bluuebunny/crossref_metadata_2025 huggingface dataset:
Ray-based geoacoustic inversion for high frequency broad band data
Authors: Siderius Martin , Hursky Paul , Porter Michael
Published: October 2013

One of the difficulties in making reliable acoustic propagation predictions in shallow water is the lack of good information about the seabed type. In recent years, matched field processing- (MFP-) based geoacoustic inversion has been shown as a practical technique for estimating properties of the seabed. The MFP inversion method compares measured acoustic fields to those generated using an acoustic propagation model. Often, thousands of forward model calculations are required to find a set of seabed parameters that correlate well with the measured data. The large number of forward model calculations is computationally demanding and this is made worse when matching at higher frequencies or over broad band data. Ray-based propagation modeling relieves some of the computational burden since the calculation time is fairly insensitive to frequency and is inherently broad band. Further, the ray arrival amplitudes and delays are well suited for interpolation and this allows the seabed parameter search space to be explored using just a few ray trace calculations. The broad band nature of the modeled data provides flexibility in choosing correlation functions and this allows for more robust inversions. In this presentation, techniques using ray-based propagation modeling will be applied to the geoacoustic inversion problem.
Quantitative theoretical explanation of Apfel’s experimental phase diagrams for sonoluminescing bubbles
Authors: Lohse Detlef
Published: October 2013

Robert Apfel had an enormous impact on the research of single bubble sonoluminescence [M. P. Brenner, S. Hilgenfeldt, and D. Lohse, Rev. Mod. Phys. 74, 425 (2002)], the light emission of a single sound-driven bubble. In 1996, at the ASA Meeting in Hawaii, he posed a challenge to the theoreticians in the field: make experimentally testable predictions on single bubble sonoluminescence. Apfel collected the predictions and gave a wonderful review talk on the state of the field. Later, he came back several times to that list, comparing the predictions with latest experimental results. Our own predictions in those days referred to the phase diagrams of single bubble sonoluminescence. Later Apfel himself, together with Ketterling, measured those phase diagrams experimentally [J. A. Ketterling and R. E. Apfel, Phys. Rev. Lett. 81, 4991 (1998); Phys. Rev. E 61, 3832 (2000)]. Though qualitatively our 1996 predictions turned out to be correct, a full quantitative model could only be developed recently [R. Toegel and D. Lohse, J. Chem. Phys. (in press); B. D. Storey and A. J. Szeri, Phys. Rev. Lett. 88, 074301 (2002)]. In the presentation we will compare the model predictions with Apfel’s data.
Attenuation measurements of passive linear and nonlinear hearing protectors for impulse noise
Authors: Murphy William J. , Kardous Chucri A.
Published: October 2013

As a part of a NIOSH Health Hazard Evaluation of law-enforcement personnel, the attenuation of several types of earplugs were measured in response to impulse noise produced by small-arms gunfire. The earplugs were primarily flanged premolded plugs produced by EAR/Aearo and Bilsom/Baccou-Dalloz. Measurements for the North Sonic Ear Valve, EAR Classic earplugs, and EAR Ultra 9000 passive nonlinear ear muff were conducted. The EAR premolded earplugs were the Combat Arms passive linear and nonlinear, HiFi and Ultratech earplugs. The Bilsom devices were the 555, 655 NST, and 655 ISL earplugs. The Combat Arms and 655 ISL earplugs both utilize a cartridge developed by the French German Research Institute de Saint Louis that provides nonlinear attenuation. The peak reduction of these devices ranged between 10 and 28 dB. The slope of peak reduction with peak level for the Ultra9000 device was about 0.5 dB/dB, while the slopes for most earplugs were about 0.1 to 0.3 dB/dB for weapons impulses between 159- and 170-dB peak level. The peak reductions ranged from 6 dB for the North Ear valve to 30 dB for the EAR Classic foam earplug.
Can we learn anything new from the animal’s perspective?: Bayesian stimulus estimation based on neural spike trains
Authors: Jackson B. Scott
Published: October 2013

The classical approach to understanding a neural coding scheme is to characterize spike trains elicited by multiple presentations of the same stimulus, an example being the rate-intensity function. However, the task for the organism is deciphering the stimulus content from spike trains elicited by single stimulus presentations, a process that is more analogous to Bayesian stimulus estimation. The usefulness of Bayesian stimulus estimation in the neural context, using the auditory periphery as a model system, was investigated. More specifically, the encoding of stimulus intensity in the spike rate of single primary auditory neurons was examined. It was found that the results of this method are heavily influenced by the a priori stimulus distribution and that apparent benefits of this approach, such as linearization and, in some instances, increased stimulus dynamic range, are offset by concomitant disadvantages, such as increased estimation error and decreased stimulus dynamic range in other instances. Hence, in this context, Bayesian stimulus estimation does not contribute meaningful additional knowledge to our understanding of neural coding, and these results suggest that related methodologies, such as stimulus reconstruction and information theoretic methods, be used and interpreted with caution. [Work supported by Syracuse University and NIH Grant 5-P01-DC000380.]
Robust automatic recognition of telephone speech
Authors: O’Shaughnessy Douglas , Sid-Ahmed Selouani
Published: October 2013

