bluuebunny/crossref_metadata_2025 huggingface dataset:
New and developing seafloor petroleum extraction technologies: The noise of processing equipment operating in extreme environments.
Authors: Stocker Michael
Published: October 2010

Fossil fuel demands are driving petroleum extraction operations into deeper marine settings and into oil reservoirs that are also deeper beneath the sea bed. To facilitate this new operating paradigm, operators are commonly placing processing equipment on the sea floor at 5000 feet or deeper where depth pressures can easily be 100–200 atm (1400–2800 psi). In areas such as the Gulf of Mexico, the reservoir pressures alone can reach 20 000 psi at the well head. The extracted product can be a mix of oil, gas, sand, and water, which all route through a chain of processing equipment such as separators, pumps, de-sanders, and routing manifolds under high pressures and high volumes. This paper will evaluate the new and developing fossil fuel extraction and production equipment to determine the scope and scale of the mechanical noises that these new technologies generate.
Horizontally refracted acoustic signals on the hind side of a solitary internal wave in shallow water.
Authors: Badiey Mohsen , Katsnelson Boris , Lin Ying-Tsong , Lynch James
Published: October 2010

Interference patterns of low-frequency, horizontally reflected sound signal from an approaching nonlinear internal wave (NIW) front toward an acoustic track was shown earlier. [J. Acoust. Soc. Am. 127, 1974 (2010).] When the first wave of NIW crosses the acoustic track, a new interference pattern emerges before the appearance of the acoustic shadow zone. Similar to the approaching behavior shown earlier, the interference pattern on the receding side of the NIW is related to the modal interference due to the interaction of low-frequency acoustic waves and the NIW. In this paper, the modal results of the receding behavior are compared with the approaching behavior and the modal order and exchange in each case is discussed. [Work supported by ONR.]
Generation of multiple direct paths and resulting interference patterns during an experiment conducted during summer 2008 off the coast of Hawaii (KAM08).
Authors: Senne Joseph M. , Badiey Mohsen , Song Aijun
Published: October 2010

During the KAM08 experiment 3 h of continuous transmissions (10–20 kHz) were made between a ship-based source and a moored vertical array. Direct path data show modulations that indicate both constructive and destructive interference of acoustic rays. These interference patterns are likely due to multiple direct paths, which arise from refractions of the signal from different portions of the thermocline. The single bottom-interaction does not show multiple paths, possibly due to a steeper angle of interaction with the thermocline. Multiple thermistor strings show the spatial and temporal changes in thermocline for this geometry. The observed thermocline variations are used to approximate the conditions under which the multiple direct paths were observed. By using ray-based and parabolic equation models, the interface conditions necessary for generating the multiple direct paths are investigated. Results show that modulations in the spatial and temporal variability of the thermocline are a potential forcing mechanism for the observed interference patterns. [Work supported by ONR.]
On the acoustic radiation force in soft tissues induced by a focused ultrasound beam with transient modulation envelope.
Authors: Donsov Egor , Guzina Bojan
Published: October 2010

Building on a recently developed plane-wave model for Newtonian fluids, this work studies the acoustic radiation force (ARF) in soft tissues generated by an ultrasound beam, modulated by a transient modulation envelope whose dominant frequency (relative to that of the ultrasound carrier) is on the order of the Mach number. Following earlier treatments of the featured “narrow-beam” problem, the analysis focuses on the spatio-temporal distribution of the ARF in the focal region and approximates the tissue’s constitutive response by a non-linear thermo-viscoelastic model. In situations involving “low-frequency” beam modulation, the usual time-averaging procedure, which works for stationary (non-modulated) beams, is unfortunately not applicable for the resulting mean acoustic quantities still oscillate at the frequency of ultrasound. To deal with the problem, the asymptotic analysis bisects the time variable into a “fast” and “slow” component which frees the “fast” time averages of the acoustic quantities from undesirable rapid oscillations. The results demonstrate that the well-known formula for the ARF in a focal region is, for the problem of interest, appended by an additional term that is linear in the “slow”-time derivative of beam intensity. Selected results are included to highlight the effect of beam modulation on the spatio-temporal distribution of the ARF.
Separation of bacterial spores from flowing water in macro-scale cavities by ultrasonic standing waves.
Authors: Lipkens Bart , Dionne Jason , Costolo Mike , Stevens Amy , Rietman Ed
Published: October 2010

