bluuebunny/crossref_metadata_2025 huggingface dataset:
Dependence of dispersive properties of cancellous bone on bone mineral and collagen content
Authors: Waters K. R. , Hoffmeister B. K. , Whitten S. A.
Published: October 2013

Cancellous bone is known to exhibit dispersion over clinically-relevant ultrasonic frequencies. However, the dependence of the dispersive properties of cancellous bone on bone mineral and collagen content are currently not well understood. Cancellous bone specimens (n=16) obtained from the proximal end of bovine tibia were cut into cubes (15 mm sides) and prepared for measurement along the anterior-posterior (AP), medial-lateral (ML), and superior-inferior (SI) directions. Bone mineral densities (BMD) ranged from 100 to 250 kg/m3. Subsets of the bone specimens were chemically treated to remove bone mineral (nM=10) and collagen content (nC=6). Ultrasonic measurements were performed before and after treatment in a water bath (23°C) using a pair of 2.25-MHz planar transducers aligned coaxially. Phase velocity was determined from Fourier analysis of the ultrasonic signals, and dispersion was calculated as the change in phase velocity across the usable bandwidth. Demineralization produced highly significant changes in dispersion along the AP (p=0.004) and ML (p=0.002) directions, but not the SI direction (p=0.6). Decollagenization produced no significant change for any direction (p>0.3). These results indicate that bone mineral plays a much larger role than collagen in the dispersive behavior of cancellous bone.
Effects of emotion on different phoneme classes
Authors: Lee Chul Min , Yildirim Serdar , Bulut Murtaza , Busso Carlos , Kazemzadeh Abe , Lee Sungbok , Narayanan Shrikanth
Published: October 2013

This study investigates the effects of emotion on different phoneme classes using short-term spectral features. In the research on emotion in speech, most studies have focused on prosodic features of speech. In this study, based on the hypothesis that different emotions have varying effects on the properties of the different speech sounds, we investigate the usefulness of phoneme-class level acoustic modeling for automatic emotion classification. Hidden Markov models (HMM) based on short-term spectral features for five broad phonetic classes are used for this purpose using data obtained from recordings of two actresses. Each speaker produces 211 sentences with four different emotions (neutral, sad, angry, happy). Using the speech material we trained and compared the performances of two sets of HMM classifiers: a generic set of ‘‘emotional speech’’ HMMs (one for each emotion) and a set of broad phonetic-class based HMMs (vowel, glide, nasal, stop, fricative) for each emotion type considered. Comparison of classification results indicates that different phoneme classes were affected differently by emotional change and that the vowel sounds are the most important indicator of emotions in speech. Detailed results and their implications on the underlying speech articulation will be discussed.
Robust speaker recognition using glottal information-based cepstral mean subtraction
Authors: Yang Pu , Yang Yingchun , Wu Zhaohui
Published: October 2013

Channel distortion and background noise often severely degrade the performance of automatic speaker recognition (ASR) system. In this paper, a new compensation method called glottal information-based cepstral mean subtraction (GIBCMS), which improves upon the conventional cepstral mean subtraction (CMS) method, is presented. Besides the cepstral information, GIBCMS has utilized the speaker’s glottal information, which also holds speaker-dependent characteristics, but is less vulnerable to environment than the cepstral one. In order to test its robustness under channel distortion, even with high level of background noise, we applied this method to the SRMC corpus which is added by noise at different SNRs. The experimental results show that GIBCMS gains better performance over other improved CMS methods on it.
Enhanced energy redistribution speech intelligibility algorithm with real-time implementation
Authors: Raghavan Manasa , Skowronski Mark D. , Harris John G.
Published: October 2013

Speech intelligibility enhancement is a concern for mobile platforms operating in noisy environments. Current noise-reduction techniques, such as subspace methods and spectral subtraction, operate on speech corrupted by acoustic and transmission noise. Yet preprocessing techniques, which operate on clean speech before noise corruption, have received little attention. Previously, the authors have developed the energy redistribution algorithm [J. Acoust. Soc. Am. 112, 2305 (2002)], which, based on characteristics of clear speech as well as the Lombard effect, redistributes energy in time from voiced regions to unvoiced regions of speech. The algorithm is designed efficiently for real-time implementation, and in this work the algorithm is demonstrated on a mobile platform, TIs TMS320C6713 DSK board. Furthermore, two enhancements to the algorithm are introduced: (1) a variable unvoiced gain factor and (2) a high pass filter (HPF). The variable unvoiced gain factor adjusts the amount of energy redistributed, and the HPF, a compact algorithm shown previously to enhance clean speech in noisy environments [Niederjohn and Grotelueschen, IEEE Trans. Acoust., Speech, Signal Process. 24(5), 277–282 (1976)], is tested on sentences from the Hearing in Noise Test (HINT) corrupted by speech-shaped noise. Results show improved speech intelligibility for both enhancements.
Acoustical simulation based on head-tracked auralization and measured high-resolution head related transfer function (HRTF) and impulse responses
Authors: Ahnert Wolfgang , Feistel Stefan , Moldrzyk Christoph
Published: October 2013

