploitation of underwater noise. This topic has research activities in both environmental monitoring (and potentially mitigation) of noise produced from anthropogenic activities, as well as the creative utilization of ambient noise structure for passive underwater sensing. The second topic area is the application of the waveguide invariant in active and passive sonar systems for enhanced detection, localization, classification. The invariance structure is being explored for deep water environments, with application to different sensor geometries and potentially new beamforming constructs. The final research area is underwater acoustic communications and networking. These systems are evolving to higher data rates (both single and multi-carrier) with adaptive and self-organizing algorithms to fully exploit the capabilities of distributed undersea networks.
Building the oceanic concert hall: Multidisciplinary applications of acoustics for ocean observing systems.
Authors: Dushaw Brian D.
Published: July 2011

Over the past decade, the oceanographic community has been constructing integrated multidisciplinary observational systems to serve the immediate and long-term needs of the society (www.ioos.gov). The overarching goal is to provide practical products and information to society as a return on its long-term investment in oceanographic research. Demand is increasing for information about the state of our oceans to address a myriad of issues ranging from climate variability to fisheries management to public education. Customers for ocean observing systems range from government agencies to commercial shipping companies to the science projects of high school students. Two developing “observing systems” in the Pacific Northwest are the Northwest Association of Networked Ocean Observing Systems (www.nanoos.org) and the Neptune Undersea Cable Network (www.interactiveoceans.washington.edu). The oceans are largely transparent to sound; hence, oceanographic, biological, and signal-processing acoustic techniques are primary tools for ocean observation and engineering. The opportunities and value of acoustical observations and techniques within these systems are boundless, yet incorporation of these techniques has been opportunistic and ad hoc. Coordination of the acoustical applications is essential. Organizations advocating acoustical observations face enormous challenges of planning, implementation, and data management to bring acoustical tools to fruition for ocean observing systems.
Influence of ribs on the radiation of a piano soundboard.
Authors: Chaigne Antoine , Vuillez Romain , Viggiano Roberto
Published: July 2011

The presence of ribs attached to the soundboard of a piano influences the transfer of energy from strings to soundboard through the bridge. Measurements and simulations show that the input admittance at the bridge depends on the distance between string end and ribs. A second effect of ribs is related to the periodicity of rib spacing. Simulations of soundboard driving-point mobility were conducted for three different spacing rules: periodic, almost periodic with small random fluctuations, and random distribution of spacing within a given interval. The results show that aperiodicity reduces the emergence of strong peaks in the mobility spectrum. Finally, measurements and simulations were made on the radiated sound in an anechoic chamber. The primary effect of ribs here is due to the additional stiffness, compared to a flat plate, which reduces the critical frequency and thus increases the frequency range where the radiated sound power is significant. In addition, the presence of ribs influences the directivity of the sound field: strong differences exist in a plane parallel to ribs compared to ribs perpendicular to them. These results also serve as references for comparison with time-domain simulations of the complete piano including strings, soundboard vibrations, and radiation.
Modeling and numerical simulation of a piano.
Authors: Chabassier Juliette , Chaigne Antoine
Published: July 2011

A complete model of a piano is built, in order to account for the acoustical behavior of the instrument from excitation to sound. A nonlinear hammer strikes the strings. The precursor at the bridge can be explained by the presence of a longitudinal vibration in the string, which is nonlinearly coupled to the transversal vibration using the geometrically exact model. A nonstandard condition must be written for the bridge model, in order to transmit transversal and longitudinal vibrations of the strings to the soundboard. A Reissner Mindlin plate model is used for the soundboard, which radiates in the air. The coupling must be reciprocal so that a global energy is preserved. A numerical discretization is proposed for the whole system. A first difficulty is due to the nonlinearity of both strings and hammer. Another one arises from the different couplings of the system: hammer/string, string/soundboard, and soundboard/air. An innovating, energy preserving scheme is built for the nonlinear system of string equations, and an energy technique is adopted for the whole problem to ensure numerical stability. The resulting complete numerical scheme conserves a discrete and consistent global energy. Numerical results are presented and compared to measurements.
Physical model of the harpsichord plectrum-string interaction during slip-off.
Authors: Perng Chao-Yu J. , Smith Julius O. , Rossing Thomas D.
Published: July 2011

A harpsichord plectrum model and the plectrum-string interaction [C.-Y. J. Perng et al., J. Acoust. Soc. Am. 127, 1733(A) (2010)] with its subsequent implementation using a digital waveguide [C.-Y. J. Perng et al., J. Acoust. Soc. Am. 128, 2309(A), (2010)] have been proposed by us earlier. In this presentation we model the final moments of the string sliding past the plectrum, a critical stage of the pluck action that provides the distinctive excitation signal. In addition, by incorporating motion data of a real harpsichord jack, we present here a revised model on the harpsichord plucked string mechanism.
Musical controller for wind instrument using Max/MSP software and ATmega128.
Authors: Chowdhury Anuva , Cho Sangjin , Chong Uipil
Published: July 2011