A method is presented to compensate cepstral coefficients (MFCCs) in a speech recognition system for degraded telephone channel conditions. The technique proposed is based on a combination of the Karhonen–Loeve Transform (KLT) and Genetic Algorithms (GA). The idea consists of projecting the band-limited MFCCs onto a subspace generated by the genetically optimized KLT principal axes. Experiments show a clear improvement when the method was applied to the NTIMIT speech database. Word recognition results obtained on the HTK toolkit platform using N-mixture tri-phone models and a bigram language model are presented and discussed.
Spectral moments and alternative methods of characterizing fricatives
Authors: Blacklock Oliver S. , Shadle Christine H.
Published: October 2013

Parameters that both distinguish fricatives in a given language, and characterize normal and disordered productions would be useful. Various parametrizations have been tried since the 1950s with only partial success. Forrest et al. [J. Acoust. Soc. Am. 84, 115–123 (1988)] defined spectral moments of speech power spectra, but found that these failed to distinguish the voiceless English fricatives. They have however proven useful for distinguishing sibilants, and for characterizing long-term place changes of sibilants in disordered speech. As discussed in this paper, other changes, and nonsibilants, are not so easily characterized. Relative amplitude differences have to date not been incorporated in spectral moment methodology. The quality of the spectral estimate (whether consistent, unbiased) affects the variability of the moments, and means of obtaining a good estimate usually involve assuming stationarity or ergodicity. Finally, moments capture little that is known about fricative production. Modern techniques for consistent PSD estimation (e.g., multitaper and wavelet analysis) have been examined; their suitability in analyzing fricatives is discussed. While these modern PSD estimation methods may improve the performance of spectral moments, they also lend themselves to new parametric methods which do not rely on assumptions of ergodicity or stationarity. [Research supported by EPSRC.]
Hearing in three dimensions
Authors: Shinn-Cunningham Barbara
Published: October 2013

One of the key functions of hearing is to help us monitor and orient to events in our environment (including those outside the line of sight). The ability to compute the spatial location of a sound source is also important for detecting, identifying, and understanding the content of a sound source, especially in the presence of competing sources from other positions. Determining the spatial location of a sound source poses difficult computational challenges; however, we perform this complex task with proficiency, even in the presence of noise and reverberation. This tutorial will review the acoustic, psychoacoustic, and physiological processes underlying spatial auditory perception. First, the tutorial will examine how the many different features of the acoustic signals reaching a listener’s ears provide cues for source direction and distance, both in anechoic and reverberant space. Then we will discuss psychophysical studies of three-dimensional sound localization in different environments and the basic neural mechanisms by which spatial auditory cues are extracted. Finally, ‘‘virtual reality’’ approaches for simulating sounds at different directions and distances under headphones will be reviewed. The tutorial will be structured to appeal to a diverse audience with interests in all fields of acoustics and will incorporate concepts from many areas, such as psychological and physiological acoustics, architectural acoustics, and signal processing.
Using 3-D modeling to design multichannel audio systems
Authors: Olson Bruce C.
Published: October 2013

Implementation of multichannel sound systems requires a balancing of a number of different parameters and criteria to produce a cohesive sound field that is appropriate to the venue. Use of a 3-D modeling application allows for an iterative and interactive approach to optimizing the system. This design process will be explored and explained using some recently completed projects. Coordination with specific acoustical features of the room is inherently part of this process.
Modal inverse methods: An overview
Authors: Becker Kyle M. , Frisk George V.
Published: October 2013

Geoacoustic inversion using modal inverse methods is based on a very simple experimental geometry. Data are acquired from synthetic aperture measurements of a point source acoustic field using a single hydrophone translating through the water column at constant depth. The modal composition of the propagating field is extracted by exploiting the Hankel transform relationship between the acoustic field, measured as a function of range, and the horizontal wavenumber spectrum for horizontally stratified media. Modal content, estimated as discrete values of the horizontal wavenumber, are used as input data for a general linear inverse problem, where small perturbations to a background sound speed profile are related to changes in individual horizontal wavenumbers. An overview of the method is presented with an emphasis placed on the application to range-dependent shallow water environments. Inversion results are presented from a recent workshop on geoacoustic inversion for range-dependent environments, as well as from recent at-sea measurements. [Work supported by ONR.]
The functional anatomy of speech perception: Dorsal and ventral processing pathways
Authors: Hickok Gregory
Published: October 2013

