bluuebunny/crossref_metadata_2025 huggingface dataset:
Hepatic vasculature flow monitoring through the ultrasonic interruption of contrast agents
Authors: Rhee R. T. , Fowlkes J. B. , Rubin J. M. , Carson P. L.
Published: July 2012

A technique has been developed to assess differential liver perfusion due to hepatic artery versus portal vein. This technique uses a continuous intravenous infusion of contrast agent and transcutaneous ultrasound to control the flow of contrast agent in selected vessels. A 1-in diameter transducer controlled the flow of perfluorocarbon-filled contrast agent in the abdominal aorta. It has been previously shown that acoustic amplitudes required to manipulate contrast agent are well within FDA limits. Pulsed ultrasound 10-cycle tonebursts at 2.25 MHz and 2.7% duty cycle were applied to the abdominal aorta for 30 sec to clear contrast agent from the hepatic artery, portal vein and liver parenchyma. The interruption transducer was then turned off, creating a sharp leading edge of contrast agent, with risetimes of approximately 0.1 sec measured in the hepatic artery. Harmonic grayscale and spectral Doppler images were obtained on a Toshiba Powervision 8000. The sharp contrast agent leading edge allows differential measurements of hepatic artery and portal vein arrival times. This time differential was measured at approximately 10 sec. These results indicate that image-guided insonification of the abdominal aorta may substantially improve discrimination of the hepatic arterial and portal venous supply in hepatic imaging.
Volume flow measurement using Doppler RF decorrelation
Authors: Tuthill Theresa A. , Fowlkes J. Brian , Rubin Jonathan M. , Hall Anne L.
Published: July 2012

A technique for monitoring volumetric flow was developed by combining standard Doppler measurements with amplitude decorrelation. The proposed algorithm uses current clinical ultrasound technology and is independent of vessel orientation. However, to accurately measure the ultrasound signals’ temporal decorrelation in blood, a contrast agent was added to increase the scattering signal strength. The algorithm uses standard frequency shifted Doppler signals, steered at two different angles, to determine the in-plane velocities. For the same firing lines, the corresponding temporal amplitude decorrelation is related to total velocity. The velocity normal to the imaging plane can then be extracted and integrated to compute total volume flow. The algorithm was tested in a flow tank using a GE Logiq 700 clinical scanner with a 7.5 MHz linear array. The output power was set at the lowest level to reduce effects from additional decorrelation due to acoustic radiation force. RF data were collected as in-phase and quadrature components. The tube flow results showed a parabolic velocity profile that scaled with volume flow. For transducer angles close to perpendicular to the vessel, the average calculated volume flow error was less than 25%. These preliminary results demonstrate the efficacy of our technique in measuring blood volume flow.
Equivalent circuit representation of Poisson-coupled electroacoustic modes
Authors: Smith R. Lowell
Published: July 2012

Mode-coupling phenomena have long been recognized as contributing to the detailed frequency-dependent behavior of electroacoustic materials and structures. Coupled modes have the effect of repelling each other from the positions in frequency space that they would occupy if the coupling interaction were not operative. Today, mode-coupling effects are most often addressed using finite-element analysis methods. This paper reinvestigates the Poisson (modes coupled via shared inertial mass) mode-coupling issue for simple electroacoustic structures from another perspective. Beginning with the piezoelectric equations of state, equivalent-circuit models are developed. These are useful for interpreting standard forms of impedance/admittance spectroscopy data for a variety of practical experimental geometries. The models are applied to measurement data and their usefulness for performance simulations is illustrated.
Relation between noise emission, friction and rolling resistance of car tires—Summary of final results
Authors: Sandberg Ulf S. I. , Ejsmont Jerzy A.
Published: July 2012

A European Union directive with noise emission limits for noise from vehicle tires is underway. It has been suggested that tires should meet certain limits also with regard to skid resistance, assuming a conflict between requirements for low noise and high friction. It is important to study whether there is any conflict between requirements with respect to noise emission from tires and other important performance characteristics. Therefore, the authors conducted an extensive experimental study on this topic in 1997–1999. Tire/road noise was measured with the CPX method on three road surfaces (8 mm medium-textured SMA, 16 mm smooth-textured DAC, and 12 mm rough-textured surface dressing) at two speeds. Noise was also measured with the drum method on three drum surfaces (replica ISO10844 surface, smooth-textured surface and rough-textured surface). Friction was measured with friction tester BV12, using optimum slip and locked wheel, at 70 km/h, on a smooth-textured DAC. Rolling resistance was measured on a laboratory drum facility, using a smooth safety walk surface and a rough-textured surface. Correlations, for 75 to 100 car tires, between all measured variables have been calculated. It is shown that there is no significant conflict between low noise and high friction, nor between low noise and low rolling resistance.
Multi-modal surveys of fin whales in the Sea of Cortez, Mexico
Authors: Clark Christopher W. , Croll Don A. , Acevedo Alejandro , Urban-Ramirez Jorge
Published: July 2012

