bluuebunny/crossref_metadata_2025 huggingface dataset:
Analog of the Fresnel reflection and refraction coefficients for smoothed boundary between two liquid medias
Authors: Maltsev Nick
Published: February 2014

The Fresnel formulas for reflection and refraction coefficients at the boundary between two liquid medias assume discontinuity of media properties at the boundary. In many physical applications this assumption is too strong. A recent report considers a model problem with the following properties: n2(z)=3D[n21exp(αz)+n20exp(−αz)/e xp(αz)+exp(−αz)]ρ(z)=3D[ρ1exp(αz)+ρ0exp(−αz)/exp(αz)+exp(−αz)] which are representing a smoothed boundary. An exact solution of Euler equations is derived in a form of plane waves and fast converging series. Physical effects are investigated for reflection and refraction of plane and spherical harmonic and pulse waves in such a model.
Acoustic method for measurements of the coolant boiling on heat generating surfaces
Authors: Boldin Sergey V. , Kolpakov Andrei B.
Published: February 2014

Acoustic method for determining the period of different duties of the coolant boiling is presented. The foundation of this method is the damping phenomenon of ultrasonic waves travelling through the boundary between different media (steam–liquid) differing widely in their acoustic properties. The experimental setup and the results of the investigations into the period of water film boiling on spherical samples are also presented. The emphasis is on aluminum and copper samples differing in their diameter (4, 5, and 8 mm) for temperature interval Δt=180–690°C at atmospheric pressure and different surrounding water temperatures (Δt=15–75°C). By analyzing the result, it is illustrated that the precision of the method is well above that of thermocouples. It is demonstrated that the measurements using this acoustic method excludes the errors typical of traditional methods.
Auditory integration of bandpass filtered sentences
Authors: Spehar Brent , Tye-Murray Nancy , Sommers Mitchell
Published: February 2014

Auditory integration is the ability to merge auditory inputs received from different channels into a single percept. The present experiments investigated auditory integration of speech using lists of filtered CUNY sentences. Within each set, separate filtering preserved either low-frequency or high-frequency spectral information. The three sets of bandwidths were 550–750 and 1650–2250 Hz; 550–800 and 1650–2350 Hz; and 550–850 and 1650–2450 Hz. Four conditions were tested for each set of bandwidths: (1) LF only: (2) HF only; (3) LF–HF monotic (LF and HF filtered sentences presented simultaneously to one ear); and (4) LF–HF dichotic (LF sentence presented to one ear, HF sentences presented simultaneously to the other ear). Scores in the single band conditions were near floor. Scores in the monotic conditions were significantly greater than would be predicted from an additive combination of the single band conditions (i.e., superadditive effects). Scores in the LF–HF dichotic condition were significantly less than those in the LF–HF monotic condition but were greater than those for either the LF-only or HF-only conditions. Taken together, these findings suggest that listeners are able to integrate information across two auditory channels but that this integration is less than optimal. Implications of these results for theories of sensory integration are discussed.
Prediction from pure-tone thresholds of the effects of interaural time delays on the speech reception threshold in a multisource environment
Authors: Culling John F. , Hawley Monica L. , Litovsky Ruth Y. , Colburn H. Steven
Published: February 2014

Speech reception thresholds (SRTs) in NoSπ are predictable from the articulation index and pure-tone thresholds; the effective level of masking noise is assumed to decrease by the size of the binaural masking level difference (BMLD) at each frequency [H. Levitt and L. R. Rabiner, J. Acoust. Soc. Am. 42, 820–829 (1967)]. This assumption was tested for the effect of interaural time delays in multisource environments. Thresholds for diotic pure tones were measured at third-octave intervals for three simultaneous, speech-shaped-noise interferers distributed in perceived azimuth using the temporal components of head-related impulse responses (ITD-only maskers). One masker (444) had all interferers directly ahead (position 4 of 7 in an arc). Pure-tone thresholds for other spatial configurations subtracted from corresponding thresholds in 444 gave BMLDs for each configuration and frequency. Copies of the 444 masker were filtered, attenuating the noise across frequency in accordance with the appropriate BMLDs for each configuration (367, 567, and 777). SRTs for speech from position 4 against each ITD-only masker and its corresponding filtered masker were measured and compared. SRTs matched for distributed configurations (367 and 567), but the filtered masker yielded lower SRTs than the ITD-only masker when all interferers were colocated away from the target speech (777).
Speech enhancement based on aspects of the auditory process
Authors: Arehart Kathryn , Hansen John
Published: February 2014