Drawing on recent developments in the cortical organization of vision, and on data from a variety of sources, Hickok and Poeppel (2000) have proposed a new model of the functional anatomy of speech perception. The model posits that early cortical stages of speech perception involve auditory fields in the superior temporal gyrus bilaterally (although asymmetrically). This cortical processing system then diverges into two broad processing streams, a ventral stream, involved in mapping sound onto meaning, and a dorsal stream, involved in mapping sound onto articulatory-based representations. The ventral stream projects ventrolaterally toward inferior posterior temporal cortex which serves as an interface between sound and meaning. The dorsal stream projects dorsoposteriorly toward the parietal lobe and ultimately to frontal regions. This network provides a mechanism for the development and maintenance of ‘‘parity’’ between auditory and motor representations of speech. Although the dorsal stream represents a tight connection between speech perception and speech production, it is not a critical component of the speech perception process under ecologically natural listening conditions. Some degree of bi-directionality in both the dorsal and ventral pathways is also proposed. A variety of recent empirical tests of this model have provided further support for the proposal.
Is there still a TRACE of trace?
Authors: McClelland James , Mirman Daniel , Holt Lori
Published: October 2013

According to the TRACE model [McClelland and Elman, Cogn. Psychol. 18, 1–86 (1986)], speech recognition is an interactive activation process involving the integrated use of top-down (lexical) and bottom-up (acoustic) information. Although it is widely accepted that there are lexical influences on speech perception, there has been a disagreement over their exact nature. Two contested predictions of TRACE are that (a) lexical influences should delay or inhibit recognition of phonemes not consistent with lexical information and (b) a lexical influence on the identification of one phoneme can trigger compensation for co-articulation, affecting the identification of other phonemes. Others [Norris, McQueen, and Cutler, BBS 23, 299–370 (2000)] have argued that the predicted effects do not occur, taking this to support an alternative to the TRACE model in which lexical influences do not affect perception, but only a post-perceptual identification process. We re-examine the evidence on these points along with the recent finding that lexical information may lead to a lasting adjustment of category boundaries [McQueen, Norris, and Cutler, Psychonomics Abstract 255 (2001)]. Our analysis indicates that the existing evidence is completely consistent with TRACE, and we suggest additional research that will be necessary to resolve unanswered questions.
Channel order selection in blind deconvolution based on eigenvector characteristics and using normal modes in conjunction with multipath compression for source identification
Authors: LaRue James P. , Ioup George E. , Ioup Juliette W. , Smith George B.
Published: October 2013

Two related topics in blind deconvolution are presented. The first concerns channel order selection. Many blind deconvolution techniques assume the filter length to be known. One technique of filter length determination uses the dimension of a nullspace obtained by decomposing correlation matrices. Although the dimension is usually determined by calculations involving the eigenvalues, this alternative method relies on the known characteristics of eigenvectors among the subspaces. Multipath functions in underwater acoustics commonly have high kurtosis. The new method can give correct estimates on filter length for signal-to-noise ratios as low as 10 dB. The matrix methods described may not work as well when the sample length increases. However, the second topic shows how normal mode properties in conjunction with time compression in modeled multipath functions may be used to enhance blind deconvolution in a waveguide when the sample length is large. Given data from a vertical array of hydrophones, one may accurately obtain a source identification. The methods of the tracking and positioning of normal modes are incorporated with an input of a linear frequency-modulated pulse propagated through an ocean environment. [Research supported by ONR.]
Performance bounds on wave-front curvature ranging
Authors: Bondaryk Joseph E. , Abbot Phillip , Gedney Charles , Sullivan Edward
Published: October 2013

Acoustic wave-front curvature ranging (WCR) is a localization method that exploits the curvature of the arriving wave-front to determine the range of an acoustic emitter. By assuming that the wave-front is circular, it provides an estimate of the range as the radius of the circle. It is based on the measurement of time of arrival differences for correlated transient signals between pairs of spatially separated receivers. Performance bounds are determined by: (1) source and receiver geometry, (2) signal characteristics, and (3) spatial coherence limits due to oceanic effects. Both theoretical and Monte Carlo studies have been done which define the performance limitations as a function of aperture size, signal structure and spatial coherence. Results show that for a given aperture and fixed range accuracy, the low-frequency limit is bounded by the signal characteristics (SNR and time-bandwidth product), whereas the high-frequency limit is determined by the coherence length allowed by the ocean. This high-frequency limit is presently estimated by extrapolation of lower frequency models and is imprecisely known due to the unavailability of high-frequency, short time spatial coherence data.
Forward modeling requirement for short-range geoacoustic inversion on a towed array
Authors: Yang T. C. , Yoo K.
Published: October 2013

For geoacoustic inversion using the tow-ship noise received on a towed horizontal line array (HLA), one of the issues is speed and accuracy of the forward model. Is the normal mode good enough or is ray propagation adequate for modeling short-range forward propagation? Earlier work [Kuperman et al., IEEE J. Oceanic Eng. 10 (1985)] showed that the continuum contribution arrives at beams far away from the forward directions. For practical applications, such contributions are often neglected to minimize interference from undesired sources. We examine in this paper the same issue (the importance of high angle arrivals at short range) in both the time and beam domain in the context of several bottom models (templates). Note that the high angle arrival contributions depend on the bottom type, the source–receiver range, the towed array aperture/spacing and the acoustic frequencies. The results of this analysis will shed light on the requirement of forward models at short ranges on a HLA. [Work supported by ONR.]
Inversion of range-dependent geoacoustic properties in South China Sea ASIAEx01 experimental site
Authors: Turgut Altan , Pasewark Bruce , Orr Marshall , Lynch James , Chiu Ching Sang
Published: October 2013