A population of fin whales (Balaenoptera physalus), resident to the Gulf of California, Mexico, was studied over two seasons using an integrated approach. Systematic vessel-based visual survey and photo-ID efforts were conducted every 5–7 days to independently estimate the number and distribution of whales within a 10×30 mi2 study area. Some whales were tagged with time-depth-recorders. Sets of 5–6 distributed autonomous seafloor acoustic recorders, operating continuously during each season’s research period, were used to detect, locate, and track vocalizing whales. A 16-element towed array tracked individual vocal whales in real-time concurrently with visual observations, allowing biopsy samples of known vocal animals. Active acoustics was used to collect data on the density and distribution of krill so as to place measured variation in whale numbers, distribution, and behavior within an ecological context. The primary whale activity was feeding. Whale feeding patterns and survey distribution followed prey distribution. Vocal whale distribution followed dial feeding patterns and prey distribution. All biopsied vocal animals were males. Numbers of whales estimated by vessel survey, photo-ID, and passive acoustics were correlated. Results suggest that under certain conditions, vocal activity is a reliable measure of distribution and relative abundance. [Work supported by ONR.]
Locating and enumerating endangered humpback whales in the eastern Caribbean with directional (DIFAR) sonobuoys
Authors: Swartz Steven , Clapham Phillip , Cole Tim , Barlow Jay , McDonald Mark , Oleson Erin , Hildebrand John
Published: July 2012

Visual surveys for whales are limited by visibility conditions at sea and whale diving behavior. Male humpback whales (Megaptera novaeangliae) sing complex ‘‘songs’’ during their winter breeding season, and low-frequency song components lend themselves to acoustic detection. A visual and acoustic survey was conducted during the peak breeding season to evaluate passive acoustic methods to assess endangered humpback whales in the Eastern Caribbean, a region where they were exploited to depletion. Directional (DIFAR) sonobuoys were used to detect singing whales. Bearing angles from the sonobuoys to singing whales were calculated in real-time, and locations determined by crossing two or more bearings. A total of 4331 km of ‘‘on-effort’’ visual survey resulted in 9 whale sightings compared to 74 acoustic detections. Acoustic detections from 350 h of monitoring resulted in an abundance estimate of 116 (95% CI: 72 293) whales in February, and 123 (95% CI: 77 313) whales in March. The paucity of visual sightings is attributed to the prevailing high winds and the brief periods that whales spend at the surface. These results demonstrate the advantage of acoustic survey methods over visual methods for detecting whales in areas with poor visibility conditions such as the Eastern Caribbean.
Simple methods for locating, counting, and tracking sperm whales underwater in three dimensions
Authors: McGehee Duncan E. , Hildebrand John A.
Published: July 2012

Sperm whales’ clicks can be used to locate source animals in three dimensions using volumetric arrays of four or more sonobuoys. Clicks are detected using algorithms that depend on gender. Males are easier to track: their clicks are loud and distinctive, and there is usually only one male nearby. A simple envelope detector suffices. Females click more quietly, at a higher rate, and they often occur in large groups, resulting in a continuous barrage of clicks. Using a single click as a matched filter detector enhances the detection of clicks coming from a particular animal. Arrival time differences between the different sonobuoys are used to locate the source in three dimensions. A particular animal can be tracked by determining its location over time. An estimate of the number of phonating animals can be made by locating all the source animals during a particular time interval. During NOAAs SWAPS97 sperm whale survey, arrays of four sonobuoys each were placed around groups of diving sperm whales. Clicks from the animals were recorded at 48ksamples/s. Analyses of these data will be presented. [Work sponsored by NOAA and ONR.]
A unique way of sound production in the snapping shrimp (Alpheus heterochaelis)
Authors: Schmitz Barbara , Versluis Michel , von der Heydt Anna , Lohse Detlef
Published: July 2012

Sound production is known in more than 50, mostly stridulating, crustacean genera. These acoustic signals occur in agonistic interactions as well as for mate attraction. The mechanism of sound production in snapping shrimp, which also serves to stun or even kill small prey, is especially interesting. The current assumption was that the sound is produced by cocking and then rapidly closing the enlarged modified snapper claw. Snapping shrimp sounds contribute most to coastal biological noise, may be heard up to 1 mile away, and resemble the crackling of dry twigs in fire or the sizzle of frying fat. Recent hydrophone measurements close to tethered shrimp (Alpheus heterochaelis) revealed pulse-like signals of 500-ns duration, comprising frequencies beyond 200 kHz, and showing enormous sound pressure levels of up to 220-dB re 1 μPa (peak to peak) at 1-m distance. Such high intensities are very unlikely to be produced by the mechanical contact of two claw surfaces. Ultra-high-speed video recordings and simultaneous hydrophone measurements reveal that claw closure results in a water jet, the high velocity of which (25 m/s) leads to the formation of a cavitation bubble, which emits the extremely loud sound upon its collapse.
Development of a miniature thermoacoustic refrigerator
Authors: Hofler Thomas J. , Adeff Jay A.
Published: July 2012