The separation of micrometer-sized bacterial spores (Bacillus cereus) from a steady flow of water through the use of ultrasonic standing waves is demonstrated. An ultrasonic resonator with a cross-section of 0.0254 × 0.0254 m2 has been designed with a flow inlet and outlet for a water stream, ensuring laminar flow conditions into and out of the resonator section of the flow tube. A 0.01905-m diameter PZT-4, nominal 2-MHz transducer is used to generate ultrasonic standing waves in the resonator. The acoustic resonator is 0.0356 m from the transducer face to the opposite reflector wall, with the acoustic field in a direction orthogonal to the water flow direction. At fixed frequency excitation, spores are concentrated at the stable locations of the acoustic radiation force and trapped in the resonator region. The effect of the transducer voltage and frequency on the efficiency of spore capture in the resonator has been investigated. Successful separation of B. cereus spores from water with typical volume flow rates of 40–250 ml/min has been achieved with 15% efficiency in a single pass at 40 ml/min.
Modal analysis of acoustic enclosures by dual reciprocity–boundary element method.
Authors: Mollinedo Helvio P. , Merodio Hector V. , Rabadan Itzala M. , Ortega Jose Angel H.
Published: October 2010

This work presents a modal analysis of acoustic enclosures by dual reciprocity–boundary element method (DR-BEM). The dual reciprocity formulation allows the use of known Laplace fundamental solutions. The boundary element method possesses the main characteristic that reduces the dimension of the problem upon the discretization of the boundary to calculate the acoustic eigenfrequencies. The DR-BEM formulation has been evaluated with the use of interior nodes and some known radial basis functions. The results obtained by this technique have been compared with the more extensively diffused competitor, the finite element method (FEM). Results shows good agreement between both methods.
Characterization of a configurable magnetostrictive drive.
Authors: Porter Scott P. , Thompson Stephen C. , Meyer Richard J.
Published: October 2010

Magnetostrictive drives require features that are not standard in piezoelectric motors: a magnetic return path, a biasing mechanism, and eddy current mitigation. In spite of this, there is not a standard design for a self-contained magnetostrictive drive. This work investigates a configurable high-power magnetostrictive drive design built from Galfenol that can be tailored to fit a variety of applications. Based on this design, a prototype drive has been constructed from laminated Galfenol structures and neodymium biasing magnets. In this paper, the author will discuss the fabrication and characterization of the device and compare the results to simulated performance.
Teaching acoustics to musicians.
Authors: Basso Gustavo Jorge
Published: October 2010

Teaching acoustics to music students shows certain challenges but, at the same time, very interesting possibilities arise. The main difficulty appears from the impracticality of applying the physical and mathematical concepts that exceed the pre-college level. On the other hand, it is feasible to take advantage of the student’s well trained musical ear that allows us to teach some hard concepts without losing the core of them. During our experience in the Universidad Nacional de La Plata, we have developed a methodology that covers the main steps of the acoustical chain—generation, propagation, and reception/perception—in a spiral fashion that allows students to keep a global vision of the process. Throughout each issue, the perceptual aspect follows immediately after its physical counterpart, and this temporal closeness enables the students to use their audio skills. This way, the two central mathematical tools of the course—the Fourier theorem and the acoustical uncertainty principle—are comprehended without difficulty. They are applied in the perceptual analysis of some simple signals and, almost immediately, to the analysis of some musical passages specially chosen. As a result, the student sees that the acoustical science is related to the musical practice, stimulating their interest in the discipline.
Programming abilities description for engineering students.
Authors: Felipe Federico D. , Acevedo Maria Elena M. , Martinez Ignacio S.
Published: October 2010

This paper focuses in specifying competences for high-level language programmers. It is very important to train engineering students in programming, because this forms the basis for solving many problems. It is not necessary to write programs in a high-level language for the solution for many problems; however PDS programs are used for acoustics, telecommunications, computer science, and many other fields. There are specified seven competences for programmers and there is a proposal for basic exercises to obtain these competences.
A successful undergraduate program in acoustical engineering.
Authors: Surez E.
Published: October 2010

Since 1967 undergraduate programs on acoustics have been offered at the Univ. Austral de Chile. At the beginning a 4-year program on Sound Engineering was offered, which was developed and expanded in 1981 to a 5-year program on Acoustical Engineering. In 2001, the curriculum was again extended to a 6-year program on Civil Engineering in Acoustics. So far, more than 300 graduates have completed training in the programs: 65 sound engineers, 230 acoustical engineers, and 10 civil engineers in acoustics. Although these programs have been unique in Latin America for many years, more recently some other programs are offered in Chile and other countries. However, most of these new programs have had the cooperation of our former graduates as faculty members and directors. The current curriculum includes mathematics, physics, music, economics, and administrative, computing, and engineering sciences, as well as acoustics and sound engineering technology. The working field of these graduates has been very wide, including areas such as recording and sound reinforcement, noise and vibration control, environmental protection, and acoustic design of architectural spaces. Many of our graduates have obtained advanced degrees abroad, working with success not only in Chile but also in several countries in Latin America, Europe, and the U.S.
Is acoustic comfort underestimated by architects?
Authors: Viveiros Elvira
Published: October 2010