Matched-field inversion of range-dependent geoacoustic properties is studied using broadband (50–200 Hz, 240–260 Hz, and 550–600 Hz) acoustic data collected in the South China Sea during the ASIAEx01 experiment. Range-dependent sediment sound-speed and attenuation profiles are inverted using a global optimization scheme based on genetic algorithms to minimize an objective function defined by the Bartlett processor output. For the forward model, an efficient coupled normal mode model is used to calculate broadband acoustic fields incorporating range-dependent bathymetry and sediment layers thickness obtained by chirp sonar surveys. Inversions were performed starting from a range-independent region with a relatively short source/receiver distance. Additional geoacoustic profiles were inverted by incorporating the previously inverted profiles as the source/receiver distance was increased. The results obtained at three different frequency bands are in good agreement, especially when the range-dependency of bathymetry and sediment layers thickness is included in the inversion. [Work supported by ONR.]
Observing sonoluminescent UV photons below the water cutoff
Authors: August Robert , Forsley Lawrence , Whitlock Robert
Published: October 2013

This paper presents sonoluminescent UV photon data observed in water below the water cut-off (200 nm) with hermetically sealed VUV photodiodes. Carbon and thin film metal filters provide both the encapsulation against water as well as the band-pass filtering of the diodes allowing spectra to be inferred. No previous data exists in this spectral region due to an effectively infinite absorption of photons after a few microns of water. The difficulty of these measurements is amplified by the requirement to use a detector normally intended for use in a high vacuum system. These measurements are important because the spectral shape in this region is unknown. We expect from published measurements at longer wavelengths that the photon fluence in this region should be high. These measurements may allow a differentiation between several competing sonoluminesence theories.
Nuclear fusion in collapsing bubbles—Is it there? An attempt to repeat an experiment that reported d–d fusion in bubble collapse induced by cavitation in deuterated acetone
Authors: Shapira Dan , Saltmarsh Mike
Published: October 2013

The experiment of Taleyarkhan et al. [Science 295, 1686 (2002)] has been repeated [D. Shapira and M. Saltmarsh, Phys. Rev. Lett. 89, 104302 (2002)] in an attempt to detect the emission of neutrons from d–d fusion during bubble collapse in deuterated acetone. Using the same apparatus and method for bubble seeding and cavitation but a more sophisticated data acquisition system, and a large liquid scintillator detector we find no evidence for 2.5 MeV neutron emission correlated with sonoluminscence from the collapsing bubbles. Any neutron emisssion that might occur is at least four orders of magnitude smaller than that necessary to explain the tritium production reported in Taleyarkhan et al. as being due to d–d fusion. We demonstrate that the proper allowance for random coincidence rates in such experiments requires the simultaneous measurement of the complex time-varying singles rates.
Learning to recognize speech that is spectrally reduced and frequency upshifted
Authors: McCabe Marie E. , Chiu Peter
Published: October 2013

The current study explored to what extent training could ameliorate the deleterious effect of large frequency upshifts in spectrally reduced speech. During each training session, subjects attempted recognition of IEEE sentences spoken by a single talker once and received feedback for their responses. Training sentences were processed to simulate an 8-channel CIS cochlear implant processor with a ‘‘6 mm frequency upshift’’ [Fu and Shannon, J. Acoust. Soc. Am. 105, 1889 (1999)]. Three test sessions were administered to all subjects to assess recognition of sentences (IEEE and HINT), consonants (/aCa/), and vowels (in /hVd/ and /bVt/ contexts) pre-, post-, as well as at the mid-point of training. Four processing conditions (i.e., unprocessed, 8-channel-unshifted, 8-channel-upshifted, and 8-channel-upshifted-and-compressed) were tested for each type of materials. Preliminary data suggest that performance improved for most subjects during training, but there were substantial individual differences in learning rates and asymptotic performance levels. Vowels were more difficult to recognize and showed smaller training-related gains when compared to consonants and sentences. The rank ordering of recognition performance was consistent among the four processing conditions (unshifted best; upshifted-and-compressed intermediate; upshifted worst) for all measures. Data comparing the efficacy of an alternative training method will also be presented.
Estimates of pitch strength for musicians and nonmusicians
Authors: Clarkson Marsha G. , Zettler Cynthia M. , Follmer Michelle J. , Faulk Margaret , Takagi Michael J.
Published: October 2013