A miniature thermoacoustic refrigerator is being developed for the purpose of cooling integrated circuits below their failure temperature in hot environments. Work has been done on a piezoelectric acoustic driver operating at 4 kHz. A simple refrigerator has been built and tested that uses one atmosphere of air as a working medium, is less than 3 in long, and has produced 12°C of cooling. A more advanced pressurized refrigerator is under development. [Work supported by DARPA and Rockwell Science Center.]
A large solar/heat-driven thermoacoustic cooler
Authors: Chen Reh-lin , Garrett Steven L.
Published: July 2012

Based on the success of an earlier solar-powered thermoacoustics prime mover which used a direct-illumination stack and no hot-side heat exchanger [Chen and Garrett, Proc. 16th Int. Cong. Acoust., Vol. II, 813–814 (1998)], a large solar/heat-driven thermoacoustic cooler was designed and fabricated. Target cooling powers of 10 to 60 W, over a 25-deg temperature span, were based on a thermal input power of 150 to 600 W. To concentrate the required amount of solar power on an 11-cm-diameter ceramic stack, a 10-ft diameter fiberglass parabolic dish, used for satellite TV, has been converted by gluing aluminized MylarTM on its surface over a 2-m diameter. A two-axis coordinated solar tracking system, driven by two computer-controlled motors, has produced the required 600 W of solar power to illuminate the hot side of the stack for a maximum of 3 h. Measured performance of the solar refrigerator will be compared to DeltaE models. [Work supported by the Office of Naval Research.]
On the use of the Sturm sequence to evaluate modal density
Authors: Shorter Philip J.
Published: July 2012

The modal density of a structural-acoustic subsystem is usually obtained analytically by considering the dispersion of various propagating wave types. Closed form expressions are available for the modal densities of simple beams, plates and shells (with curvature in one or two directions). However, one is often interested in subsystems with complex geometry which may possess inhomogeneous material and physical properties. The classical asymptotic formulations are not appropriate for such subsystems and numerical methods are often adopted. The most straightforward approach is to perform a finite-element-based modal analysis and count the number of eigenvalues that fall within various frequency bands. However, the computational expense associated with solving the full eigenproblem is often prohibitive. Significant computational savings can be made by employing the Sturm sequence property to evaluate the modal density. This paper describes the approach in more detail and provides a numerical example.
Sonar performance in turbid and bubbly environments
Authors: Richards Simon D. , Leighton Timothy G.
Published: July 2012

The presence of both solid particles and gas bubbles in coastal waters may have a significant effect on sound propagation, particularly at high frequencies, and may therefore be partially responsible for the observed variability in high-frequency sonar performance in shallow waters. Suspended particles increase volume attenuation through the processes of thermo-viscous absorption and scattering. Microbubbles also attenuate sound through viscous and thermal dissipation and scattering. The presence of microbubbles in the water column may also modify the sound speed and result in a dispersive medium. The effects of dilute suspensions of solid particles on the sound speed may generally be neglected for practical sonar applications. Algorithms for estimating the acoustic absorption coefficient and the speed of sound in water containing suspensions of fine mineral particles and populations of microbubbles have been developed. These have been incorporated into a sonar performance prediction model based on the ray method. Results are presented from this model which show that, for populations of suspended mineral particles and microbubbles which are commonly encountered in shallow-water environments, the predicted sonar performance is significantly different from that predicted when the additional effects are not taken into account. []
Geographically distributed acoustical monitoring of migrating birds
Authors: Mills Harold
Published: July 2012

A geographically distributed system of largely automated acoustical monitoring stations was developed to monitor the migration of small passerine birds. The targeted species, wood warblers and sparrows, migrate at night and produce short (about 120 ms at the longest), high-pitched (between roughly 6 and 10 kHz) calls as they fly, often audible from the ground. The monitoring system consisted of ten stations, located in New Jersey, Maryland, Pennsylvania, and New York, and a central data repository located at the Cornell Lab of Ornithology. Each station comprised an outdoor, weatherproof microphone connected to a volunteer’s home computer. The computer automatically ran acoustic transient detection software each night that listened continuously to the microphone signal and extracted all transients matching certain criteria, each to its own audio file. The detected transients were uploaded by the volunteer each morning via the Internet to the central repository for classification and archival. Results, including spectrograms of all detected transients and bird call counts, were displayed on a public web site updated daily. [Work supported by EPA.]
A compact digital recording system for autonomous bioacoustic monitoring
Authors: Calupca Thomas A. , Fristrup Kurt M. , Clark Christopher W.
Published: July 2012

A digital recording system with several months’ endurance was designed to monitor biological sounds. The system combined off-the-shelf components (microcontrollers, notebook hard drives) with an IDE interface designed for this purpose. The present storage capacity of the system is 25 Gbytes, with a maximum sampling rate of 50 kHz at a resolution of 12 bits. Power consumption during typical long-term continuous recording averages 250–500 milliwatts, and its physical dimensions are approximately 3×7×10 cm, including the hard drive. Hardware and software support for GPS clock synchronization enables correction of clock drift in the system, as well as precise synchronization of multiple units in an array. This unit has been extensively utilized in bottom-moored recording buoys, called ‘‘popups.’’ Popups include an acoustic communication system that receives and acknowledges commands from a surface support vessel, and an electronically controlled galvanic release. Everything is housed in a single 17-in. glass pressure housing. More recently, these systems have been used to monitor low-frequency terrestrial signals in studies of forest and savannah elephant populations. An upcoming project will drift these systems over inaccessible habitats using helium balloons.
Self-sustained oscillation as a function of blowing pressure in a one mass lip reed model
Authors: Moffitt Jeffrey , Ludwigsen Daniel
Published: July 2012