A very desirable feature of modern acoustical simulation programs is the easy, fast and reliable auralization of the prediction results. In this paper we consider a new auralization method, based on a head-tracked headphone system with high spatial resolution and real-time convolution. We discuss the way to measure the directional head-related transfer functions, the calculation of the directional binaural impulse responses and the realization as a real-time convolution software. Furthermore, high-resolution impulse responses have been measured to compare reality, measurement and prediction results of an example room. The measurements were performed with a newly developed software tool EASERA. We conclude that, using this new method, auralization results are obtained equivalent to the human perception in reality.
Binaural simulation of virtual stage environments for evaluation by the solo violinist
Authors: Chu William
Published: October 2013

A unique computational and testing model for assessing solo violinists’ response to various acoustical conditions on-stage was developed and shown to be an effective study tool. Realtime binaural auralizations of CATT-Acoustic models of several different virtual stage-acoustic designs were examined with a small group of professional violinists. Perceived differences as expressed by the subjects were discussed and explored in light of the specific measured acoustical descriptors available through the model. Unlike acoustical descriptors developed for audience receivers, ease of playing, support, and other soloist concerns do not necessarily show strong correlation between different subjects. It was found that this binaural simulation technique could provide a basis for developing a personal contextual vocabulary, to better understand the desired acoustical response of individual musicians.
Expected Doppler shift and spread in long range scattering from fish schools on the continental shelf
Authors: Makris Nicholas C. , Lai Yisan , Bertsatos Ioannis , Symonds Deanelle , Lee Sunwoong , Ratilal Purnima
Published: October 2013

A stochastic model for the velocities of individual fish in a school, consistent with continous wide area observations [Symonds et al., J. Acoust. Soc. Am. 115, 2618 (2004)], is developed. This is combined with the model for scattering from a moving target submerged in a stratified ocean waveguide [Lai and Makris, J. Acoust. Soc. Am. 113, 223–244 (2003)] to estimate the Doppler shift and spread expected in long-range scattering from fish schools on the continental shelf. The spectral characteristics expected for randomly swarming and migrating fish schools are then compared with those for stationary seafloor features and other moving targets such as underwater vehicles.
Influence of the precision of spectral backscatter measurements on the estimation of scatterer size in cancellous bone
Authors: Jenson Frederic , Padilla Frederic , Laugier Pascal
Published: October 2013

Ultrasonic backscatter measurements can be used to characterize trabecular bone structure and to estimate trabecular thickness (Tb.Th, i.e., the size of the scatterers). Our objective was to evaluate the performance of this estimator as well as others spectral estimators, like the frequency dependence and the mid-band amplitude of the backscatter coefficient. The performance of these estimators is degraded mainly by two factors: interference noise due to random positioning of the scatterers and attenuation. We have simulated rf-lines backscattered from trabecular bone assuming a random positioning of the trabeculae (leading to a fully developed speckle), a Gaussian form factor for the scatterers and a linear-frequency dependent attenuation. It is found that the variance in the estimation of the frequency dependence of the backscatter coefficient is as high as the variance due the biological variability in Tb.Th, in agreement with the results showed by K. Wear [J. Acoust. Soc. Am. SA, 110 (2001)]. In contrast, the variances on the mid-band amplitude and on the estimated trabecular thickness are lower than the variance due the biological variability. We also show that the effect of attenuation may be compensated by using an appropriate attenuation-compensation function. These results suggest that the inverse problem can be appropriately addressed.
Assessment of cortical bone microstructure and material properties using high resolution scanning acoustic microscopy
Authors: Raum Kay , Chandelier Florent , Leguerney Ingrid , Talmant Maryline , Saied Amena , Laugier Pascal , Peyrin Françoise
Published: October 2013

Combined evaluation of bone microstructural and mechanical information remains a challenging task which is required for bone phenotyping or accurate finite element modeling. Our objective was to assess the value of quantitative scanning acoustic microscopy (SAM) for bone characterization in comparison to synchrotron radiation computed tomography (SR-CT). Ten specimens of human cortical bone (radius) were investigated using SR-CT and SAM (200 MHz) with spatial resolution of 10 and 8 μm, respectively. An image fusion and analysis software was developed to derive site-matched estimates of (1) microstructural parameters, e.g., haversian cavity density and mean diameter and porosity, and (2) tissue properties such as mineral density (MD, SR-CT) and acoustic impedance (Z, SAM) for distinct anatomical regions of interest (osteons, interstitial tissue). Local stiffness c33 was derived from the combination of MD and Z. An almost perfect correlation was found for all microstructural indices derived by both techniques. Impedance was correlated to the square of MD (R2=0.39, p<1e−4). The derived stiffness c33 (35.9±12.8) was highly correlated with Z (R2=0.99, p<1e−4). These findings suggest that SAM fulfills the requirement for a simultaneous evaluation of cortical bone microstructure and material properties at the tissue level.
Maximum effective aperture size for source direction estimates in a complex terrain
Authors: Moran Mark , Wilson D. Keith , Greenfield Roy
Published: October 2013