To measure the strength of the pitch of iterated rippled noise (IRN), 19 adults were tested in an operant conditioning procedure. Seven adults had music training and currently played an instrument; 12 adults had no training and did not currently play an instrument. To generate IRN, a 500-ms Gaussian noise stimulus was delayed by 5 or 6 ms (pitches of 200 or 166 Hz) and added to the original for 16 iterations. IRN stimuli having one delay were presented repeatedly. On signal trials the delay changed for 6 s. Stimulus level roved from 63-67 dBA (background of 28 dBA). Adults learned to press a button when the stimulus changed. Testing started with IRN stimuli having 0-dB attenuation (i.e., maximal pitch strength). Stimuli having weaker pitches (i.e., progressively greater attenuation applied to the delayed noise) followed. Strength of pitch was quantified as the maximum attenuation for which pitch was discerned. For each subject, threshold attenuation for pitch strength was extrapolated as the 71% point on a psychometric function depicting percent correct performance as a function of attenuation. Mean thresholds revealed that the pitch percept was similar for both nonmusically trained (18.70 dB) and musically trained adults (18.73 dB).
Difference limen for perception of aspiration noise
Authors: Shrivastav Rahul , Sapienza Christine M.
Published: October 2013

The relationship between the vocal acoustic signal and the perception of voice quality has been the subject of much research. An increase in the intensity of aspiration noise is associated with a perception of greater breathiness in voices [Klatt and Klatt (1990); Childers (1993)]. Shrivastav (2001) found that subjective ratings of breathiness could be better predicted using measures calculated from an auditory spectrum of the vocal acoustic signal. These measures were found to vary with changes in the spectral slope as well as the intensity of aspiration noise in voice. The aim of the present experiment was to determine the smallest change in the intensity of aspiration noise that could be perceived by listeners. Six voice continua, with increasing intensity of aspiration noise, were generated using the Klatt synthesizer. Five listeners participated in an adaptive listening test, where these stimuli were presented in pairs and listeners were asked to identify the stimuli in each pair as being same or different. Listener responses were used to determine the difference limen for changes in intensity of aspiration noise in voice. These findings will help understand the perception of breathy voice quality and to develop objective measures for its quantification.
Confidence ratings and awareness measures in word recognition testing
Authors: Goshorn Edward L. , Goshorn Jennifer D.
Published: October 2013

Word recognition tests primarily use percent correct to measure performance. Additional information may be gained by analyzing awareness of accurate perceptions (AA), awareness of errant perceptions (AE), composite awareness (AC), and awareness symmetry (AS). Awareness measures were derived from subjects’ assignment of confidence ratings to a two-item multiple choice response test by designating ‘‘YES’’ or ‘‘NO’’ that their chosen response is accurate. Each response/confidence rating was categorized as a hit, miss, false alarm, or correct rejection. Awareness equations were: AA=hits/(hits+misses); AE=correct rejections/(correct rejections+false alarms); AC=SQRT (AA squared+AE squared); AS=0.707(AA−AE). Thus, AC is the vector to Cartesian coordinates AA, AE; AS is the distance of this point from a diagonal representing symmetrical awareness. Word recognition and awareness was investigated under two signal-to-noise ratios (3 and 6 dB). The Diagnostic Rhyme Test was presented at 50 dBHL to eight normal-hearing adults. Six replicates were obtained. Awareness measures provided additional performance information. Percent correct increased significantly as signal-to-noise ratio improved, but AE decreased and AS did not change significantly.
What is the best method for structural damping identification? Part 1: A survey of methods
Authors: Phani Srikantha , Woodhouse J.
Published: October 2013

Identification of damping matrices from measured frequency response functions is a difficult and error-prone procedure. In practice, the assumption of proportional damping is frequently made without physical justification. Several researchers have suggested identification algorithms for the full damping matrix. A critical review is presented of algorithms which work in the frequency domain. Some novel approaches are included: one new algorithm is based on matrix perturbation theory, while others are hybrid combinations of earlier methods. The sensitivity of the algorithms to measurement noise and other commonly encountered difficulties in realistic measurements is addressed. Representative results from a simulation study comparing all the methods will be presented, leading to recommendations for the choice of identification algorithms in practice.
Active audition for humanoid robots that can listen to three simultaneous talkers
Authors: Okuno Hiroshi G. , Nakadai Kazuhiro
Published: October 2013

The direction-pass filter (DPF) separates sounds originating from a particular direction by using a pair of microphones embedded in each ear of humanoid robot. DPF first extracts harmonic structures from each channel, finds a corresponding pair on right and left channels, and then calculates their interaural phase difference (IPD) and interaural intensity difference (IID). These IPD and IID are matched with reference data obtained by HRTF or by the geometrical relation to determine the sound source direction. The direction obtained by face detection may be used as a candidate for the direction. Finally, all subbands from the direction are collected to synthesize a wave form by inverse FFT. The allowance of collection depends on the direction; narrow (10 deg) at center, while wide (30 deg) at the periphery. This property is called ‘‘auditory fovea’’ and is exploited by DPF actively to improve performance of sound source separation. In addition, a humanoid actively turns its head toward the speaker to listen better. Real-time DPF is implemented by distributed processing with five PCs. Preliminary experiments of active audition in speech recognition of three simultaneous utterances of digits in a normal room is also reported. [Work supported by JSPS.]
Interference cancellation in a waveguide using signal subspace decomposition
Authors: Hursky Paul , Porter Michael B. , Ianniello John P.
Published: October 2013