The lip reed is one of the least understood components of the brass player-instrument system. Its biological nature allows a wide range of player control over the parameters that are vital to the regeneration of the pressure wave. The lumped-element model of Adachi and Sato [J. Acoust. Soc. Am. 99, 1200–1209 (1996)] accomplishes the goal of self-oscillation and displays the correct phase relationships with only two parameters to specify. The embouchure is captured by the natural frequency of the mass-spring system, and the blowing pressure must be sufficient to develop self-sustained oscillation. Now the behavior of a model player-instrument system can be compared with the results of the thorough investigation of blowing pressure of human players by Fletcher and Tarnopolsky [J. Acoust. Soc. Am. 105, 874–881 (1999)]. The 2D parameter space is mapped to show regions of self-sustained oscillation as a function of both blowing pressure and the lip’s natural frequency. [Work supported by REU Grant NSF-DMR No. 9987850.]
A simple procedure for the selection of singular values in acoustical travel-time tomography
Authors: Bassrei Amin
Published: July 2012

Inverse problems are in general ill-posed and acoustical travel-time tomography is no exception. One common way to solve this inverse problem is by the generalized inversion through singular value decomposition (SVD). The crucial issue that arises here is the avoidance of small singular values which generally contribute to inconsistent solutions. The selection of singular values used in this work is based on the analysis of some estimators: rms error between the true model and the estimated one, rms error between the observed data and the calculated one, rms of the model, and the entropy of the model. In a numerical simulation with N equations and N unknowns, the SVD technique provides N singular values; N inverse matrices are then built, using 1,2,..., up to all the N singular values. For each matrix one solution is obtained and the above estimators are calculated and analyzed. Solutions with, for instance, small rms error are accepted, and solutions with inconsistent entropies are rejected. Thus, it is possible to determine the number of singular values related to the solution with a maximum of information and a minimum of discrepancy. [Work supported by CNPq and PGS.]
Greens function modified spherical harmonic analysis
Authors: MacGillivray John , Sparrow Victor
Published: July 2012

In an effort to develop a model for impulse responses in reverberant spaces, techniques for source radiation modeling were explored. One method of decomposing a source field involves an analysis based on spherical harmonic expansions of the field. Spherical harmonic decomposition is a powerful tool, but it is a mathematical approach which is completely independent of the type of source that creates the field. As a consequence, this technique cannot, in general, be used to gain insight into the underlying physical nature of the source. Moreover, because this is a brute force technique, the analysis can be extremely inefficient. In this presentation, modifications to the spherical harmonic analysis and synthesis equations are made to overcome these problems. These are based on the (theoretical) source types associated with the individual spherical harmonic expansion coefficients, and determination of the location of the primary components of the (physical) source. These modifications can greatly improve the computational efficiency and allow the source to be described by a set of coefficients which is more directly related to actual source types present. Illustrative examples will be shown. [Work supported by the Applied Research Laboratory at Penn State University under the Educational and Foundational Research Program.]
Stress measurement using critically refracted longitudinal waves
Authors: Bray Don E.
Published: July 2012

Critically refracted longitudinal waves (LCR) offer considerable advantage in achieving nondestructive measurement of stress in engineering components and structures. Of the various wave types, the LCR is most sensitive to stress and least sensitive to texture. Also called the surface skimming longitudinal wave (SSLW), the LCR wave skims along just underneath the surface of a material at bulk wave speeds. Further, there is experimental evidence of frequency dependence where the higher frequencies concentrate their energy more toward the surface than do the lower frequency waves. This effect, which has not been fully explained theoretically, enables stress gradient evaluation below the surface. Further, there is observed scatter in the travel times found at low stress levels. At higher levels of stress, the data often become very linear. The specific nature of the LCR wave, acoustoelasticity, LCR probes and beam patterns, and methods for its excitation, are described. Probe frequencies from 1.0 to 20 MHz have been used. System travel-time repeatability is 0.01% or better and stress resolution of ±21 MPa (3 ksi) is obtained in typical applications. Tests showing the frequency dependence and the scatter observed at low levels will be discussed. Several applications described using the LCR technique for stress measurements include pressure vessels, piping, welds, titanium turbine engine blades, aluminum plate and a turbine rotor. Future applications and research using the technique will be mentioned.
Specific learning phenomena in pitch discrimination tasks
Authors: Demany Laurent , Semal Catherine , Maubaret Cecilia , Noblia Odon
Published: July 2012