Source resolution and clutter suppression from ground sensor arrays scales directly with sensor array aperture. However, boundary layer complexities can disrupt acoustic and seismic wavefronts at a variety of spatial scales. Hence, usage of an array dimension that is larger than these wavefront distortions can lead to degraded resolution and poor clutter suppression. Using experimental data, we demonstrate the maximum effective aperture for determining a line of bearing to a light moving ground vehicle. Our data were collected in mountainous terrain with open fields, dense forests, and steep hills. We deployed a field of 36 seismic sensors (geophones) and 36 high quality microphones. These sensors were distributed in triangular subclusters within a larger equilateral triangle that was roughly 100 m on each side. Our microphone separations were as small as 10 cm, with common separations of 1, 5, 10, and 25 m. Our seismic subclusters had common separations of approximately 5, 15, and 30 m. We apply a high resolution 2-D maximum-likelihood (ML) wavenumber estimation method to analyze array gain characteristics at multiple aperture sizes. Our results show large angular and range dependencies that correlate strongly with terrain complexity and disruptions of line-of-sight with a moving vehicle.
Impedance and Brewster angle measurement for thick porous layers
Authors: Leary Del , Hickey Craig J. , Allard Jean F. , Henry Michel
Published: October 2013

For thin nonlocally reacting porous layers, a method derived from the work of Chien and Soroka [J. Sound Vib. 43, 9–20 (1975)] has been used to localize the pole of the reflection coefficient located at an angle close to grazing incidence and to measure the surface impedance at this angle. A very simple experimental setup is used to obtain measurements on materials with large flow resistivity, at frequencies larger than 1 kHz, using samples with areas on the order of 1 m2. This method is used in the present work to measure the surface impedance of acoustically thick porous layers of Ottawa sand and glass beads. There is good agreement between the measurements and predicted values. The method is also applied to study the effects of surface sealing. Sealing is modeled as a thin screen of higher flow resistivity at the surface. Reasonable agreement between the measured and predicted additional flow resistance is obtained.
Modal analysis of broadband acoustic signals propagating in the top layer of the ground
Authors: Fokin Vladimir N. , Sabatier James M. , Howard Wheeler B.
Published: October 2013

Experimental measurements of the frequency dependence of sound propagation losses in the top layers of the ground were performed in the frequency range between 80 and 420 Hz. These measurements revealed that the distribution of energy on the frequency-range plane has a regular structure. The features of this structure were modeled in the modal approach for a gradient model of the ground. It was shown that average sound speed in the ground and the approximate depth of the layers may be estimated through analysis of the experimentally measured frequency-range distribution of energy. The technique explores the waveguide properties of the top layer of the ground. According to the mode theory, two asymptotes coming from the point (0,0) should exist on the wave number-frequency plane. From the slope of these asymptotes, the minimum and maximal sound speeds in the waveguide were found to be 100 and 530 m/s. The approximate depth of the layers was estimated. Comparisons of obtained sound speeds and depth of layers with independently estimated values show satisfactory agreement. [Work supported by ONR Grant N00014-02-1-0878.]
Variation in acoustic to seismic signatures associated with natural depth variability of hardpans
Authors: Howard Wheeler B. , Hickey Craig J. , Sabatier James M. , DiCarlo David A.
Published: October 2013

Acoustic to seismic coupling techniques have been successfully used in the investigation of near-surface soils. Spatial variation in the soil profile and associated near-surface properties are manifested as differences in the acoustic to seismic transfer function. Two field sites were chosen which differ by the average depth of a naturally occurring hardpan. In this presentation, the measured acoustic to seismic response at multiple point locations within these two agricultural field sites is presented. Variability in attributes of the acoustic to seismic transfer function, namely the frequency, half-width, and amplitude of the resonance, are compared to variability of the depths to the hardpan.
Fifty years of progress in acoustic phonetics
Authors: Stevens Kenneth N.
Published: October 2013

Three events that occurred 50 or 60 years ago shaped the study of acoustic phonetics, and in the following few decades these events influenced research and applications in speech disorders, speech development, speech synthesis, speech recognition, and other subareas in speech communication. These events were: (1) the source-filter theory of speech production (Chiba and Kajiyama; Fant); (2) the development of the sound spectrograph and its interpretation (Potter, Kopp, and Green; Joos); and (3) the birth of research that related distinctive features to acoustic patterns (Jakobson, Fant, and Halle). Following these events there has been systematic exploration of the articulatory, acoustic, and perceptual bases of phonological categories, and some quantification of the sources of variability in the transformation of this phonological representation of speech into its acoustic manifestations. This effort has been enhanced by studies of how children acquire language in spite of this variability and by research on speech disorders. Gaps in our knowledge of this inherent variability in speech have limited the directions of applications such as synthesis and recognition of speech, and have led to the implementation of data-driven techniques rather than theoretical principles. Some examples of advances in our knowledge, and limitations of this knowledge, are reviewed.
Fifty years of progress in speech analysis (LPC)
Authors: Itakura Fumitada
Published: October 2013