When the presence of a strong interferer hides a quieter target, an eigenanalysis of the data covariance matrix (also called a principal component analysis) may separate the quieter target from the strong interferer. However, the strong interferer may occupy more than a single principal component when the data covariance matrix is estimated over a time interval during which the strong interferer has moved between multiple spatial resolution cells of the receive array. Also, when the sources to be distinguished are of comparable strength, the principal components fail to isolate the sources, forming mixtures of the sources instead. Results will be presented of applying this technique to the SWellEx-96 experiment data, recorded on both vertical and horizontal arrays, in a shallow water environment off the coast of San Diego. Projections of the principal eigenvectors onto beams and matched-field processing range-depth cells will be shown to indicate to what degree a principal component analysis isolates the different sources. Strategies for coping with source motion will be discussed.
Acoustics and its relation to language: The influence of Dick Bolt
Authors: Stevens Kenneth N.
Published: October 2013

Under the mentoring of Dick Bolt, and the stimulation he provided in the Acoustics Laboratory at MIT, many students were exposed to a range of topics in acoustics, including mechanisms of sound generation, radiation, and propagation, noise control, acoustics of resonators and rooms, human responses to sound, and speech perception under various adverse conditions. As someone who became interested in speech communication, I have recognized that this kind of quantitative background in acoustics is an important requirement for developing models of how humans produce speech, how they perceive and understand speech, and how children acquire these skills. Speech production involves sound sources produced by a nonlinear mechanical system and by noise arising from turbulent airflow. Sound is propagated in a vocal tract with yielding walls, and acoustic coupling is introduced by lossy resonators attached to the vocal tract, including the trachea and the nasal cavity. These acoustic principles of sound generation create an inventory of sound types that give rise to distinctive responses in the ears and brains of listeners. The solid grounding in acoustics provided by Dick Bolt and his leadership have helped in the formation of this linkage between acoustics, speech physiology, linguistics, and human perception.
Backscattering and extinction cross sections of two swimbladdered fishes at the lowest resonance, as modeled by the boundary-element method
Authors: Foote Kenneth G. , Francis David T. I.
Published: October 2013

The boundary-element method has been applied to backscattering and extinction of sound by swimbladdered fish at the lowest, breathing-mode resonance. Corresponding cross sections have been computed for specimens of two representative kinds of swimbladder-bearing fish, namely physostomes and physoclists, which, respectively, possess and lack an external duct. The respective fishes are herring (Clupea harengus) and pollack (Pollachius pollachius), for which swimbladder morphometric data are available. The depth dependences of the cross sections are computed over the range 0–500 m. Comparisons are made with measurements and other modeled results for a number of species. [Work supported by ONR.]
Inversion of the depth, thickness, and absorption coefficient of a layer of fish (anchovies) from transmission loss measurements in the Yellow Sea
Authors: Diachok Orest , Wales Stephen
Published: October 2013

Estimates of bioacoustic parameters of fish (anchovies) and geoacoustic parameters of the bottom were simultaneously inverted from the TL measurements of Qiu et al. (1999) in the Yellow Sea. Replica fields were calculated with BIO-C-SNAP, a C-SNAP based normal mode model, which permits inclusion of bio-absorption layers. The inversion was based on minimizing the rms difference, D, between measured and calculated values of TL at multiple source and receiver depths and ranges, and involved a simultaneous search for bio-layer depth (d), bio-layer thickness, bio-alpha (a), geo-sound speed, and geo-alpha. The resultant values of D were extremely small (1.9 dB at 1.35 kHz). By contrast, inversion calculations, which assumed that all excess attenuation at this site was due to the bottom, resulted in unacceptably large values of D (9.5 dB at 1.35 kHz). The inverted value of d, 6.8 m, was consistent with laboratory measurements of the resonance frequencies of 10-cm anchovies (1.35 kHz at 6 m), the dominant species in the Yellow Sea. The inverted value of a was consistent with previously reported number densities of anchovies in the Yellow Sea. Inverted geoacoustic parameters were within previously reported bounds. [Work supported by ONR.]
Studies on geometrical backscattering models of marine bodies
Authors: Parameswaran Anil Kumar C. , Pai Sajith N. , Soniraj N. , Pillai P. R. Saseendran , Kurian James , Hariharan Supriya M. , Madhavan C. , Mani T. K.
Published: October 2013

The target strength of marine bodies depends on two components—one of them is easy to measure and a good relationship can be established with the target strength value, while the other presents a higher variability that no method can help to reduce, which include the orientation of the fish and a range of environmental conditions and a set of biological facts. Studies on physical models and aspects are expected to provide a clear insight into the issues relating to the target strength variability. Such physical models are developed by converting the physiological shape of the fish into standard and simple geometrical shapes. Data obtained from some of the commercially important species, individually positioned at the center of the acoustic beam, 3 m from the transducer in a test facility were used for the computation of target strength. The target strength value obtained from these reference targets is an indicator of the model performance. Mathematical description of the scattering by some of the species and subsequent comparison with laboratory data have demonstrated that the scattered level by an individual due to a single ping, strongly depends upon size, shape, frequency, material properties, and orientation. Perhaps one of the most notable peculiarities of this work is the simplicity of the approximation and the close agreement between the real world value and the model solution.
A vibration model for frequency analysis of arterial tubes with tissue
Authors: Zhang Xiaoming , Fatemi Mostafa , Greenleaf James F.
Published: October 2013