In experiment 1, listeners were trained to discriminate, during ten sessions (11000 trials), a monaural pure tone of 3000 Hz from ipsilateral pure tones with slightly different frequencies. The resulting perceptual learning (improvement in thresholds) appeared to be frequency-specific since, in subsequent sessions, a new learning was observed when the tone frequencies were shifted to a range near 1200 Hz, or 6500 Hz. By contrast, when the initial tones were subsequently presented to the contralateral ear, thresholds were immediately optimal; therefore, the initial learning was not ear-specific. In experiment 2, performed on different listeners, training in pitch discrimination was initially provided using complex tones that consisted of harmonics 3–7 of a missing fundamental (near 100 Hz for some listeners, 500 Hz for others). After 11000 training trials, these complex tone stimuli were replaced by pure tones with frequencies that compared to the complex tone frequencies in three different ways: (condition A) their missing fundamentals, (condition B) their median spectral components, (condition C) completely novel. New learning was observed in each condition. Although the pure tones used in condition A were matched in pitch to the previously presented complexes, there was no evidence for a transfer of learning in this condition.
The upper boundary of temporal pitch coding inferred from electric stimulation of the auditory nerve
Authors: Zeng Fan-Gang
Published: July 2012

Pitch can be represented by a vibration pattern in the cochlear partition (the place code) or a spike firing pattern in the auditory nerve (the temporal code). Generally, pitch of high tones has been assumed to use the place code and pitch of low tones to use the temporal code. Because both place and temporal codes change simultaneously as a function of frequency in a normal auditory system, the upper boundary of temporal encoding of pitch is not clear. Here, we use cochlear implants to infer the upper boundary of temporal pitch coding. The cochlear implant electrically stimulates the auditory nerve, providing an opportunity to assess the temporal pitch coding independent of the place code. Cochlear implant listeners performed pitch estimate and discrimination tasks as a function of frequency. The tasks were performed when either basal or apical electrodes were stimulated independently. The results showed that pitch is monotonically increased for frequencies lower than 300 Hz and does not change for frequencies above that. Pitch discrimination results also showed a break point at about 300 Hz. Together, the present results suggest that 300 Hz is the upper boundary of temporal pitch coding.
Testing a new theory of Fechnerian scaling: The case of auditory intensity discrimination
Authors: Zimmer Karin , Colonius Hans
Published: July 2012

Recently, Dzhafarov and Colonius [Psych. Bull. Rev. 6, 239–268 (1999)] proposed a theory that refines Fechner’s ideas of deriving the perceptual distance between stimuli from discriminability measures in the unidimensional case, and generalizes it to multidimensional stimulus domains. In this theory, the existence of a Fechnerian metric is based on testable assumptions about the shape of the psychometric functions. Thus, ‘‘same–different’’ psychometric functions: (1) have to attain a single minimum at some point; (2) must be locally symmetric around their minima; and (3) must show the same degree of cuspidness, or flatness, along the stimulus continuum. In the present investigation, these assumptions were evaluated experimentally employing acoustical stimuli differing along one stimulus dimension, intensity. In a ‘‘same–different’’ paradigm, using the method of constant stimuli, seven 1000-Hz tones deviating from 0 to 2.1 dB with respect to a standard were presented diotically to five observers. For every participant, psychometric functions were assessed at five hearing levels ranging from 40- to 80-dB HL. The outcome of the ‘‘same–different’’ experiments generally supports the assumptions of the theory. Therefore, data such as these are, in principle, suitable for deriving a global loudness scale from local discriminability measures.
Intensity-difference limens as a function of level for Gaussian-shaped packets of broadband noise
Authors: Nizami Lance , Jesteadt Walt , Reimer Jason F.
Published: July 2012

The nonmonotonic level dependence of the intensity-difference limens (DLs) for a broadband stimulus, the click [Raab and Taub, J. Acoust. Soc. Am. 46, 965–968 (1969)], can also be observed for a narrow-band stimulus, a Gaussian-enveloped 2-kHz tone [Nizami et al., J. Acoust. Soc. Am. 107, 2820 (2000)]. Although the DLs for a 32-ms tone-pip (measured at plus-or-minus four standard deviations) follow the near-miss to Webers Law, the DLs for a 4-ms tone-pip are higher, and peak mid-level, at about 40 dB SL. The DLs are larger still, and the peak more profound, for an 8-ms tone-pip. Energy-spectrum changes as duration changes, however, obscuring the source of the nonmonotonicity. Energy-spectrum was therefore held constant by using a broadband-noise carrier, while preserving the Gaussian envelope. DLs were obtained from two male and four female subjects (ages 18–40) having normal hearing thresholds. Noise-pip levels ranged from 30 to 90 dB SPL at durations of 2, 4, 8, 16, and 32 ms. Although the across-subject mean DLs for 32- and 16-ms noise-pips follow Weber’s law, the means for 8 ms are larger, and rise at mid-level. The DLs grow as duration decreases from 8 to 4 to 2 ms, unlike the case for 2 kHz.
Audiometric tests with various earphones, signals of short duration, and signals of high frequencies
Authors: Lydolf Morten , Karlsen Brian L. , Santilln Arturo O. , Moller Henrik
Published: July 2012