The effectiveness of a speech enhancement algorithm on both the intelligibility and quality of speech-in-noise by listeners with cochlear-based hearing loss of moderate severity was tested. The enhancement scheme employed is based on an approach that uses the auditory masked threshold in conjunction with a version of spectral subtraction to adjust the parameters used in the subtraction process based on the masked threshold of the noise across the frequency spectrum. The ability of listeners with hearing loss to perceive stimuli from the nonsense syllable test (NST) and from the diagnostic rhyme test (DRT) in the presence of communication channel noise and automobile highway noise was measured at several different signal-to-noise ratios in both degraded and enhanced conditions. The algorithm effectiveness was also assessed through subjective ratings of speech quality. Results of the listening tests show that some but not all of the listeners with hearing loss benefited from the enhancement algorithm. We will discuss how the algorithm—which is based on models of auditory processing in normal-hearing listeners—might be reformulated to address specific auditory deficits in listeners with cochlear hearing loss. [Work supported by National Organization of Hearing Research, Council on Research and Creative Work, University of Colorado.]
Speech task affects quality of articulation in Parkinsonian dysarthria
Authors: Van Lancker Diana , Tafuro Lisa , Kempler Daniel
Published: February 2014

To examine the effect of speech task on articulation, this study assessed speech in a dysarthric patient across four production tasks: spontaneous speech, repetition, reading, and repeated singing, using the same phrases for all tasks. The study design ensured that all phrases were heard in all production tasks, but no listener heard any phrase twice. Data were obtained from listeners’ identification and difficulty rating scores, perceptual judgments, and acoustic analyses. Items produced spontaneously were significantly less intelligible than the same phrases produced in the other three tasks (29% vs 79% correct identification). Spontaneous speech was also rated as more difficult to understand than other speech tasks. While relative intensity and word duration were not linked to intelligibility, dysfluencies and distorted articulatory/resonance patterns were found significantly more often in spontaneous than in the other tasks. These indications of variable speech production efficiency were found in a patient with Parkinson’s disease, which is associated with basal ganglia impairment. These results can be related to recent models of normal basal ganglia function, which emphasize motor planning and monitoring. One explanation for the different articulation patterns reported here may be the availability of an external ‘‘model’’ which aids planning in the reading and repetition tasks.
Backward-masking and syllable–rate thresholds have only minor associations with reading achievement, compared to several cognitive and linguistic measures
Authors: Kidd Gary R. , Watson Charles S. , Lowther Andrya , David Eddins A.
Published: February 2014

Rate thresholds for syllable–sequences [the Indiana Test of Auditory Memory and Processing Rate—ITAMPR, Watson and Eddins, J. Acoust. Soc. Am. 105, 1236 (1999)] and backward-masking thresholds, were obtained from 220 children. These measures were only weakly associated with reading achievement and with speech recognition. Other tests, however, accounted for roughly half of the variance in teacher-assigned reading grades. Results of 36 sensory, perceptual processing, linguistic, and cognitive tests and subtests administered to the population from which these children were obtained (approximately 98% of the 472 entering first-graders in Benton County Indiana from 1995 to 1997), were well-described by four factors. These factors and the systematic variance in reading achievement associated with them were: Factor 1 (31%): reading-related skills (phonological awareness, letter- and word-identification); Factor 2 (12%): visual cognition (visual perceptual abilities, spatial perception, visual memory); Factor 3 (9%): verbal cognition (language development, vocabulary, verbal concepts); and Factor 4 (1%): speech processing (the ability to understand speech, measured under difficult listening conditions). These data are inconsistent with the hypothesis that temporal processing deficits are causally related to reading impairments. [Work supported by NIH/NIDCD, Indiana Univ., and the Benton County (Indiana) School Corporation.]
Hearing research and the ASA
Authors: Hirsh Ira J.
Published: February 2014

In the same year that the ASA was founded, its first President, Harvey Fletcher, had summarized research on Speech and Hearing (1929). The knowledge then about auditory sensitivity, discrimination of intensity and frequency, masking, auditory localization and associated binaural effects, loudness and annoyance and speech recognition is very humbling now, 72 years later. The vacuum-tube amplifier, soon to be replaced by the transistor, enabled auditory psychophysics (or psychological acoustics or, more briefly, psychoacoustics) to measure and control sounds, and extend detailed knowledge of hearing. Fletcher’s group at Bell Telephone Laboratories consisted of people who were equally comfortable with the engineering aspects of sound making and control, and the psychophysical idea of putting concepts to test with human listeners. From the early 1930’s onward, psychoacoustics has contributed much information about auditory processes and theoretical models of application to mathematical, physical and biological systems. The juxtaposition of physical and psychological sciences was just right for the Acoustical Society. Publications, discussions at technical meetings, and opportunities to serve the Society as well as the community at large were enriched by scientists of different training. And that tradition stood well as psychoacoustics reached across further to speech and music perception.
Universal spectral properties of complex structures and random matrix theory
Authors: Legrand Olivier
Published: February 2014