Although the history of statistical linear prediction is very long, it was first applied to speech analysis in 1966 at NTT to estimate the all-pole speech spectrum envelope in order to implement ASR and vocoder. All-pole spectral parameters are investigated in detail to find a better representation with respect to quantization and interpolation characteristics, partly at NTT and Bell Labs, leading to PARCOR and LSP. These parameters were applied to narrow band speech coders or LPC vocoders and speech synthesizer chips in the late 1970s and early 1980s, but the speech quality was insufficient for digital mobile telephone application. The problem was later solved by using hybrid CELP and MLP coding mainly at BTL. LPC analysis was also applied to acoustic front-end for ASR. Again it was found that LPC suffers with additive noise and linear/nonlinear distortions. Whereas computational efficiency of LPC was used to be the most prominent advantage, today we have gained thousands of times the processing power at a ten thousandth of the cost in 40 years. It is hoped a novel series of speech analysis methods, whose competence is comparable to human auditory system, should be developed at any expense of computational complexity.
Acoustics at the shrine of St. Werburgh
Authors: Lubman David
Published: October 2013

England’s Chester Cathedral (Anglican) contains a shrine to its patron saint, St. Werburgh, a 7th century Saxon princess of Mercia who became a nun and abbess. The 8th or 9th century shrine is far older than the 16th century cathedral. The shrine was enlarged around 1340, apparently because of its popularity as a place of pilgrimage and reported miracles. It was smashed by 16th century Henrician reformers and restored in the 19th century. The present shrine contains six recesses. In pre-Reformation times, kneeling pilgrims placed their heads in recesses to deliver spoken petitions to the saint. Analyses of binaural recordings made in a recess reveal that vocalizations are dramatically reinforced (10.4 dBA, 15.5 dB overall) and distorted by resonances (the strongest is 25.1 dB @ 125 Hz). Documented acoustical features suggest the shrine comprises a forgotten or previously unknown form of religious theater-for-one in which the sound experience was important. Modern listeners can enjoy a pre-Reformation auditory virtual reality experience via binaural recordings made at the shrine (best with quality headphones). [The author gratefully acknowledges architect Peter Howell of the Dog Rose Trust, a registered English charity, who reported the unusual shrine acoustics; and the cooperation of Chester Cathedral authorities.]
Use of surrogate samples to study variation of diffuse field absorption coefficients of fiberglass with altitude
Authors: Godfrey Richard D.
Published: October 2013

ASTM C 423 identifies air temperature and relative humidity as significant parameters, but does not address air density effects. In previous papers, normal and diffuse field analysis showed significant changes in predicted absorption coefficients with altitude. These predictions were validated experimentally for normal incidence in a vacuum chamber, and by using surrogate samples, thus showing the feasibility to study altitude effects in a single laboratory. Mechel design charts are normalized by two parameters. One is not dependent on air density. The other (R) is the ratio of flow resistance and the impedance of air. At constant thickness, the effect of lowering air density can be studied by increasing the sample flow resistivity. Samples with flow resistivity ratios of 1.25 and 1.5 were studied in a diffuse field following ASTM C 423 methodology. These values correspond to altitudes of 6000 and 10700 ft in altitude, respectively. These results followed the predicted trends. At both altitudes, samples with R’s of 2 and 4 had higher average absorption coefficients. For R=8, the reverse effect was observed. It was concluded that the effects of altitude are not limited to the impedance tube method, but are also present in the reverberation room method.
Studying social cognition in marine mammals
Authors: Tyack Peter
Published: October 2013

Ron Schusterman has played an important role in broadening the perspective of marine mammal cognition studies from narrow comparisons to human language to more general cognitive concepts. He has also contributed to our understanding of learning mechanisms for individual recognition in pinnipeds, linking naturalistic observations with controlled studies in captive settings. I discuss how odontocetes learn to develop signals for individual and group recognition. Bottlenose dolphins use vocal learning to develop individually distinctive whistles in the first 1–2 years of life, but they also maintain the ability to imitate whistles throughout their lifetime. As maturing males form a coalition, their whistle repertoires converge. Species with more stable groups than dolphins use vocal learning to develop repertoires that are group distinctive. Schusterman has recently developed theoretical approaches to thinking about how animals form categories of social knowledge such as coalition or group. Depending upon the social context, animals that modify their vocalizations based upon auditory input and social relationships may use similar vocal learning mechanisms to develop quite different vocal repertoires. I will discuss the interaction between communication, social knowledge, and cognition in marine mammals from the approaches suggested by Schusterman for the study of social knowledge.
Relationship between auditory evoked potential (AEP) and behavioral audiograms in odontocete cetaceans
Authors: Houser Dorian S. , Finneran James J. , Carder Donald A. , Ridgway Sam H. , Moore Patrick W.
Published: October 2013