New audiometric standards on tests with signals of short duration and tests with signals of high frequencies are in the course of preparation by the international working group, ISO/TC 43/WG 1. In this paper, first a selection of five different audiometric earphones is compared by using threshold measurements in the frequency range 125 Hz–8 kHz. These data are meant for validation of the reference threshold for regular pure-tone audiometry given in ISO 389-1. Signals of short duration consist of either rectangular pulses with a duration of 0.1 ms or pure-tone samples including only seven full periods at the octave band frequencies 0.5–8 kHz. Both single events and bursts with 20 signals presented during 1 s were used as stimuli. Finally, audiometric tests were made with high-frequency pure-tone stimuli of 1-s duration at: 8, 9, 10, 11.2, 12.5, 14, and 16 kHz. For all experiments the chosen psychophysical method was the ascending method, implemented nearly as described in ISO 8253-1. Twenty-five young, otologically normal test subjects were used in order to contribute to the standardization work. Representative reference thresholds are needed with respect to signals of short duration for the coming ISO 389-6 and for high frequencies in ISO 389-5.
Finite-element method for determination of electromechanical coupling coefficient for piezoelectric and capacitive micromachined ultrasonic transducers
Authors: Fraser John D. , Reynolds Paul
Published: July 2012

Research has been reported on ultrasonic transduction using capacitive micromachined ultrasonic transducers (cMUTs). These are thin membranes of diameters of order 100 μm suspended over a silicon substrate with a gap of order 1 μm. The devices become transducers when a bias voltage is applied, causing electrostatic forces to draw the membranes closer to the substrate. Operation is by applying an electrical signal to excite a vibration in the membrane, or an ultrasonic wave to excite an ac voltage. The cMUT literature includes equivalent circuits and studies on losses and spurious waves, and discussion of output, sensitivity, and bandwidth. Transducer engineers would like to see analogies to the properties of piezoelectric transducers, in order to compare the cMUTs to traditional designs. An extension of the definition of the piezoelectric coupling coefficient to an effective coupling coefficient for cMUT transducers is presented. A finite-element method has been developed using the pzflex software package to determine its value, and used for evaluating cMUT designs. Accepted values are obtained for a piezoelectric material, and it is shown that an ideal cMUT has a coupling coefficient approaching 1 at the membrane collapse voltage, while practical devices are often limited to lower values.
Intelligibility of words by reduced-channel and cochlear implant listeners
Authors: Carrell Thomas , Shapley Kathy , Koch Dawn
Published: July 2012

One of the differences in the way a cochlear implant, versus a normally functioning cochlea, delivers sound to auditory centers in the brain is that the implant delivers far fewer channels of frequency information. This experiment investigated how speech intelligibility in cochlear implant (CI) listeners may be related to reduced frequency resolution as opposed to other factors. A closed-response set Modified Rhyme Test was administered to CI listeners [House et al., J. Acoust. Soc. Am. 37 (1965)] using naturally produced words spoken in isolation. The stimuli presented to the listeners without implants were a reduced-channel (RC) version of exactly the same stimuli, which were filtered and processed in software in a manner similar to the hardware method described by Shannon [Science 270 (1995)]. The effect of the signal processing was to divide each word into four frequency channels for the first group of RC listeners and into eight channels for the second group. Results indicated final phoneme identification appeared to be more difficult for CI listeners than for four- or eight-band RC listeners. Results were more variable for phonemes in the initial position. CI listeners were better than both four- and eight-band listeners on affricates and better than eight-band listeners for weak fricatives.
Cochlear-implant effects on sibilants
Authors: Matthies Melanie , Perkell Joseph , Vick Jennell , Zandipour Majid
Published: July 2012

To investigate sibilant production in persons with cochlear implants, this study compared acoustic analyses of three sets of speech stimuli: (1) A synthesized continuum of sibilants from ‘‘sa’’ to ‘‘sha’’ was constructed where the characteristics of the sibilant noise as well as the F2 and F3 transitions were varied. (2) The synthesized speech was processed through a cochlear-implant simulation program [cf. Shannon et al., Science 270, 303–304 (1995)]. (3) Longitudinally recorded speech production data were obtained from postlingually deafened adults who had received cochlear implants. Formant frequencies and amplitudes were obtained at midvowel and at vowel onset. The amplitude in the sibilant at the frequency of vowel F3 was measured, as were the spectral characteristics of the sibilant noise for all three sets of speech. Judgments about the identity and quality of the synthesized sibilants were obtained from a panel of listeners with normal hearing. Correlations indicated that the selected acoustic cues were highly salient to the listeners for both the unprocessed and cochlear-implant-simulated sibilants. Observed improvements in sibilant production from pre-CI to post-CI are likely due to the improved availability of these salient cues for auditory feedback and self-monitoring. [Work supported by MURI Z883402, NIDCD DC03007.]
Gross versus detailed spectral cues in spectrally distorted speech
Authors: Kiefte Michael
Published: July 2012