A survey was carried out among architects and undergraduate students in order to identify their understanding of acoustics. It can be seen that some myths in architectural acoustics are solidly established. Simultaneously, the viewpoint of lay people was also looked into and it was found that their demands are often in opposition to what architects consider relevant. The reason why what is expected in buildings is not delivered by the architects are discussed and their academic profile is investigated.
On a new course: The physics of music and language.
Authors: Ortega-Rodríguez Manuel
Published: October 2010

The presentation describes the experience of designing and teaching the course “The Physics of Music and Language” at Universidad de Costa Rica. Even though the course is taught at the Physics Dept., it has drawn large enrollment from music students as well as other majors such as education, biology, and the social sciences. The course has now been taught five times and has become part of the mandatory curriculum for music students. In this course, a traditional approach to acoustics is extended to include topics on psychoacoustics as well as a discussion of the anthropology of music and the relationship between music, language, and poetry. This makes the course a truly transdisciplinary exercise. Rather than despairing at the heterogeneity of the audience, it is argued that the situation should be seen as an opportunity to generate debate among students from diverse majors.
Brain synchronization for learning.
Authors: Lizana Pablo
Published: October 2010

For decades the technology of binaural beats has been used effectively to produce several brain states associated with profound psycho-physical relaxation and altered states of consciousness for learning. These rythms were discovered by Dove in 1839, but only when it was published by Oster in 1973 was its significance appreciated. It seems that these states are created by the beats through the nucleus of the medulla to the cerebral cortex, with frequencies below 30 Hz, by the combination of two higher frequencies, which generates a third one in the required range. Then the brain follows this signal. Although these sounds are used for example, to promote creative statements, improve concentration and memory, and to help eliminate anxiety, insomnia, depression, chronic fatigue, headaches, or psycho-physical, it must be kept in mind that not everyone respond the same way to the same stimuli. The general idea is to apply it in order to improve learning.
Perception and acoustical analyzes of traditionally orchestrated musical structures versus non-traditional counterparts.
Authors: Kendall Roger A , Vassilakis Pantelis N.
Published: October 2010

Many studies use single-tone or dyadic-unison stimuli in the exploration of musical timbre. Musical contexts, however, often employ larger numbers of simultaneous voices. Recently, we have conducted perceptual/acoustical studies of consonance, dissonance, and roughness in multitimbral triads often finding that chord quality (major, minor, diminished, augmented) had perceptual primacy over timbre. The present research extends this work to include major and minor triads in similarity scaling, blend, and identification experiments. In addition, four-tone chords were orchestrated according to procedures articulated in orchestration texts: juxtaposition, interlocking, enclosure, and overlapping. Sampled instrument tones (Sibelius/Kontakt) of oboe, flute, and clarinet were used, allowing comparison to previous research, and non-traditional combinations with saxophone and trumpet were included. Triads with each tone assigned a different timbre provided nontraditional orchestrations to be compared and contrasted with traditional orchestrations that duplicated timbres within a chord. Perceptual data were correlated to acoustical analyzes of spectral distribution and time-variancy.
Efficient computational modeling of piano strings for real-time synthesis using mass-spring chains, coupled finite differences, and digital waveguide sections.
Authors: Smith Julius , Kuroda Junji , Perng Jack , Abel Jonathan
Published: October 2010

In previous waveguide synthesis models for piano, longitudinal waves have been neglected, although it is known that there is audible coupling from transverse to longitudinal vibration in piano strings [e.g., Conklin lectures, http://www.speech.kth.se/music/5/_lectures/]. A general method for accurate 3-D string simulation is the mass-spring chain [Rowland and Pask, Am. J. Physics, (1999)], which reduces to half-coupled wave equations at low amplitude. At yet lower amplitudes, when string slope times string curvature can be neglected, the longitudinal and transverse waves reduce to linear superposition, for which digital waveguides are most efficient for simulation [http://ccrma.stanford.edu//~jos/pasp/]. In this presentation, a hybrid piano-string model is proposed which employs a mass-spring chain at the hammer and a digital waveguide model elsewhere. The spatial force vector at either edge of the mass-spring section drives (and couples) three digital waveguide models corresponding to one longitudinal and two transverse vibrational components. At the bridge, a 3-D force vector is formed which drives the vertical, horizontal, and longitudinal transfer functions of the soundboard. A simplified 2-D model neglects one of the transverse planes of vibration. Beyond piano strings, the model can transition adaptively (across time and/or space) among digital waveguides, half-coupled finite-difference schemes, and full mass-spring-chain models.
Playing the Continuum Fingerboard: New performance techniques for a continuous instrument.
Authors: Haken Lippold , Eagan Edmund
Published: October 2010