Vibro-acoustography is a new noncontact imaging method based on the radiation force of ultrasound. We extend this technique for tissue characterization of arterial tubes by vibration techniques. The arterial tube can be excited remotely by ultrasound at its resonant frequencies where the vibration and acoustic emission of the tube can be measurable. From these resonant frequencies, the material properties of the arterial tube can be found. A theory for a tube with tissue is formulated using first-order shear deformation theory to include the effects of transverse shear deformation and rotary inertia. A wave propagation approach is applied for easy handling of the boundary conditions. Experimental studies were carried out on a silicone tube embedded in a cylindrical gel phantom. A confocal transducer is used to produce the radiation force of ultrasound for exciting the tube–phantom structure. The vibration of the tube and the phantom are measured with a laser vibrometry system. The fundamental mode of a tube–phantom structure is well excited by the radiation force of ultrasound, and was measured to be 81.8 Hz, which is close to the theoretical prediction of 83.3 Hz. Both excitation and measurement are remote and noncontact, important attributes for future study of arteries.
Analysis of levels of support and resonance demonstrated by an elite singing teacher
Authors: Scherer Ronald C. , Radhakrishnan Nandhakumar , Poulimenos Andreas
Published: October 2013

This was a study of levels of singing expertise demonstrated by an elite operatic singer and teacher. This approach may prove advantageous because the teacher demonstrates what he thinks is important, not what the nonsinging scientist thinks should be important. Two pedagogical sequences were studied: (1) the location of support—glottis (poor), chest (better), abdomen (best); (2) locations of resonance—hard palate/straight tone (poor), mouth (better), sinus/head (best). Measures were obtained for a single frequency (196 Hz), the vowel /ae/, and for mezzo-forte loudness using the /pae pae pae/ technique. Sequence differences: The support sequence was characterized by formant frequency lowering suggestive of vocal tract lengthening. The resonance sequence was characterized by flow (AC, mean flow) and abduction increases. Sequence similarities: The best locations had the widest F2 bandwidths. The better and best locations had the largest dB difference between F2 and F3. Although acoustic power increased through the sequences, the acoustic efficiency was not a discriminating factor. Open and speed quotients were not differentiating. The flow resistance was highest and aerodynamic power the lowest for the first of each sequence. Combined data: The maximum flow declination rate correlated highly with the AC flow (r=−0.92) and SPL (r=0.901).
Sex and the singer: Gender categorization aspects of singing voice
Authors: Ternström Sten
Published: October 2013

The singing voice exhibits many systematic differences by gender and age. The physiological differences between the voice organs of males, females, and children are well known and give rise to several acoustic differences, including acoustic power, pitch range, and spectral distribution. Vocal artists often strive to widen their range of expression, and it is not uncommon for males to sing in a femalelike register, as in counter tenors and in some pop/rock genres. The opposite, however, is quite rare. While ambiguous or contradictory gender in speech is usually a social disadvantage, in singing it can be a desired effect. The physical differences in singing voice production between males and females are reviewed in detail. Some interesting borderline cases are examined from an acoustic standpoint.
Comparison of hearing and voicing ranges in singing
Authors: Hunter Eric J. , Titze Ingo R.
Published: October 2013

The spectral and dynamic ranges of the human voice of professional and nonprofessional vocalists were compared to the auditory hearing and feeling thresholds at a distance of one meter. In order to compare these, an analysis was done in true dB SPL, not just relative dB as is usually done in speech analysis. The methodology of converting the recorded acoustic signal to absolute pressure units was described. The human voice range of a professional vocalist appeared to match the dynamic range of the auditory system at some frequencies. In particular, it was demonstrated that professional vocalists were able to make use of the most sensitive part of the hearing thresholds (around 4 kHz) through the use of a learned vocal ring or singer’s formant. [Work sponsored by NIDCD.]
Acoustical analysis and modeling of reciprocating compressors, noise produced by gas pulsation, using four-pole method. II
Authors: Herfat Ali T. , Seel Robert V.
Published: October 2013

Presented in Paper II is the noise analysis of reciprocating compressors (such as air conditioning and refrigeration reciprocating compressors) using the four-pole method. The gas pulsation noise inside compressor head cavities, mufflers, and through-valves can be analyzed by applying the FPM. This method formulates the characteristics of acoustic elements by establishing a relationship between their input and output gas pressures and volume flow rates. When the acoustic elements in the system (compressor) are connected at points between them, the FPM allows an easy assembly of element equations to obtain system acoustical model.
Turbulent pressure statistics in the atmospheric boundary layer from large-eddy simulation
Authors: Miles Natasha L. , Wyngaard John C. , Otte Martin J.
Published: October 2013