Auditory evoked potentials (AEPs) offer an alternative to behavioral methods of determining auditory sensitivity in marine mammals. The technique can be performed without the need for animal training, substantially expediting the process, and has the potential for application to stranded and rehabilitating marine mammals, thus providing an opportunity to determine hearing sensitivity in animals not likely to be kept in captivity. As an emerging technology in the field of marine mammalogy, the equivalence of AEP and behavioral thresholds remains to be quantitatively assessed. Human and laboratory animal AEPs are typically −5 to +20 dB of behaviorally determined thresholds and vary by technique and frequency tested. To be an effective tool in the field of marine mammalogy, the expected variation in AEP thresholds relative to behavioral thresholds in marine mammal species needs to be determined. We compare the behavioral and AEP audiograms of several odontocetes covering a range of normal hearing to profound hearing loss and demonstrate the offsets between results obtained with the two methods. Thresholds determined by the two methods show generally good agreement and demonstrate the utility of AEPs as an emerging technology in the study of marine mammal audiometry.
MEMS-based acoustic arrays: Promise and challenges
Authors: Sheplak Mark , Nishida Toshikazu , Cattafesta Louis
Published: October 2013

A review of microelectromechanical system (MEMS)-based directional acoustic array technology is presented. The prospects for reducing cost, improving speed, and increasing mobility over conventional array technologies is critically reviewed. The advantages and limitations of existing devices are discussed. Finally, unresolved technical issues are summarized for future sensor development. A specific example of a system is presented that uses 16 hybrid-packaged silicon-micromachined piezoresistive microphones mounted to a printed-circuit board and a high-speed signal processing system to generate the array response over 2400 scan locations in under 20 s. The hybrid microphone packages show an average sensitivity of 0.8 mV/Pa with matched magnitude (0.6 dB) and phase (1 deg) responses between devices. The measured array response matches the theoretical response over the frequency range of 3 to 8 kHz with a localization error of 0.3 in. The array has a minimum detectable signal of 43.5 dB SPL for a 1-Hz bin at 6 kHz and a maximum pressure input of at least 160 dB SPL. These results represent a proof-of-concept demonstration of a high-speed, low-cost directional acoustic array system.
Design and fabrication of 128-channel MEMS-based acoustic array
Authors: Shams Qamar A. , Humphreys William M. , Sealey Bradley S. , Adams Jimmy K. , Comeaux Toby , Ingham John C. , Babel Walter C.
Published: October 2013

Surface-mount microphones based on MEMS (micro-electromechanical system) technologies have recently become viable as component-level engineering solutions for acoustic measurements. In addition, advances in microelectronics, flexible circuitry, and array processing have motivated the design of a high-speed, low-cost acoustic array system for aeroacoustic measurements. A variety of microphones are available in the market today. Each type of microphone has its benefits and drawbacks. For example, standard condenser microphones have excellent sensitivity, stability, and high frequency response, but tend to be unwieldy, expensive, and require relatively high operational voltages. Electret microphones are small and fairly inexpensive but their performance deteriorates if exposed to moderately elevated temperatures. MEMS microphones combine the best features of the electret and condenser microphones while occupying a volume of less than 20 cubic millimeters. This paper details the custom-made 128-channel MEMS-based acoustic array for wind tunnel applications as well as the electrical, mechanical, and acoustic properties of the MEMS microphones utilized here.
Free-field calibration of the pressure sensitivity of air-condenser, electret, MEMS, and piezoresistive microphones at frequencies up to 80 kHz
Authors: Herring Gregory C. , Zuckerwar Allan J. , Elbing Brian R.
Published: October 2013

A free-field substitution method for calibrating measurement microphones at frequencies up to 80 kHz is demonstrated with both grazing and normal-incidence geometries. The method is suitable for newer technology microphones that cannot be calibrated with the electrostatic actuator (EA), currently the industry standard for high frequency calibrations. A substitution-based method, as opposed to a simultaneous method, avoids problems associated with the nonuniformity of the sound field and uses a 1/4-inch pressure sensitive microphone as a known reference. A commercially available reference sound source (centrifugal fan) is used as a broadband acoustic source. Although the broadband excitation produces smaller instantaneous signal-to-noise ratios than tonal excitation, it minimizes reflection-related interferences that often plague free-field measurements. Calibrations were performed on 1/4-in. air-condenser, electret, piezoresistive, and MEMS microphones in an anechoic chamber. Five repetitions of a single microphone, over three months, give a typical reproducibility of ±0.2 dB. Because the air-condenser microphone can be calibrated with the EA, it was possible to estimate the accuracy of this free-field method by comparing the pressure sensitivity, as derived from the free-field measurement, with that of the EA calibration. A typical comparison gives a rms difference of ±0.4 dB, over the range 2–80 kHz.
On the acoustical design of the ears of flue organ pipes
Authors: Sakamoto Yumiko , Yoshikawa Shigeru , Angster Judit M.
Published: October 2013