The Continuum Fingerboard is a Midi performance controller that offers real-time 3-D continuous control for every finger that is placed on its playing surface. Its novel design presents unique performance challenges and possibilities. Like a fretless instrument, the performer must rely on audio feedback, finger memory, and manual dexterity for accurate intonation and expression. The Continuum requires its own technique, different from any other instrument. With a traditional music keyboard, it is normal for the performer to feel the key hit a hard stop for each note, even if the performer is playing quietly. Also, traditional keyboards are usually velocity sensitive; a single velocity value is determined by the speed of the key movement. In contrast, the Continuum is both velocity sensitive and, more importantly, pressure sensitive. It initially outputs a single velocity level, and then continually outputs a stream of pressure, pitch, and front-back position updates until the note is terminated. The Continuum is designed for a fast and relatively lighter touch than other musical instruments, so it is unusual to hit the hard stop (or “bottom out”) except for the very loudest notes. Various Continuum-specific performance techniques utilized by performers will be presented.
Sound design for the Continuum Fingerboard: Effective mapping of finger data to synthesis parameters.
Authors: Eagan Edmund , Haken Lippold
Published: October 2010

The Continuum Fingerboard tracks the position and pressure of fingers on its playing surface. Its internal computer encodes this finger placement information into Midi data streams. The Continuum has an internal dsp that provides high-quality on-board timbres, but often performers expand on this built-in sound palette with sounds from external synthesizers. The Continuum connects to synthesizers via Midi, and/or to analog modular synthesizers via the Continuum voltage converter. It can be used to control a wide variety of sound synthesis algorithms. When designing sounds for the Continuum, it is important to keep in mind that it is not a keyboard. Sounds that work well with keyboards generally do not work well with the Continuum and will not exploit the full performance potential of the instrument. Sound design techniques for producing high-quality synthesis algorithms suitable for the Continuum will be presented.
Soil profile meter with ultrasonic sensor.
Authors: Garcia Martinez Ana Trinidad , Martinez y Cardenas Jose Rodolfo , Ruiz Vega Jaime
Published: October 2010

The Agricola research at CIIDIR Oaxaca requires characterization of the line profile made by the seeding appliances, in order to allow the people involved to get a surface description for quality control purposes. This paper presents the design of a profile meter with a non contact ultrasonic sensor, so the soil surface is not altered. The system is programmable, simple to use, and stores the measurement results.
Maximum power extraction from ocean wave energy conversion devices.
Authors: Pierce Allan D.
Published: October 2010

A generic energy conversion device at the ocean surface is considered. Incident water waves of given frequency and amplitude are incident on the device and one asks just how much power PD can be converted into electricity by the device. A primary descriptor of the water wave is the time averaged power PW carried by the wave, integrated over water depth, per unit length of wave crest. The relevant descriptor of the device’s performance is the capture length, equal to the quotient PD/PW. Basic theory, not taking into account the details of the device’s design, predicts a maximum attainable capture length in terms of the wavelength and of the overall dimensions of the device. For example, the recent work by Thiam and Pierce indicates that the maximum capture length of a Pelamis-like device is equal to a constant times the area of the device viewed from above divided by the wavelength. Another analysis for a “point” vibrating resonator indicates that the maximum capture length is of the order of 40 times the cross-sectional diameter of the device. Knowledge of the maximum capture length enables a quick assessment of whether any proposed wave energy farm will deliver a desired amount of power.
Segmented and continuous beam models for the capture of ocean wave energy by Pelamis-like devices.
Authors: Thiam Amadou G. , Pierce Allan D.
Published: October 2010

Pelamis-like devices are elongated structures (floaters rather than sitters) aligned perpendicular to crests which flex or rotate longitudinally because of incoming waves. The Pelamis device is similar to Cockerell’s rafts and consists of a chain of cylinders connected so that there is a resistance to the changing of the pitching angles between adjacent cylinders. The flexing causes pistons to move, which, in turn, drive turbines that generate electricity. If the segments are sufficiently short and numerous, the device can be modeled as a continuous beam, with the conventional partial differential equation for the beam dynamics being modified by inclusion of a term proportional to the time derivative of the fourth spatial derivative of the vertical beam displacement. The effect of the incoming wave is approximated by a quasi-static buoyancy term proportional to the difference of the local heights of the beam’s center line and that of the water height associated with the incoming wave. A theory is given for the prediction of the maximum power output that can be achieved for the continuous beam model and of how well different segmented models can approach the ideal limit. Performance is markedly better when the segment lengths are less than a quarter wavelength.
Vibration energy harvesting: A basis for novel technologies.
Authors: McCoy John
Published: October 2010