The physical appearance of the proposed musical controller is composed of eight buttons, two potentiometers, and one for sensing resistor (FSR). The eight buttons each represent one octave scale. One of the two potentiometers selects kinds of instruments, while the other changes octave. FSR is for tremolo effects. For the real-time play, we used a Max/MSP software. All recorded sounds were sampled and quantized as 44 100 Hz and 16-bit, respectively. To use wavetable synthesis, samples of three to ten periods of the sounds were stored in each table. We also used an AVR board equipped with an ATMega128 8-bit microcontroller. This microcontroller acts as an interface circuit between the sensor and PC as well as transmits data about notes and playing style to the Max/MSP by using a serial communication under RS-232. Considering that the controller is a prototype of the stand-alone musical instrument and is worked on the PC, the latency caused by RS-232 communication can be easily solved embedding the program onto the microprocessor such as digital signal processor. [Work supported by Basic Science Research Program through the National Research Foundation of Korea (NRF) funded by the Ministry of Education, Science and Technology (No. 2010-0005092).]
An approximate causal Green’s function for the Stokes wave equation.
Authors: McGough Robert J.
Published: July 2011

A causal analytical approximation is derived for the time-domain Green’s function of the Stokes wave equation. This causal approximation, which contains a modified Bessel function of the first kind, is the exact analytical time-domain Green’s function for a related wave equation that approaches the Stokes wave equation in the low frequency limit. An asymptotic expression is also obtained for this approximate Green’s function. The causal approximation and the asymptotic expression are evaluated and compared to the numerically computed time-domain Green’s function and the analytical frequency-domain transfer function of the Stokes wave equation and to other causal and noncausal approximations. The results show that this approximate Green’s function closely matches the numerically computed Green’s function and the analytical frequency domain transfer function of the Stokes wave equation near the source, and at greater distances, the asymptotic expression accurately represents the time- and frequency-domain behavior of the Stokes wave equation. Time-domain analysis demonstrates that the analytical approximation and the asymptotic expression are dispersive, whereas causality is verified in the time-domain and in the frequency-domain. The differences and similarities between these causal approximations and other causal and noncausal approximations are also identified and discussed. [Work supported in part by NIH Grant No. EB012079.]
Radiation by a subwavelength-sized acoustic monopole: Energy considerations and application to subwavelength time-reversal focusing.
Authors: Bossy Emmanuel , Carminati Rmi
Published: July 2011

In this work, the radiation of pulsed acoustic waves by an acoustic monopole is theoretically studied based on energy consideration. Specifically, the instantaneous acoustic intensity and the instantaneous acoustic energy contained in a spherically symmetrical pulsed radiated field are calculated. In the case of a subwavelength-sized acoustic monopole, it is shown that the radiation of a given amount of energy involves energy exchanges between the field and the source that are much larger than the radiated energy. It is also shown that even after the source has been turned off, the instantaneous acoustic intensity may be directed inwards, although the field cannot exchange energy with the source anymore. By time-reversing the radiated field, we demonstrate that subwavelength refocusing requires a subwavelength energy exchanger, which is not only needed to absorb the incoming energy as well-known, but importantly also to transiently provide energy to the field before reabsorbing it. In other words, we demonstrate in a homogeneous medium that the acoustic sink needed to perform subwavelength time-reversal must transiently be an active source of energy.
Using relative amplitude and travel times from geometric acoustics to determine nocturnal effective sound speeds.
Authors: Blom Philip , Waxler Roger
Published: July 2011

On clear dry nights over flat land, a temperature inversion and stable nocturnal wind structure lead to an acoustic duct in the lowest few hundred meters of the atmosphere. An impulsive signal undergoes strong dispersion during propagation and is received at long ranges from the source as an extended wave train consisting of a series of distinct arrivals followed by a low frequency tail. The leading distinct arrivals have been shown to coincide with the direct and single reflection geometric ray paths. At a range of 2 Km from the source, these ray paths propagate through the lowest 50–70 m of the atmosphere while larger range arrivals reach higher into the atmosphere. Using the solutions of the eikonal and transport equations, travel times, amplitudes, and caustic structures of the arrivals can be determined. Arrival details can then be used to fit a low order Taylor series expansion of the effective sound speed profile. Approximations of the atmospheric conditions in the duct will be presented along with both predicted and measured arrival waveforms and meteorological data from the time of propagation.
R. Bruce Lindsay Award
Authors: Sabra Karim G.
Published: July 2011

The R. Bruce Lindsay Award (formerly the Biennial Award) is presented in the Spring to a member of the Society who is under 35 years of age on 1 January of the year of the Award and who, during a period of two or more years immediately preceding the award, has been active in the affairs of the Society and has contributed substantially, through published papers, to the advancement of theoretical or applied acoustics, or both. The award was presented biennially until 1986. It is now an annual award.