It is now widely recognized that universal properties of spectral fluctuations for waves in complex or disordered cavities can be accounted for by random matrix theory (RMT). These properties are manifested in short- and long-range correlations of the spectral response. In nondissipative systems, the wave evolution operator may be related to a Hermitian matrix. The lack of information about the system due to complexity or disorder is translated in RMT through statistical ensembles of such matrices (e.g., the Gaussian orthogonal ensemble) in which the only relevant features of the system are its modal density and its global symmetries, such as time-reversal invariance. Here, it is shown that, beyond quantum or electromagnetic systems, these concepts are relevant to acoustics and elastodynamics by presenting the few basics of RMT and a simple argument yielding Wigner’s surmise for the distribution of frequency spacings between neighboring resonances, exemplifying level repulsion. Within the high-frequency geometrical limit of rays, a semiclassical argument due to Berry is sketched for the spectral long-range reduction of fluctuations named spectral rigidity. This is one of the most convincing arguments in favor of the conjecture which stipulates that wave systems which are chaotic in the limit of rays should follow RMT.
Equipartition and mean square responses in large undamped structures
Authors: L. Weaver Richard
Published: February 2014

It is shown that the smoothed spectral energy density at one point in a large complex structure may be approximated in a simple manner as proportional to the frequency-smoothed admittance at the receiver and at the source, and inversely proportional to the global modal density. Thus the mean square response may be estimated with little computational burden, knowing only the local properties at source and receiver and the size of the system. The approximate predictions are compared with the results of direct numerical simulations and found to be accurate except in the presence of Anderson localization. [Work supported by NSF.]
Numerical evidence for the Wigner surmise
Authors: Brown Alastair W. M. , Langley Robin S.
Published: February 2014

The ‘‘Wigner surmise’’ states that the probability density function of the eigenvalue spacings of a random system follows the Rayleigh distribution. Thus far the surmise has been proved mathematically only for systems described by the ‘‘Gaussian orthogonal ensemble (GOE)’’ of matrices. Random matrices associated with practical engineering structures do not have the form of the GOE, and yet there is strong evidence that the Wigner surmise may still be applicable. The aim of the present work is to determine the conditions under which the natural frequency spacings of an engineering structure will have a Rayleigh distribution. To this end, numerical simulations have been performed for a simply supported flat plate, a simply supported string, and a system composed of two spring-coupled plates. In each case the structure was made uncertain by the addition of randomly placed masses or springs. It was found that the validity of the Wigner surmise depends on three factors: (i) the level of randomness, (ii) the degree of symmetry, and (iii) for the coupled system, on the strength of coupling between the two subsystems. The relative importance of these factors has been quantified for the examples considered.
Level repulsion and spectral rigidity in reverberant microwave cavities
Authors: Barthélemy Jérôme , Legrand Olivier , Mortessagne Fabrice
Published: February 2014

It is now widely acknowledged that spectral fluctuations of waves in complex systems are well described by random matrix theory. As a model system for complex or disordered reverberant cavities, a two-dimensional reverberant microwave cavity is investigated in a high-frequency regime where resonances are still below the large modal overlap regime. Modifying the cavity through the inclusion of scatterers makes the dynamics of rays chaotic or diffusive, thus rendering its spectral response strongly correlated over short and long frequency range. This can be illustrated by the distribution of frequency spacings between neighboring resonances which clearly exhibits the so-called level repulsion. Longer range correlations are demonstrated by considering an averaged time response of the cavity for times shorter than the inverse modal density. By scaling down the hyperfrequency range to audio frequency range and proper sampling, an auditive perception of the differences between the spectral responses of a regular (pure rectangular cavity) and a complex reverberant cavity is demonstrated.
Intensity maps of systems with complex eigenvalues
Authors: Li Xianhui , McDaniel J. Gregory
Published: February 2014

Maps of structural and acoustic intensities have been widely used to understand the dynamics of complex systems with time-harmonic forcing. Identification of the dominant power paths helps designers introduce modifications that control them. For example, damping treatments can be placed to dissipate the maximum power and effectively reduce unwanted vibration. Similarly, high-impedance elements can be configured to block the power flow to sensitive system components. In typical applications, the applied force distributions are chosen to be representative of the expected in situ forces. In this presentation, an extension is described that allows one to construct an intensity map for one mode of a system with complex eigenvalues. This work is motivated by the need to map structural intensity for a squealing mode of an automotive brake system, but it would also be useful in understanding modes of highly damped systems. The proposed extension isolates such modes by choosing a time-harmonic force vector that is parallel to the complex-valued eigenvector, in which case a single mode is isolated. Numerical examples will illustrate the features of this map and contrast it to conventional maps. [Work supported by NSF.]
Speech synthesis by mapping articulator movement patterns to a shape-based area function model of the vocal tract
Authors: Story Brad H.
Published: February 2014