An experiment was designed to address the importance of gross versus detailed spectral features in the perception of stop-consonant place of articulation. This is done by presenting listeners with noise-vocoded stimuli in which the spectral resolution of the processed tokens is severely reduced. While it can be assumed that such a transformation will effectively eliminate detailed spectral features, such as burst-peak and formant frequencies, previous work has shown that this is not necessarily the case [M. Kiefte, J. Acoust. Soc. Am. 106, 2273 (1999)]. Nevertheless, it is shown that gross spectral features, such as spectral tilt and compactness, are better able to predict listeners’ identifications of noise-vocoded stimuli. While this suggests that listeners attend primarily to gross spectral shape cues, which are assumed to be particularly robust in such distorted speech, it is also shown that detailed spectral features are better able to model listeners’ responses to undistorted speech. [Work supported by SSHRC.] a)Now at the Dept. of Psychology, University of Wisconsin, Madison, WI.
Long-term speech spectra for Taiwanese–Mandarin bilingual speakers
Authors: Tseng Chin-Hsing , Chen Hsiao-Chuan , Chen Shing
Published: July 2012

Audiologists and otologists in Taiwan often rely on the English normative data during the hearing aid gain prescription procedure. Thus, the measurement of long-term speech spectra for both Taiwanese and Mandarin was deemed a research priority. Average long-term rms 1/3-octave band speech spectra were generated for 60 Taiwanese–Mandarin speakers with gender, age, and language as the three major independent variables. The male speakers differed significantly from the females in the low-frequency bands across languages and age groups. However, a slight gender difference in the 4000-Hz band was observed only in the young adult group (aged 20–40) for both Taiwanese and Mandarin. Across languages, the elder group (aged 41–60) differed significantly in the low-frequency bands for both genders, but high-frequency differences existed only in the females. The overall spectra will provide a useful reference for hearing aid gain prescription in the Chinese-speaking communities. Cross-language comparisons can also shed insight on the speech communication processes. [Work supported by the Council of Labor Affairs.]
Effects of lexical and talker variability factors on spoken word recognition in native and non-native, normal and hearing-impaired listeners
Authors: Takayanagi Sumiko , Dirks Donald D. , Moshfegh Anahita , Noffsinger P. Douglas , Fausti Stephen A.
Published: July 2012

The effects of lexical difficulty and talker variability on word recognition were examined in four groups of listeners: native English/normal hearing; native English/hearing impaired; non-native English/normal hearing; and non-native English/hearing impaired (hearing level matched to the native hearing impaired). Lexical difficulty was measured by the difference in performance to 75 lexically ‘‘easy’’ and ‘‘hard’’ words based on word frequency and Neighborhood Activation Theory [Luce and Pisoni (1998)]. The effect of talker variability was measured by the difference in performance between single and multiple talker (nine talkers) conditions. The familiarity of the 150 words was rated on a seven-point scale. An up–down adaptive procedure was used to determine the sound pressure level for 50% performance. Non-native listeners in both normal and hearing-impaired groups required a greater intensity for equal intelligibility than for the comparative native normal and hearing-impaired listeners. Results, however, showed significant effects of lexical difficulty and talker variability in all four groups. Structural equation modeling demonstrated that an auditory factor estimated by pure tone average, etc., accounts for four times more variance to performance than does a linguistic fluency factor measured by word familiarity ratings and native versus non-native status, however, the linguistic fluency factor is also essential to the model fit.
Effects of filtering preceding noise on the intelligibility of voiced plosives
Authors: Ainsworth William A. , Cervera Teresa
Published: July 2012

The intelligibility of voiced plosives in continuous noise is greater than in noise which is coincident with the speech [W. A. Ainsworth and G. Meyer, J. Acoust. Soc. Am. 96, 687–694 (1994)]. It is suggested that this could be due to some adaptation process. In order to test this hypothesis, experiments have been performed to measure the intelligibility of voiced plosives in isolated syllables with different durations of preceding noise. The effects of the spectral shape of the noise have also been examined. It was found that, on average, 800 ms of band-stop filtered noise preceding the syllable increased the recognition rate of the plosives by 14%, band-pass filtered noise by 12%, low-pass filtered noise by 7%, but high-pass filtered noise by only 3% compared with co-gated noise. The time constant of the adaptation process was estimated to be 165 ms. [Work supported in part by EU TMR Network SPHEAR.]
Across-frequency timing in speech perception: Effective processing of contrasting temporal speech patterns requires proper synchronous alignment
Authors: Healy Eric W. , Bacon Sid P.
Published: July 2012

Although narrow-band temporal speech patterns produce virtually no intelligibility when presented individually, a contrasting pair of such patterns can produce appreciable intelligibility [E. W. Healy, Dissertation, University of Wisconsin–Milwaukee (1998)]. In the current study, the role of across-frequency timing in speech perception was investigated using these contrasting temporal speech patterns which allow the strict isolation of across-frequency integration. Across-band timing was investigated by varying the synchrony of paired temporal patterns. Individual temporal patterns were created by amplitude modulating tones at 750 and 3000 Hz to follow the temporal fluctuations of corresponding 1/3-octave sentences. The pair of individually unintelligible narrow-band temporal patterns were mixed and presented to listeners either in temporal synchrony, or with offsets of 12.5, 25, 50, or 100 ms. High intelligibility scores in the aligned condition were reduced when as little as 12.5 ms of misalignment was introduced, and performance fell to values near zero at the largest misalignment. These results indicate that normal-hearing young listeners are relying heavily on across-channel timing information provided by the synchronous temporal patterns. Further work employing this technique may provide insight into the role of timing in the suprathreshold speech processing deficits of elderly or hearing-impaired listeners. [Work supported by NIDCD.]
Effects of time reversal on consonant identification
Authors: Bharadwaj Sneha V. , Assmann Peter F.
Published: July 2012