At the forefront of the art of vibration energy harvesting are devices capable of harvesting at a rate measured in milliwatts. At this level the energy harvested in a reasonable time is a vanishing percentage of the total energy in the vibrating structure. The applications for such energy harvesters are to power small sensors and micro-computers. Looking forward, one might ask: Is an energy harvester capable of harvesting a significant percentage of the energy available in a vibrating structure technically feasible? What would be the behavior characteristics of such an energy harvester? And, to what uses could the harvester be put? In this presentation, the challenges to harvesting at the level envisioned are identified, the outlines of a device described as a “meta-material vibration energy harvester” are described, and several novel technologies made possible by the availability of the device are discussed. Among these novel technologies would be one based on designing structures for the purpose of capturing the kinetic energy in a multiplicity of dynamic processes that occur in nature, as structural vibration energy can then be subsequently converted to electricity.
The impact of talker and prosodic variability on speech perception in noise for children with dyslexia.
Authors: Hazan Valerie , Rosen Stuart , Messaoud-Galusi Souhila
Published: October 2010

This study investigated whether children with dyslexia (DYS) are more affected by talker and prosodic variability when perceiving speech in noise than age-matched average readers (ARs). 34 DYS and 25 AR children carried out syllable-initial consonant perception tests, with natural consonant-vowel (CV) tokens in multi-talker babble noise (0 dB signal-to-noise ratio). Twelve-alternative identification tests were presented in four conditions varying in degree of talker and intonation variability. The discrimination of place (/bi-di/) and voicing (/bi-pi/) contrasts was investigated with the same test conditions. When identifying CV syllables in noise, the DYS group made more errors than the AR group, but only for conditions with variable intonation. This group difference was primarily due to poorer perception by DYS children of fricative voicing and greater /m/-/n/ confusions, i.e., contrasts with less salient acoustic differences. DYS children showed lower discrimination scores than AR children for both the voicing and place contrasts, across the single/multiple talker and single/multiple intonation conditions. These results are compatible with the view that some children with dyslexia have speech perceptual acuity at the lower end of a normal range, which may be exacerbated by non-sensory factors, and thus affected by the type of task used. [Work funded by the Wellcome Trust (076499/Z/05/Z)].
The development of referential understanding.
Authors: Hendrickson Kristi , Sundara Megha
Published: October 2010

Previous research has shown that 14-month-olds are able to form arbitrary word-object associations. [Werker et al. 1998).] However, merely associating words and objects is insufficient for the higher level concept that words “stand for” objects. In order to investigate at what age referential understanding emerges, we tested 14- and 17-month-olds on their ability to comprehend the correct meaning of a word when the word and object are presented separately from one another. We used an infant-controlled version of the visual fixation procedure presented using HABIT X [Cohen et al. (2004).] Infants heard audio passages containing two target nouns. Subsequently, infants were presented with silent animations of four words—two with which they were familiarized (target) and two they had not heard during familiarization (non-target). Looking time to target and non-target animations were averaged and compared using paired t-tests. The 14-month-olds looked longer at the target animations although this result was not significant. In order to identify at what age infants understand the referential relationship between words and objects, we are currently testing 17-month-olds on the same task. The results will be discussed in the context of emerging referential understanding of nouns during development.
Four and a half month-olds’ phonotactic knowledge.
Authors: Thatte Victoria , Sundara Megha
Published: October 2010

For several decades, linguists and psychologists have been researching whether infants primarily utilize domain-specific cognitive mechanisms or domain-general cognitive skills to help them learn language grammar from a continuous acoustic signal. This research investigates that question by testing 4.5-month-olds’ reactions to phonological patterns motivated by articulatory simplification versus patterns without such motivation. Results from experiment 1 show that 4.5-month-olds listen significantly longer to stimuli following a pattern with articulatory motivation (intervocalic voicing: “pi, fi, pivi”) than those following an unmotivated pattern (intervocalic devoicing: “tu, zhu, tushu”), *N*(12) *p*=0.02. Results from a control experiment indicate that this preference is not due to the 4.5-month-olds’ preference for the “inputs” to the patterns (“pi, fi” versus “tu, zhu”), *N*(15) *p*=0.46. The results of experiment 1 are interpreted as advancing the claim that humans have an unlearned, domain-specific attentional bias in favor of phonetically motivated phonological processes. Additionally, the control experiment results demonstrate that the bias does not emerge from early sensitivity to native language phonotactic constraints regarding placement of voiced versus voiceless fricatives. A further control experiment will ensure that the first experiment’s results are not due to infant preference for only the two-syllable units in the voicing pattern.
Perceived similarity of English, Korean, and Japanese stops for native English, Korean, and Japanese listeners.
Authors: Nozawa Takeshi , Cheon Sang Yee
Published: October 2010