Orthogonal components of tongue and lip displacement data from two speakers in the Univ. of Wisconsin x-ray microbeam database (Westbury, UW–Madison, 1994) were determined with a principal components analysis (PCA). The PCA was performed on steady-state vowels and at the sentence level. For every time frame, the analysis consisted of fitting a cubic spline to four pellet points on the tongue and to modified positions for the incisor pellet and the upper and lower lip pellets. The modifications brought the pellet positions to the level of the airway. The PCA was performed over a collection of frames and showed that the first two (most significant in terms of variance) orthogonal components were similar in shape to those determined from a set of MRI-based vocal tract area functions [Story and Titze, J. Phonetics 26, 223–260 (1998)]. Next, the coefficients determined from a sentence-level PCA of the microbeam data were normalized and used as input to an area function model based also on orthogonal shaping components. The result is that articulatory movement patterns from data in the x-ray microbeam database can be transformed to movement patterns of a vocal tract area function to reproduce (synthesize) the original sentence. [Work supported by NIH R01-DC04789-01.]
Variant frequency in American English flap production
Authors: Patterson David , Connine Cynthia M.
Published: February 2014

This study examined the dominance of flaps as the standard pronunciation of medial /t/ in American English compared to other allophonic variants, namely [t] or glottal stops. A large conversation speech database was utilized to generate statistics about the frequency of occurrence of medial flaps. Results confirmed the dominance of flapping in American English. Interesting effects of morphological complexity and lexical frequency were found to have an effect on the allophonic variant distribution patterns. Morphologically complex words (e.g., waiting) showed a redistribution of variants from flaps to [t] compared to dominant flapping in morphologically simple words (e.g., quota). A similar redistribution of variant production occurred for low-frequency words compared with high-frequency words. The pattern of results was maintained when the morphological analyses were conducted separately for low- and high-frequency words. A second analysis examining vowel length prior to medial /t/ an medial /d/ showed that the vowel preceding medial /d/ tended to be longer, even though both stops were pronounced as a flap. The relevance of the variability in flap occurrence was discussed in relation to rule based phonology and in reference to potential processes in recognizing spoken words.
Coproduction in VCV disyllables produced by children and adults: A comparison of anticipatory and carryover effects
Authors: Gelfer Carole , Bell-Berti Fredericka
Published: February 2014

Previously, we reported spectral data showing similar patterns of coproduction for children and adults in schwa preceding a stop + vowel [C. E. Gelfer and F. Bell-Berti, J. Acoust. Soc. Am. 107, 2855 (2000)]. The absence of a clear developmental pattern differs from some previous reports of children’s speech gestures [e.g., Nittrouer et al., J. Speech Hear. Res. 39, 379–389 (1996)]. For both groups in our study, the alveolar place blocked the effects of the following vowel, while vowel effects were observed in the context of the bilabial and velar stops. These results also suggested that the gestures associated with a segment are an integrated unit: The lip gesture associated with the vowel seems to be delayed until the lingual gesture associated with the alveolar closure occurs. This study continues our investigation of these patterns and compares the productions of children 4 to 8 years old to those of adults. Here we study carryover as well as anticipatory effects, using VC+schwa and schwa+CV utterances, where V is /i/ or /u/ and C is /p,b,t,d,k,g/. F2 measurements taken at various intervals during schwa will be used as an index of these effects and the presence or absence of developmental trends.
Stability of intra- and inter-articulatory timing for contiguous jaw cycles
Authors: Shaiman Susan
Published: February 2014

The current study compared how contiguous cycles of jaw movement, composed of different phonetic contexts, were impacted by global changes across the entire utterance and by more local, phonetic changes. Specifically, articulatory kinematics were examined to determine if patterns of intra- and inter-articulatory timing across manipulations of speaking rate were maintained for both jaw cycles, while being reorganized for phonetic changes (i.e., in coda composition) specific to the second jaw cycle. Five normal speakers repeated the syllables /paep/, /paeps/, and /paepst/, embedded in the carrier phrase, ‘‘Now say —- again,’’ using slow, normal, and fast speaking rates. Changes in speaking rate impacted the utterance globally, with both jaw cycles evidencing similar patterns of kinematic change. Conversely, changes in coda composition for the second jaw cycle resulted in local changes to that cycle only, without impacting the first cycle. Individual speakers demonstrated distinct, but systematic patterns of intra- and interarticulatory timing as a function of coda composition and speaking rate. While functional groupings of articulators may be broken apart and reconfigured for subsequent movements, global suprasegmental changes result in timing patterns which are consistent across these reorganized functional groupings. [Work supported by CRDF–University of Pittsburgh and NSERC.]
Vocal tract length development: MRI procedures
Authors: Vorperian Houri K. , Kalina Cliff M. , Kent Ray D. , Yandell Brian S. , Gentry Lindell R.
Published: February 2014