Application of the ears to the flue organ pipe is one of the important voicing techniques. Ears are the projections on both sides of the pipe mouth. Organ builders say the ears make not only the sound lower and darker, but also the buildup of tone smoother and quicker. The aim of this research is to confirm their recognitions and make the causes clear. For that purpose, we made acoustical and flow measurements (measurement of the velocity profiles at the mouth) with model pipes. As a result, we could confirm the recognition of the organ builders. In addition, our experiments indicate a slight increase in the blowing pressure in the foot and an increase in the inharmonicity of the pipe eigenmodes. The ear reduces the maximum jet velocity but keeps the characteristic profiles. In some cases, the profiles move as a whole more inside of the pipe. Recent acoustic measurements (eigenmodes of the pipe resonator and of the mouth tone, attack transient and stationary spectrum) on real organ pipes with ears of different heights will be also reported at the meeting.
A comparison of different expression devices in pipe organs
Authors: Braasch Jonas , Rossing Thomas D.
Published: October 2013

After the introduction of the orchestra crescendo at the end of the 18th century by the ‘‘Mannheim school,’’ the ability to play the organ expressively like an orchestra was one of the organ builders’ greatest concerns. Soon, both the wind swell and door swell came into fashion, and later another swell system, the crescendo wheel, was introduced. While both door swell and crescendo wheel can be successfully applied to all types of organ stops, the wind swell only works well with free reeds for tuning reasons. In this investigation, all three swell systems were measured on various instruments and compared to each other. In addition, two free-reed pipes were measured at the Northern Illinois University using a laser vibrometer. The crescendo wheel was found to be most effective, and for frequencies around 2 kHz the increase in sound-pressure level could be up to 50 dB between the softest and the loudest adjustment. The maximum dynamic range for the wind and the door swells is approximately 10 dB in the same frequency range. While the dynamic range is lower for the wind swell and the door swell, their advantage is the continuous variability.
Distributed exhaust for jet noise reduction
Authors: Schein David B.
Published: October 2013

Acoustic detection of low flying, jet-powered military air vehicles and takeoff noise levels from commercial jets are often driven by jet mixing noise radiated from the exhaust. Cueing provided by the noise signature results in increased opportunity for a ground-based visual observer against such a target. Significant reductions in detectability and takeoff noise levels can be achieved through reductions in jet noise. Research has been conducted over the past several years to develop innovative, quiet, distributed exhaust nozzle (DEN) concepts, which attempt to achieve revolutionary reductions in jet mixing noise while minimizing propulsion penalties. The DEN’s benefit relies on discharging the exhaust flow through many miniature nozzles rather than one or two large nozzles. Noise suppression from the DEN concept results from a favorable shift in frequency content compared to conventional jets. Significant increases in atmospheric attenuation and decreases in the ear’s sensitivity at these higher frequencies result in much reduced detection ranges and perceived noise levels. This technology is applicable to both subsonic and supersonic aircraft, and particularly to low flying fixed-wing unmanned air vehicles, special operations forces transports, and future commercial transports with especially stringent noise reduction requirements. [Work supported by NASA LaRC.]
Lateral attenuation and airport noise modeling
Authors: Shepherd Kevin P.
Published: October 2013

Prediction of airport noise is generally accomplished using semiempirical methods such as those contained within the F.A.A.’s Integrated Noise Model or the U.S.A.F.’s Noisemap. One component of these models addresses the prediction of noise for observers located at lateral, or sideline, positions for which the elevation angle to the source is small. Under these conditions the effect of the presence of the ground surface on sound propagation is important, as is the source directivity. Recent efforts in the United States and Europe have been aimed at separating these two phenomena with the result that acoustical directivity characteristics have been quantified for a range of commercial aircraft. Differences between aircraft with wing- and tail-mounted engines are significant. These recent studies will be reviewed and attempts made to identify the physical mechanisms responsible for the observed source directivity characteristics.
Fluctuating pressure distributions around spherical foam windscreens
Authors: Webster Jeremy , Raspet Richard , Dillion Kevin
Published: October 2013

The low-frequency wind noise reduction produced by spherical foam windscreens outdoors can be estimated by measuring the steady-state flow pressure distribution around the windscreen and then area averaging to produce a pressure response at the center of the sphere [Zheng (2003)]. We have constructed a foam windscreen with a distribution of four probe microphones near the surface to investigate the validity of this hypothesis at low frequencies and to measure correlations between the pressure fluctuations at different positions on the surface and at the center. The aim of this research is to develop methods of wind noise reduction which combine the effectiveness of passive foam windscreens with active multiple microphone processing techniques. In this paper we describe the construction of the test windscreen and initial measurements made outdoors.
The near ground structure of the nocturnal sound field: The existence of a quiet height in the frequency domain
Authors: Waxler Roger , Talmadge Carrick L. , Gilbert Kenneth E. , Dravida Shantharam
Published: October 2013