Stops in /Ca/ context produced by four native speakers of English, Korean, and Japanese were presented in an AXB discrimination task to examine and compare interlingual stop identification, where the listeners heard three stimuli in sequence and decided whether the second (X) is more similar to the first (A) or to the third (B). Fifteen native speakers of the three languages heard three stops per trial, each of which was produced by a native speaker of a different language. The results revealed the effects of the listeners’ respective L1. While English and Korean listeners perceived Korean aspirated stops as closer to English voiceless stops than to Japanese voiceless stops, Japanese listeners perceived Korean aspirated stops as closer to Japanese voiceless stops (which are said to be unaspirated or weakly aspirated). English and Korean listeners perceived English voiced stops closer to Korean tense stops rather than Japanese voiced stops, while Japanese listeners perceived English and Japanese voiced stops as similar. These results suggest that perceived similarity of interlingual stops are affected by the phonological system of one’s L1, and may imply that perceived boundaries between voiceless and voiced stops and between aspirated and unaspirated stops are different across the listeners’ L1.
Perception and production boundaries between fricative and affricate in Japanese speakers.
Authors: Amano Shigeaki , Yamakawa Kimiko
Published: October 2010

Perception and production boundaries between a fricative /s/ and an affricate /ts/ in Japanese were investigated to clarify their characteristics and relationships. In a perception experiment, 40 Japanese native speakers listened to 2-D stimulus continua coordinated with a rise time and the sum of steady and decay time of [s]/[ts] in 1–4-mora Japanese words. They made two-alternative forced choice between /s/ and /ts/, and the identification ratio of these phonemes was obtained. In a production experiment, one female Japanese speaker pronounced 403 Japanese words in 1–4 moras long having /s/ or /ts/. The pronounced words were digitally stored with 16-bit quantization and 16-kHz sampling frequency. A rise time and the sum of steady and decay time of [s]/[ts] in the words were measured with a waveform editor. Logistic regression analysis for perception data and discriminant analysis for production data revealed that the perception and production boundaries were both represented by very similar linear functions with the rise time and the sum of steady and decay time of [s]/[ts]. This result shows a correspondence of perception and production boundaries between /s/ and /ts/, suggesting a close connection between speech perception and production.
Acoustic characteristics of central vowels in American English.
Authors: Allen Phoebe E. M. , Weismer Gary G.
Published: October 2010

Surprisingly little is known of the acoustic characteristics of the central vowels caret (/̂/) and (/ə/), even though they occur frequently in English and contribute significantly to its rhythmic characteristics [Umeda, J. Acoust. Soc. Am. 58, 434–445 (1975)] Moreover, analyzes of schwa in Dutch [van Bergem, Speech Comm. 16, 329–358 (1995)] show that its acoustic characteristics are surprisingly variable and very much affected by surrounding context. This paper reports temporal and spectral measurements of the vowels /̂/ and /ə/ produced by 10 speakers (5 male and 5 female) from central Wisconsin. These vowels were produced in real words in a variety of phonetic environments in both a reading task (three different rates of speech) and in spontaneous speech. Descriptive analyzes will be presented as well as discriminant function analysis in which the acoustic measures are used to classify the two vowels. The effects of speech rate and speech material on the two vowels will also be examined.
Regional dialect variation in standard Italian.
Authors: Tamati Terrin N.
Published: October 2010

Italy is a linguistically diverse country where a standard language (standard Italian) is in contact with various regional languages. Within this setting, the regional languages have influenced the standard Italian spoken throughout Italy and regional dialects of Standard Italian have emerged. Previous studies on the regional dialects of Standard Italian have predominantly taken a descriptive approach to discussing the phonological differences between the dialects and a broad acoustic study of the dialects is lacking. The present study seeks to identify several acoustic characteristics of regional varieties of spoken standard Italian. In the study, an acoustic investigation of vowels produced by male and female talkers from Northern, Central, and Southern Italy was carried out. Acoustic measures obtained included vowel duration and first and second formant frequencies. These measures were used to examine possible dialectal differences in vowel quality, vowel duration, and the overall acoustic vowel space. From these measures, several dialect-specific phonetic features were identified. These dialect-specific acoustic properties will be discussed. The results of the acoustic analysis will serve as a basis for future studies investigating the acoustic properties of the regional varieties of standard Italian and for identifying the types of cues listeners might use to distinguish the dialects.
Frequency/wavenumber domain multiple channel estimation.
Authors: Preisig James
Published: October 2010

The estimation of the impulse response of the acoustic channel between a transmitter and one or more receivers is important to establishing reliable communication links in high-rate, phase coherent communications systems. [J. Preisig, J. Acoustic. Soc. Am. 118, 263–278.] In underwater acoustic communication systems, the channel impulse response often exhibits both a large delay spread and rapid fluctuations, which make the channel estimation problem challenging. One method of improving the ability to estimate the time-varying channel impulse response is to reduce the ratio of the number of parameters needed to represent the channel divided by the number of observations available to generate estimates of those parameters. In systems where the channel impulse response is characterized at a sample rate higher than the Nyquist rate (i.e., fractionally spaced systems), this can be accomplished by representing the channel in the frequency domain. Fourier coefficients are estimated only for the bandwidth of the transmitted signals. This has been shown to substantially improve channel estimation accuracy. This work extends that approach to estimation of the impulse response at an array of sensors by representing the spatial structure in terms of wavenumber coefficients. The approach is derived and its performance is demonstrated using field data.
On the use of instantaneous acoustic intensity for the classification of defects in air-filled pipes.
Authors: Horoshenkov Kirill , Bin Ali Tareq , Tait Simon
Published: October 2010