From infancy to adulthood, vocal tract length increases by about twofold. The purpose of this study is to assess the developmental changes in the various hard and soft tissue structures in the vicinity of the vocal tract that contribute toward its length. Magnetic resonance images (MRI) from children (birth to 6 years) and adults were used since MRI provides detailed visualization of the soft tissues in the oral and pharyngeal regions along with adequate visualizations of related bony and cartilaginous structures. Using previously established measurement procedures (Vorperian et al., 1999), the following structures were measured: lip thickness, hard and soft palate length, tongue length, oro- and naso-pharyngeal length, mandibular length, and position of the hyoid bone and larynx in relation to the nasal spine. Findings will be discussed in terms of: (a) the relative contribution of the various structures toward vocal tract length and how the extent of contribution changes with age; and (b) the relative and relational growth of the different structures. Findings provide normative data on the various vocal tract structures measured; also, they contribute toward understanding the anatomic changes that may be a substrate to speech emergence and development. [Work supported by NIDCD Grant No. R03-DC 4362.]
Developmental changes in the production of obstruents. I. An acoustic analysis of voiceless obstruents
Authors: Nissen Shawn L. , Frieda Elaina , McGory Julie , Fox Robert A. , Friedman Kelly , Rosenbauer Kim
Published: February 2014

This is one of two papers measuring the acoustic structure of prevocalic obstruents in children and adults, examining age-related differences. This study looks at voiceless consonants. The data will be drawn in part from a large scale acoustic database developed in collaboration with ATR and include word initial voiceless obstruents from real words (in a range of phonetic contexts) as well as a series of syllable initial intervocalic obstruents drawn from nonwords (in the constrained context [a —a]). Of particular interest are the intervocalic consonants, as there is a relatively limited amount of knowledge on the acoustic nature of obstruents produced by children in this position. Utterances elicited from speakers ranging in age from 3–70 years were recorded online in a quiet room environment using high quality microphones and direct analog-to-digital conversion to computer disk. Acoustic measurements collected include the spectral moments [K. Forrest et al., J. Acoust. Soc. Am. 84, 115–123 (1988)], formant transitions, VOT, closure duration (for stops), noise duration (for fricatives), and preceding vowel duration. Acoustic differences obtained will be discussed as a function of age group, gender, and phonetic context. [Work supported by an INRS Award from NIH and research funding from ATR (Fox, P.I.).]
Automated head and transducer support (AHATS) system for ultrasound data collection
Authors: Lundberg Andrew Jon , Stone Maureen , Davis Edward P.
Published: February 2014

Ultrasound has been shown to be a useful tool for measuring the vocal tract during speech. This is largely because ultrasound is noninvasive, safe, relatively inexpensive, and can capture 2-D images at video frame rates. In previous work, a head and transducer support (HATS) system was developed which allowed stable positioning of the subject and the ultrasound transducer, and led to improved measurement accuracy. We have now automated the system (AHATS), replacing the manually positioned transducer holder with a computer controlled robotic arm that supports automatic positioning and logging of the transducer holder. Computer controlled positioning of the data collection device increases the speed at which we can collect ultrasound data sets with multiple image planes. It also increases our confidence in the accuracy of combined 3-D data sets (composed of multiple 2-D data collections within the 3-D space), and facilitates efficient 3-D reconstructions which require data collection positions to be optimized for individual subjects. [Supported by NIH-NIDCD DC01758 and Northrop Grumman, Inc.]
On the error distributions of speech production data
Authors: Kollia H. Betty , Jorgenson Jay
Published: February 2014

In previous works [J. Acoust. Soc. Am. 102, 3164–3165(A) (1997) and 105, 2(A), 1354(A) (1999)] the following problem was presented. Assuming that two-dimensional data (such as that obtained in most kinematic speech studies) are approximately linear with error terms about the mean coming from a quadratic—rather than a normal—distribution, then the regression analysis is best performed using a maximum likelihood stochastic linear regression methodology, instead of a least squares methodology. It was shown that one could expect, among other results, sharper confidence intervals about the mean. It remains to be determined if the distribution of error terms about the mean (for speech production data) is better described by a quadratic than by a normal distribution. Indeed, using the classical Kolmogorov–Smirnov test, it is shown that in many cases one can assume that the error terms follow a quadratic distribution and that at any reasonable confidence level one can reject the hypothesis that error terms follow a normal distribution. The result of this test substantiates the theoretical premise assumed in the earlier work.
Fundamental frequency change as a result of repeated corrections
Authors: Menezes Caroline M. , Fujimura Osamu
Published: February 2014