Time-reversed speech has the same long-term amplitude spectrum as natural speech, but is largely unintelligible. To study the perceptual effects of disrupting the temporal integrity of formant transitions on consonant identification, CVC syllables were divided into brief, successive, nonoverlapping segments and each segment was time reversed. In experiment 1, CVC syllables from two talkers were time reversed using 20, 40, 60, 80, or 100-ms temporal windows. The initial consonant was either /p/, /t/, /k/, /b/, /d/, /g/, /s/, /∫/, /f/, or /v/. The medial vowel was /i/, /u/, or /ʌ/, and the final consonant was /t/. Identification accuracy (12 listeners) showed a progressive decline from 99% for unmodified syllables to 63% for the 100-ms window. Time reversal was less disruptive when syllables were presented in a natural carrier sentence, dropping to 75% for a 100-ms time window. Experiment 2 replicated this pattern of findings with a larger sample of ten talkers. The effects of time reversal were not uniform across the 12 consonants, but showed highly disruptive effects on the identification of stops and relatively minor changes in fricatives. Overall, the results suggest that time reversal provides a useful method for quantifying the role of dynamic information in consonant perception.
Speech communication within the Acoustical Society of America
Authors: Harris Katherine , Ladefoged Peter , Stevens Kenneth
Published: July 2012

Speech communication has been an important field of the Acoustical Society from its earliest years. Harvey Fletcher, the first President of the Society, along with many colleagues at Bell Telephone Laboratories, provided new data and techniques for investigating speech. The invention of the sound spectrograph during World War II led to greater understanding of the physics of speech, and this in turn produced developments in speech synthesis, summarized through the mid 1980’s by Klatt [J. Acoust. Soc. Am. 82, 737–793 (1987)]. Physiological, psychological and linguistic aspects of speech have also been much investigated within the Society.
Broadband performance of a time-reversing array in shallow water
Authors: Sabra Karim G. , Dowling David R.
Published: July 2012

Future underwater communication systems may be developed from the spatial and temporal focusing properties of a time-reversing array (TRA). These properties ensue from the interactions between the underwater environment and the TRA. In a shallow-water sound channel, the influence of the TRA element number (6, 12, 20) and spacing (5 m, 3 m), and source-array range (1 to 40 km) on the spatial focusing and temporal compression will be presented. Here, the results show that compromises are necessary. Short ranges produce better spatial focusing while longer ranges produce better temporal compression. In addition, theoretical results for the performance of the TRA in an underwater environment with an ambient noise field has been evaluated for a family of pulses that span the range from narrow- to broadband. The theory, which has been the subject of previous presentations, accounts for the number of array elements and the time–bandwidth product of the signal. Based on this theory and numerical experiments, we will first investigate the effect of the sound channel characteristics on the performance of broadband signals and then compare broadband signals to narrow-band signals for several source-array ranges. [Work sponsored by the Office of Naval Research.]
Spatial resolution of time reversal arrays in shallow water
Authors: Kim Seongil , Edelmann Geoffrey F. , Hodgkiss William S. , Kuperman William A. , Song Hee Chun , Akal Tuncay
Published: July 2012

A series of time reversal (phase conjugation) experiments was performed in shallow water including a range-dependent slope environment. Time reversal arrays implemented with center frequencies of 445 and 3500 Hz achieved sharp focal regions up to ranges of 30 and 13 km, respectively, in 110–130 m shallow water. Analysis with an image method indicates that the focal size approaches the diffraction limit of an array for given waveguide conditions, i.e., waveguide geometries and attenuation. The measured focal size has implications for the maximum achievable resolution of matched field processing, which is a computational implementation of the phase conjugation process. [Work supported by ONR.]
Applying the time-reversal operator to detection in a shallow-water environment
Authors: Fromm David , Gaumond Charles F.
Published: July 2012

The decomposition of the time-reversal operator (DORT) is a method of isolating scatterers with a multiple-source/multiple-receiver system [C. Prada et al., J. Acoust. Soc. Am. 99, 2067–2076 (1996)]. A series of simulations is shown that elucidates the resolution of a DORT system in a reverberant, shallow-water environment. The system consists of vertical line arrays of sources and receivers in monostatic and vertically bistatic configurations. Two basic resolution problems are considered: the ability to discriminate two scatterers in the water column and the ability to detect a single scatterer in the presence of bottom clutter. The performance of the DORT system is shown relative to array configuration, location of scatterer in the water column, and the level and type of bottom clutter. [Work supported by ONR.]