The sound pressure in an air-filled pipe has been measured using six pairs of matched microphones. These data have then been analyzed to determine the six vectors of instantaneous particle velocity and acoustic intensity. The acoustic intensity vectors have been processed coherently using the matched field processing (MFP) technique to study the effect of changes such as blockages, cracks, and water level on the ambiguity surface. The results show that the technique is sensitive to very small changes in the pipe cross-section and the boundary conditions on the pipe wall. It enables us to locate a defect to within 1% of the measurement range. It also enables us to discriminate between blockages and wall cracks using the spectral characteristics of the ambiguity surface and the intensity spectrograms. The results are compared against that obtained via a standard 2-D cross-correlation method. The proposed MFP technique does not require acoustic modeling of the sound field in the pipe, but it relies on the sound pressure data recorded on three or more microphones in the absence and presence of a defect.
A theoretical study of sound field reproduction with a planar array of loudspeakers.
Authors: Fazi Filippo M. , Nelson Philip A.
Published: October 2010

The problem of reproducing a desired sound field with a planar array of loudspeakers is addressed. The array is modeled by a continuous distribution of monopole-like secondary sources arranged on an infinite plane, and the reproduced field is represented by a single layer potential. The target field is defined on a plane parallel to the secondary source layer. An expression for the solution is derived and this can be regarded as a linear superposition of propagating and evanescent plane waves. It is shown that the contribution of the evanescent waves does not represent the evanescent component of the desired field and is not limited to the near-field of the array. For this reason this component of the field is called pseudo-evanescent. The case of the so-called focused sources is analyzed. It is shown that an exact solution of this problem does not exist, but an approximate solution can be computed using regularization techniques (for example, by avoiding an exact reproduction of the pseudo-evanescent component of the target field). The time-reversal technique, widely used by wave field synthesis systems for the reproduction of focused sources, is also studied.
Feasibility study of underwater acoustic communications between buried and bottom-mounted sensors.
Authors: Edelmann Geoffrey F. , Zajić Alenka G.
Published: October 2010

This paper presents a feasibility study of underwater communications between buried sensor network nodes. To investigate this problem, two experiments have been conducted, where some or all sensor network nodes are buried in the sediment. The orthogonal frequency division multiplexing (OFDM) communications have been chosen to test underwater communications because of its unique strength in handling transmissions over long dispersive channels. It is demonstrated that error-free performance can be achieved between buried sensor nodes using the proposed OFDM receiver. [Work supported by the Office of Naval Research.]
Beamforming methods for reproduction of three-dimensional sound fields over spherical loudspeaker arrays.
Authors: Atkins Joshua , West James
Published: October 2010

Methods for capturing arbitrary 3-D sound fields using spherical microphone arrays have advanced to the point where commercially available arrays with performance up to fourth order are available. However, the reproduction of these captured fields for human listeners is still held back by issues such as the presence of scatterers (humans, furniture, etc.) within a loudspeaker array which distort the reproduced field, the fact that the field is only captured accurately up to an aliasing frequency of around 1–3 kHz, and technical issues dealing with sampling the sphere and constructing the array properly. This work looks at the reproduction of the acoustic field as a set of beamforming optimization problems and proposes various constraints to address these issues. The proposed subband design leverages the advantages of mode-matching at low frequencies with other optimization methods at mid and high frequencies that take into account the presence of scatterers and perceptual cues. Comparisons with VBAP, conventional Ambisonics, and dual-band decoding based on Gerzon’s localization theory are also presented.
Eigenvector adaptive noise suppression applied to vector sensor arrays.
Authors: Daniels Jeffrey A.
Published: October 2010

Due to the characteristics of acoustic intensity beamforming, it is desirable to reduce the noise level of the measured spectrum. In this paper, eigenvector noise suppression is adaptively applied to a vertical line array of underwater acoustic vector sensors. Eigen decomposition is applied to the cross-spectral density matrix (CSDM) of the beamformed complex spectrum from an array of Wilcoxon VS-206 vector sensors. The array elements are integrated sensors with a triaxial accelerometer and an omni hydrophone. The beamformer steering vector is applied to each of the eigenvectors for a range of steering directions within the noise field of interest. Using a known target bearing, the signal-to-noise ratio (SNR) is computed for the beamformed eigenvectors as the frequency averaged ratio between the target steering direction and all other steering directions. The eigenvectors with an SNR below a certain level are determined to be dominated by interference. These eigenvectors are subtracted from the CSDM using a projection matrix to suppress the interference. The results using calibrated source level data are presented. [Work supported through ILIR grant from Carderock Division, NSWC.]
Difficulties in the indirect experimental inference of the low-frequency intrinsic acoustic attenuation in marine sediments.
Authors: Pierce Alllan D. , Carey William M.
Published: October 2010