Articulatory and intonational patterns were studied, using a simulated discourse-like paradigm, in which subjects were asked to make repeated corrections of a three-digit sequence street address consisting of fives and nines and Pine. Articulatory and acoustic signals were collected using the x-ray microbeam system for four native American English speakers at the University of Wisconsin. Based on the C/D model, a linear pulse train representing the rhythmic organization of an utterance can be inferred by analyzing jaw movements as syllable magnitude over time. Fundamental frequency contours were examined to study the effects of phrasing as a result of contrastive emphasis (digit corrected is assumed to have contrastive emphasis) and repeated corrections in the three-digit sequence. These patterns were then studied in relation to the jaw movement patterns. The data show that subjects change speech strategies when they need to make a correction or repeated corrections [Erickson et al. (1992)]. The word in particular that is corrected is made prominent from the rest of the utterance, in all repeated corrections. Effects of speaker variability will also be discussed. [This work has been supported in part by NSF (BCS-9977018) and ATR/MIC, Japan.]
Channel characterization for high-frequency acoustic communications
Authors: Porter Michael B. , McDonald Vincent K. , Baxley Paul A. , Rice Joseph A.
Published: February 2014

Undersea internets or seawebs are currently being developed and tested for a variety of applications in which some kind of device (e.g., an ADCP, hydrophone, vertical line array, autonomous undersea vehicle) needs to pass information to another (e.g., radio buoy, surface ship). The underlying physical layer is often an acoustic modem. However, the ocean medium, though it can carry acoustic waves with remarkable fidelity to long distances, is also somewhat unreliable. Storms may cause network outages by both driving up the ambient noise and corrupting the clarity of the acoustic mirror formed by the air–sea interface. Meanwhile, as the SONAR community is well aware, the refractive effects of the ocean can lead to sweet spots and dead zones (shadows). The required sound output of a modem varies accordingly. We have been conducting a program (SignalEx) to: (1) better understand propagation in the high-frequency range (around 10 kHz), currently of greatest interest for modems; (2) understand how the environment degrades different modulation schemes; and (3) develop more robust schemes or select the appropriate scheme for the given conditions. This talk will summarize results from recent SignalEx sea tests.
Characterization of very shallow water time-varying acoustic channels
Authors: Brady David
Published: February 2014

An accurate and tractable model for communication channel dynamics is important for the design of adaptive transceivers. This channel model should account for hardware-related dynamics (oscillator drift) as well as environmental effects (platform and scatterer motion). Typically, the parameters of a channel model are estimated using a time series of noisy channel outputs, due to a known time-series excitation. Additionally, when ancillary environmental measurements are available (wind, current, water-column history), the channel parameter estimators should also incorporate this information. An investigation of channel dependence on environmental measurements is the main focus of this talk. In this work the environmental dependence of a multipath acoustic communication channel on environmental measurements is investigated for a very shallow water environment. Conjectures about this dependence are confirmed by a data from multiday experiment in the New England area. Observations for this experiment include noisy channel outputs, wind, current, and water-column measurements.
Environmental factors that affect acoustic propagation and underwater communications in the surf zone
Authors: Deane Grant B.
Published: February 2014

Acoustic signals propagating through the surf zone are strongly influenced by scattering from surface gravity waves, reflection from the sea-floor and absorption and scattering by the clouds of bubbles entrained by breaking surf. Shoaling gravity waves create focal regions in surface-reflected wavefronts, and introduce systematic variations in wavefront travel times. Both of these surface effects are important to the design and operation of very shallow water communication systems. To study them, an experiment was conducted 40 m north of Scripps Pier in November of 2000. Single cycle acoustic pulses centered at 10 kHz and phase modulated pseudorandom sequences were repetitively transmitted over a 40 m path to a 3-element vertical hydrophone array in 5 m deep water. Environmental characterization was provided by a 10-element pressure array mounted along the propagation path to measure the shoaling gravity wave field. The results of this experiment and additional experimental data collected at Scripps Pier during the Fall of 2000 over several regimes of environmental conditions (the SZATE experiment) will be discussed. [Work supported by ONR.]
Statistical modeling of very shallow water/surf zone underwater communications channels
Authors: Green Dale
Published: February 2014