The sound field generated by a point source in the downward refracting atmosphere typical of the nocturnal boundary layer is considered. Theoretical arguments are given that, at a fixed frequency, at ranges greater than a few hundred meters from the source the magnitude of the sound field necessarily has a deep minimum a few meters above the ground. The precise height of this minimum depends on the sound speed profile as well as the frequency, decreasing with increasing frequency. It is shown that a recently developed modal description of the nocturnal sound field yields a simple explanation for the existence of this quiet height and its emergence with increasing range. Experimental results verifying the existence and frequency dependence of the quiet height are presented.
A 3-D hierarchic finite-element tool with infinite elements for structural acoustic scatter modeling
Authors: Zampolli Mario , Burnett David S. , Tesei Alessandra , Blottman John B. , Westermann Timothy A.
Published: October 2013

Fully three dimensional prolate spheroidal acoustic infinite elements have been added to a steady state finite-element structural acoustics tool (FESTA), which is based on hierarchic polynomial shape functions. The infinite elements are used as an alternative to the Bayliss–Turkel approximate radiation condition to ensure outward propagation of the scattered and/or radiated acoustic fields surrounding one or more objects. The main advantages of the infinite elements compared to the Bayliss–Turkel condition are the reduction of the distance required between the object(s) and the boundary of the finite element computational domain, and meshes which can be used over a broader frequency band. Furthermore, the infinite element formulation adopted, which is based on the prolate spheroidal version of the Atkinson–Wilcox expansion, is ideal for modeling the scattering and/or radiation from elongated structures of particular interest to the underwater acoustics community. The performance of the tool is assessed by comparison to an axisymmetric thin-shell FE/virtual source code, and to a hierarchic axisymmetric FE code with infinite elements. Furthermore, the tradeoffs between the infinite elements and the prolate spheroidal Bayliss–Turkel conditions are assessed.
Children’s reliance on spectral information for understanding speech in noise
Authors: Nelson Peggy , Nie Yingjiu
Published: October 2013

Children have significantly greater difficulty understanding in noisy situations than do adults. This has serious implications for the design and structure of classrooms and for communicating effectively with young children. The reasons for children’s susceptibility to noise are not well understood. One factor may be children’s reliance on frequency cues for speech understanding. This project investigated children’s perception of speech in noise by systematically evaluating the perceptual importance of speech frequency information. Typically developing children (ages 7 to 13 years) and adults were tested using full spectrum speech as well as 4-, 8-, and 12-band simulations of cochlear implant processing (after Shannon et al., 1995). Stimuli were color and number words selected from the coordinate response materials (Brungart, 2001), and were presented in quiet and in pink noise at +5-dB signal-to-noise ratio. Preliminary results support the hypothesis that children rely more on the frequency content of speech than do adults for understanding speech in noise, and that children need fuller frequency representations of speech than do adults to segregate speech from background noise. [Work supported by the Graduate School of the University of Minnesota.]
Interactions between the frequency allocation, stimulation mode, and pitch perception on speech recognition by cochlear implant listeners
Authors: Galvin John , Fu Qian-Jie
Published: October 2013

In cochlear implants, speech recognition depends strongly on the acoustic frequency-to-electrode assignment. In a previous experiment, the pitch of an electrode was shown to vary with stimulation mode, especially for widely spaced electrode pairs. In the current study, Nucleus-22 users’ speech recognition was measured as functions of the stimulation mode and frequency allocation. Two stimulation modes were tested: BP+1 and ‘‘mixed-mode’’ [the active electrode was fixed and the return electrode varied, i.e., (2,22), (2,21), etc.]. A range of frequency allocations was tested to measure the effect of spectral mismatch between the acoustic input and the electrode configurations. Preliminary results showed a differential effect of frequency allocation, depending on the stimulation mode used in the speech processor. For the BP+1 configuration, multi-talker vowel recognition decreased significantly as the frequency allocation was shifted from Table 9 (150–10823 Hz) to Table 3 (85–6184 Hz). However, for the mixed-mode processor, performance improved as the allocation was shifted from Table 3 to Table 9. Because Table 3 may have produced less spectral mismatch (given the generally lower pitch percepts for widely spaced electrode pairs), mixed mode processors may be used to extend the pitch range and thereby map more low-frequency spectral content to apical electrodes.
Time-reversal arrays in underwater acoustics
Authors: Roux Philippe , Akal Tuncay , Hodgkiss W. S. , Kuperman W. A. , Song Hee Chun , Stevenson Mark
Published: October 2013

Since 1996, several time-reversal experiments have been performed in shallow water acoustics involving source and receive arrays made of many elements. This equipment allowed us to reach different goals among which were long-range time-reversal focusing, passive and active multiple input multiple output communications, bottom refocusing from reverberation, acoustic barrier detection, comparison between reciprocal and nonreciprocal time-reversal and shallow water tomography. This presentation will review this work with emphasis on array signal processing.
Effect of multipath propagation on the statistics of reverberation in shallow-water environments
Authors: LePage Kevin D.
Published: October 2013