Intrinsic attenuation is ideally measured for plane compressional waves in an unbounded homogeneous medium. For typical marine sediments, at frequencies below 5000 Hz, sufficient realization of this idealization cannot be achieved. Reported indirect measurements typically have source and receiver in the water column and require hypothesized theories and previously measured parameters. An extended theory-based discussion is given of various experiments with substantially different designs that have led to widely different results. Relevant comparisons are the attenuation at 1 kHz and the general dependence on frequency over the range 100–1000 Hz. Data from a single experiment are typically amenable via approximate graphing as being proportional to frequency raised to some power n, usually between 1 and 2. Some theoretical notions say that n should be exactly 2, others that it should be exactly 1 or perhaps only slightly greater than 1. It is pointed out that most experimental designs involve questionable assumptions. If the true exponent is 2, then the attenuation at very low frequencies is going to be especially difficult to infer. Other physical mechanisms can enter into an indirect inference of attenuation, and mechanisms that would lead to a perceived linear frequency dependence seem to be ubiquitous.
Sound speed dispersion measurements and their interpretation in the presence of a shallow buried layer.
Authors: Osler John C. , Hines Paul C.
Published: October 2010

Sharing the same experimental apparatus, two complementary techniques were developed to measure the frequency-dependent speed of sound in marine sediments during the SAX04 sea-trial. The first technique enabled direct time-of-flight measurements of acoustic wave speed along all three Cartesian axes. The second technique determined the acoustic wave speed based on the arrival angle of pulses generated in the water column and refracted upon entry into the seabed. None of the results could be modeled or explained when the seabed was parametrized as a sand half-space. However, both techniques suggested the presence of a thin muddy layer, 0.05–0.2 m thick within the top 1 m of the sediment. For the time-of-flight technique, in the vertical direction, the interference from the layer created considerable apparent variability in sound speed measurements. In the horizontal direction, the layer caused the measured sound speeds to be lower than what a simplified poro-elastic model would predict, unless one accounts for the higher porosity material in the layer. For the arrival angle technique, the layer explains the complicated frequency- and geometry-dependent results; however, the sound speed dispersion was a second order effect that could not be unambiguously extracted.
Shear wave attenuation in underwater granular media.
Authors: Chotiros Nicholas P. , Isakson Marcia J.
Published: October 2010

Compared to compressional waves, there are relatively few measurements of shear waves in underwater sediments. In sandy sediments, shear waves are known to be slow and lossy compared to compressional waves. They contribute to the reflection loss at the water-sediment interface and they are known to determine the optimum propagation frequency in a shallow water wave guide. The frequency dependence of shear attenuation is known to be proportional to frequency, while that of the compressional wave is proportional to the square of frequency. This difference suggests that they are subject to different loss mechanisms. The linear frequency dependence is likely associated with solid contact losses, while the squared frequency dependence is indicative of viscous losses associated with the pore fluid. A broad band model of shear wave propagation should include both loss mechanisms. The frequency at which one mechanism gives way to an other may be estimated from existing models and measurements. [Work supported by the Office of Naval Research, Ocean Acoustics Program.]
Combination tones in pre-Columbian and contemporary double flutes.
Authors: Rawcliffe Susan
Published: October 2010

Long before Europeans identified the phenomenon of combination tones as “Tartini” tones, multiple flutes from preclassic and classic prehispanic cultures were being made that produce strong heterodyne/combination tones. Beats, buzzes, and heterodyne tones are produced through an interaction between two source tones. These phenomena exist on a continum of increasing separation in the frequency of the source tones. Many prehispanic double and some triple whistles and ocarinas survive, as well as double, some triple, and occasionally quadruple tubular flutes. Sets of flutes were probably also designed to produce these tones when played together. As will be demonstrated by sound samples of flutes and their sonograms from several prehispanic cultures, the ancient instrument builders utilized these phenomena in a variety of fascinating ways. Depending on the individual flute, air pressure variations in conjunction with the use of any available finger holes can be used to produce a range of beats, buzzes, rising and falling difference tones, and combination tone melodies. A sonogram and the sounds of a Rawcliffe double combination flute will further illustrate combination tones and their melodies. Time permitting, the presentation will conclude with a brief performance on the same flute.