The performance of a phase-coherent underwater communications system in a nonstationary environment is dependent in large measure on the temporal coherence of each of the principal components (paths) over which the signal arrives at the receiver. It was found that even though a path may bring considerable energy to the receiver, if it is a rapidly scintillating arrival (i.e., short temporal coherence), it can degrade the performance of a channel equalizer in the same way that a noise burst at the same delay and power would do. A method is described for estimating the temporal and inter-lag coherence properties of the time-varying channel impulse response. The method is applied to an extensive database reflecting a recent experiment in the VSW/SZ channel at the Scripps’ pier in California. A statistics-based channel model is described, which, when used in a real-time simulator, provides validated analog realizations of signals of opportunity passing through the channel. [Work supported by ONR, Code 32 OE, under the direction of Dr. Tom Swean.]
Intensity variations of high-frequency broadband acoustic scatter from sea surface in coastal environment
Authors: Lenain Luc , Badiey Mohsen , Forsythe Stephen E.
Published: February 2014

Data from a broadband propagation experiment for mid-to-high-frequency (0.6–18 kHz) in the very shallow water (15 m) of Delaware Bay are studied over a wide range of environmental conditions. Physical parameters such as temperature and salinity as well as hydrodynamic parameters such as surface waves, tide, and current were measured simultaneously with acoustic transmissions. Three hydrophones were utilized to record the received signals in the source/receiver range of 389 m. A beamforming technique is used to allow examination of the arrival angle as a function of arrival time. Statistical correction is applied to normalize the arrival signals and to correct for sidelobes. Four ray paths having only one interaction and scattered by the sea surface into micro-multipaths are considered for analysis. The focus of this study is on the evaluation of these signals as a function of sea surface roughness during several tide cycles. To analyze the data, PE calculations are conducted and parameter studies for different wind speeds are performed using the measured sea surface wave spectra. [Work supported by ONR and Sea Grant.]
A comparison of normal mode and parabolic equation range-dependent propagation models as tools for acoustic communication in shallow water
Authors: Sidorovskaia Natalia A. , Field Robert L. , Sephus Cheryl L. , Ioup George E. , Ioup Juliette W.
Published: February 2014

Accurate prediction of the effects of a shallow-water propagation channel on acoustic signal distortion would aid in the optimal design of shallow-water communication systems. It has been shown that the correlation between modeled (based on a parabolic equation approach) and measured ocean responses decreases significantly with increasing propagation range because of the mismatch in relative amplitudes and time delays between the modeled and measured bottom-interacting arrivals. This raises the question whether the discrepancies are related to the particular modeling algorithm or to insufficient information about the propagation channel. A new normal mode range-dependent algorithm is used to simulate the acoustic response for the source–receiver configuration used in an experiment in the Atlantic Ocean at the Southern end of the Blake Plateau in which a 25–150 Hz linear, frequency-modulated source signal is received on a 15-element vertical array. The normalized correlation coefficients between source and receiver signals as a measure of signal distortion are computed as functions of receiver depth and propagation range and compared for both modeling algorithms. The influence of the inaccuracy of the bottom parameters, particularly the compressional to shear wave conversion in determining high distortion areas, is also discussed.
The effect of diffused arrivals on the performance of phase coherent underwater acoustic communications
Authors: Yang T. C. , Yoo Kwang , Al-Kurd Azmi
Published: February 2014

The delayed multipath arrivals cause intersymbol interference (ISI) in underwater acoustic communications, which can sometimes extend over tens to hundreds of symbols. The performance of the decision feedback equalizer (DFE) requires a reasonably accurate estimate of the number of tap coefficients as too few taps fail to remove the ISI and too many taps increases the mean square error (the processor noise). An initial estimate of the channel impulse response is often used to determine the number and positions of the taps. This decision is clear-cut for well-defined (usually high-amplitude) arrivals but ambiguous for diffused arrivals that are weak and undistinguished. It should be noted that diffused arrivals can be a significant component of the signal, resulting usually from sound scattering from fluctuating surface/bottom and random media. Its amplitude and complexity are compounded and enhanced by platform motion. This work illustrates the effect of diffused arrivals on the performance of the equalizer by introducing diffused arrivals in a simple model and proceeds to show the effect of wind speed on the bit error rate in realistic ocean channels. [Work supported by ONR.]
Experimental structure factor in stratified rockwool samples
Authors: Picard Lopez M. A. , Solana Quiros P. E. , Arizo Serrulla J. V.
Published: February 2014