Turbulent pressure fluctuations advected past a sensor contribute to wind noise and can thus significantly degrade acoustic signals. In this study, large-eddy simulation is used to calculate the turbulent pressure field in three types of atmospheric boundary layers. A Poisson equation is used to represent turbulent pressure as the sum of mean-shear, turbulence-turbulence, subfilter-scale, Coriolis, and buoyancy parts. At variance-containing scales in the free-convection case, turbulent-turbulent pressure dominates over the entire boundary layer. It dominates also up to midlayer in the forced-convection case; above that mean-shear pressure dominates. In the stable case mean-shear pressure dominates over the entire layer. Part of the inertial subrange of the pressure spectrum is resolved in the forced- and free-convection cases; it is dominated by the turbulence-turbulence pressure and has a three-dimensional spectral constant of 4.0. This agrees well with quasi-Gaussian predictions but is a factor of 2 less than results from direct numerical simulations at moderate Reynolds numbers. Although past measurements of turbulent pressure have been hampered by instrumental problems, such measurements that could be used to determine the inertial subrange pressure spectral constant would be most useful. [Work supported in part by the U.S. Army through the National Center for Physical Acoustics.]
Performance comparisons of microphone windscreens
Authors: Maniet Edward R.
Published: October 2013

The effectiveness of microphone windscreens is evaluated using a unique low-noise wind tunnel facility that enables controlled, repeatable experiments to be performed [E. R. Maniet, Jr., J. Acoust. Soc. Am. 111, 2373–2374 (2002)]. Wind noise levels are characterized as a function of frequency, mean wind velocity and turbulent velocity component for several standard windscreen types, including porous foam balls and hollow shell windscreens. The experimental results are used to develop a model for wind noise levels based upon Strasbergs [J. Acoust. Soc. Am. 83, 544–548 (1988)] dimensional scaling analysis. Also presented is a comparison of microphone wind noise levels for omnidirectional and supercardioid microphone elements.
An investigation of outdoor wind noise reduction by spherical windscreens
Authors: Webster Jeremy , Raspet Richard
Published: October 2013

Phelps [RCA Rev. 3, 203–212 (1938)] proposed that wind noise pressure fluctuations measured at the center of a spherical wind screen are the area averages resultant of the pressure fluctuations at the surface of the screen. If this hypothesis is applied to the steady state pressure distribution around a spherical windscreen, good agreement between data and theory should be achieved for turblence structures which are large compared to the windscreen. In this paper we report on experiments which extend the work done by Morgan (Ph.D. dissertation, University of Mississippi, 1992) on spherical windscreens. In this experiment, probe microphones placed within reticulated foam windscreens were used in conjunction with anemometers placed directly in the airflow near the windscreen to investigate the correlation between incident flow and the resulting pressure measurements throughout the foam.
Measuring immeasurable sound pressure levels
Authors: Gilbert Kenneth E. , Talmadge Carrick L. , Di Xiao
Published: October 2013

The AWSUM(K) processors developed by R. A. Wagstaff make possible the detection of narrowband signals deeply buried in wind noise. However, the signal-to-noise ratio one obtains with the AWSUM(K) processors is not the ratio of signal power to noise power, but a nonphysical ratio that depends on the coherence of the signal and the noise distribution. A recently developed theory, used in conjunction with the AWSUM(K) processors, allows the recovery of a physically meaningful signal-to-noise ratio, even when wind noise prevents a direct measurement of the signal. Thus, it is possible not only to detect a signal buried in wind noise, but also to estimate the physical sound pressure level. Such a capability can be important for identifying a source using, for example, the ratio of harmonic amplitudes produced by a particular engine in a ground vehicle. The theory for recovering the physical sound pressure level will be discussed and applications to field data will be presented.
Jozef Zwislocki’s integrated approach to psychoacoustics
Authors: Hellman Rhona P.
Published: October 2013

Early in the psychophysics phase of his illustrious and varied scientific career, Joe Zwislocki recognized that loudness is of key importance for understanding signal processing by the auditory system. To better comprehend just how loudness can be incorporated into basic auditory theory, Joe sought to determine the loudness-intensity relation down to near threshold levels. Together with Hellman, this aim motivated the development of absolute magnitude-scaling procedures. Later, Joe demonstrated that absolute scaling yields results compatible with nonmetric measures of loudness additivity. His search for a comprehensive theory of auditory function led him to deduce that both the observed proportionality between loudness and sound intensity near threshold and the compressive nonlinearity in the loudness function at moderate-to-high levels are generated by the peripheral auditory system. These innovative concepts were incorporated into his theoretical analyses of temporal summation and central masking. They also provided the basis of a loudness model formulated to describe loudness growth in quiet and in noise. Not only are Joe Zwislocki’s extraordinary insights compatible with recent loudness measures, his loudness model can be extended successfully to predict the growth of loudness characteristic of cochlear-impaired hearing. [Work supported by NEDO, Japan.]