The envelope of direct-path scattered energy has been shown theoretically to deviate from Rayleigh statistics for small ensembles of scatterers with non-Gaussian amplitude distributions [Abraham and Lyons, IEEE J. Ocean. Eng. (2002)]. Here, we investigate the effects of multipath propagation, scatterer correlation length scale, scatterer amplitude distribution, and sonar source and receiver characteristics on the distributional properties of reverberation envelopes in shallow-water environments. For this study we employ a broadband version of the coherent time-domain R-SNAP reverberation model. Model predictions indicate that the non-Rayleighness of shallow-water reverberation caused by distributions of homogeneously distributed diffuse scatterers is time dependent, and sensitive to the number of resolved multipath in the waveguide, the source bandwidth, and the source/receiver beamwidths of the sonar system under consideration. [Work supported by ONR.]
Analysis of digital side-scan sonar survey data in support of KauaiEx
Authors: Caruthers Jerald W. ,   The KauaiEx Group
Published: October 2013

A side-scan sonar (SSS) survey was conducted off the northwest coast of Kauai, HaI, in support of the high-frequency, channel-characterization experiment (KauaiEx). The SSS used in this survey was a modified system that operates alternately at 150 and 300 kHz and produces digital data as well as standard tiff images of the seafloor. This paper discusses analyses of the digital backscattering data. Previous reports discussed the system characteristics and analyses of standard tiff images produced by the SSS as originally configured. The digital data are high-resolution in space and in dynamic range and offer improved interpretations of the character of the bottom in the KauaiEx range. The data in the range yield interpretations of sand ripples parallel to the depth contours with wavelengths of about 1 m and heights of a few centimeters. These ripples appear to cover the entire KauaiEx range. Data taken at a shallow-water site nearby provide additional information on bottom characteristics that include lava flows, mud, and sand. [Work supported by the Ocean Acoustics Program of ONR.]
Gaussian beam summation formulation for rough surface scattering in complex configurations
Authors: Gordon Goren , Heyman Ehud , Mazar Reuven
Published: October 2013

A Gaussian beam summation (GBS) representation for wave propagation problems in the presence of rough surfaces is introduced. In this formulation, the coherent and incoherent scattered fields are decomposed into a discrete phase space summation of Gaussian beams (GBs) that emanate from a discrete set of points and directions on the rough surface. The formulation therefore involves stochastic GB-to-GB scattering matrices for the coherent and incoherent fields, and deterministic GB propagators. It benefits from the simplicity and accuracy of the latter, and can be used in applications involving propagation in complex environments and in inverse imaging. The coherent and noncoherent GB2GB scattering matrices are calculated from the statistical moments of the rough surface scattering amplitude, which is given either analytically or empirically. As an example, closed form expressions are calculated within the small perturbation regime. The GB2GB matrices are then used to analyze the propagation along rough-surface waveguides and to estimate the bi-static reverberations. The results clearly demonstrate and explain the phase-space footprints of the stochastic multiple scattering events and of the deterministic propagation between the surfaces.
Interpretation of the ground wave arrival as a head wave sequence
Authors: Choi Jee Woong , Dahl Peter H.
Published: October 2013

The ground wave arrival (sometimes referred to as a precursor arrival) is interpreted in this work as a train of the head waves, defined here as the head wave sequence. For a Pekeris waveguide, the spectrum of the ground wave arrival is found to be that corresponding to the spectrum of a discrete (time-separated) head wave arrival, evaluated at the modal cutoff frequencies, and the spectrum of a discrete head wave arrival is known to be proportional to the source spectrum divided by frequency. These properties are illustrated for a Pekeris waveguide via Fourier synthesis of a narrow-band complex parabolic equation (PE) field, using the RAM PE code. Although the ground wave arrival in a more realistic waveguide is complicated by several effects, e.g., layering, gradients in the sediment sound speed, some of the above ideas carry over. The introduction of a sound speed gradient in the sediment also produces a key difference in the spectrum of a ground wave arrival (or head wave sequence); this effect is demonstrated through PE simulation, and is also related to data collected in the Yellow Sea. [Research supported by ONR Ocean Acoustics.]
Musical instrument sound levels
Authors: Hoover K. Anthony , Carballeira Andrew , Harght Pam , Ortallono Sam
Published: October 2013

Sound-pressure levels produced by different types of musical instrument can vary considerably. Convincing auralization and appropriate sound-isolation design can be greatly affected by this variation. This paper will present A- and C-weighted statistical sound levels produced by a variety of musical instruments and musical genres, measured at Berklee College of Music. A simple comparison between measured sound isolations using a broadband noise source and a reproduced-music source will also be discussed.
Spectrum-based analysis and synthesis of percussion sounds
Authors: Beauchamp James W.
Published: October 2013

From the standpoint of the listener, convenient ways to catagorize percussion sounds are in terms of mode frequency positions, mode frequency spacing, and decay rates. Some sounds, like those of bells, chimes, and other struck bars, have modes which are widely spaced. Other sounds, such as those of timpani, tam-tam, and cymbals, have many more closely spaced modes. Measurements of average modal spacing, time-varying spectral shift, spectral incoherence, attack time, and decay rate are presented for a variety of percussion instrument sounds. Examples of percussion synthesis using this information will be presented.