In the study of porous materials, the structure factor allows us to relate the phenomenological approach and the microestructural. In these approaches the structure is supposedly more rigid than that of the contained air in its pores. In the literature it is demonstrated that remarkable efforts have been made to modeling the shape of the pore whose section in a first approach was considered cylindrical. But when the shape is studied on the microscope stratified commercial samples, one can observe that as the density diminishes it is practically impossible to assign a shape to the pore. For these situations, it is not easy to assign a value to the structure factor by means of geometric considerations in commercial samples. In this work an acoustic evaluation method has been developed in the phenomenological approach that can be applied to real samples. The existing formulation has been revised in several suppositions. The function of the obtained results can be considered as a parameter that helps toward the design and improvement of these materials and the information is complete with other parameters, such as flow resistivity.
Source dependency on anechoic chamber validation
Authors: Cunefare Kenneth A. , Biesel Van , Albanese Anne-Marie , Chang Lisa , Holdhusen Mark
Published: February 2014

The ANSI S12.35-1990 and similar ISO standard establish procedures for measuring sound power in anechoic and hemi-anechoic chambers. In addition, the standards outline procedures for use in the qualification of the test chambers that are used for such measurements. The basic qualification procedure requires a draw-away test from a sound source. Different sources are recommended to generate test signals over low-, medium-, and high-frequency ranges. These ranges overlap at crossover frequencies, where two different sources may be used to test the chamber. For example, the mid-range source is constructed by bolting together two 10-cm speakers face-to-face. The recommended high-frequency source is a baffled speaker radiating through a narrow cylindrical tube. The work presented here investigates how the qualification tests at mid and high frequencies are impacted by the use of sound sources fabricated to the recommended configurations, but using speakers from different manufacturers, and, for the high-frequency source, constructed with different diameters and lengths of tube.
Intensity as a quality assessment method for anechoic chambers
Authors: Babineau Francis J. , Tinianov Brandon D.
Published: February 2014

Data regarding the Johns Manville hemi-anechoic chamber qualification according to ISO 3745 Annex A and ANSI S12.35 Part 9 were presented at ASA/NoiseCon in December 2000. In addition to techniques based on these standard test methods, an alternate qualification technique was devised using a pressure–pressure intensity probe. The basis of the technique stems from the fact that the sound pressure and intensity should be nearly equal in a free-field environment. The difference between the sound pressure and sound intensity at a point is known as the pressure-intensity index (p-i index), a common ISO measure of source strength. P-i index measurements were made along several measurement paths and at discrete points at the predicted boundaries of the room. Results of this nonstandard test were compared to those of traditional qualification techniques. Although the method uses a possibly uncommon instrument, the results suggest a greater tolerance for various sound sources and allow for quick computation of chamber qualification.
Computational study of cat eardrum mechanics
Authors: Fay Jonathan P. , Puria Sunil , Steele Charles R.
Published: February 2014

The function of the eardrum and the middle ear is to resolve the acoustic impedance mismatch between the air of the outside world and the fluid of the inner ear. Without an impedance matching device, very little acoustic energy would be absorbed by the inner ear and hearing would be severely limited. It remains a mystery how the eardrum accomplishes this over the audible frequency range. A computer simulation of the cat eardrum was constructed. The vibrations of the eardrum were fully coupled to the acoustics of the ear canal and the dynamics of the middle ear bones. The eardrum’s fibrous microstructure was taken into account. This model appears to be valid up to 20 kHz. This represents a significant improvement over previous modeling efforts. With this model, the following questions surrounding the evolution and design of the eardrum have been addressed: (1) Why does the eardrum have its distinctive conical and toroidal shape? (2) What is the significance of its highly organized fibrous structure? (3) What is the advantage of its angular placement in the ear canal? and (4) What is the benefit of complex vibrational modes on the eardrum? [Work supported in part by research grant R29 DC03805 from NIDCD.]
The ultrasonic receiving system of a Pacific white-sided dolphin: Part 1. Experimental results
Authors: Takagi Shigeru , Takemura Akira , Koido Takashi , Yoshizumi Kazuhiro
Published: February 2014

To investigate the ultrasonic receiving system of a dolphin’s sonar, three experiments were carried out using dead dolphins’ heads. The first experiment was an anatomical investigation by magnetic resonance imaging followed by dissection. The second experiment was measurements of the receiving sensitivity distribution on a dolphin’s head surface to locate receiving areas with high sensitivity. The third experiment was muscle excision to determine which organs contribute to the ultrasonic sound wave propagation from the highly sensitive areas to the tympanic bone. These experimental results show that the receiving sensitivity is highest at the middle points of both the left and right mandibula, at which points the left and right digastrici terminate, respectively. The highly sensitive areas lie along lines just inside of the lower mandibula, where both digastrici are located. Furthermore, not only are the highly sensitive areas along the right mandiblum connected acoustically to the right tympanic bone but those on the left side are also cross-connected by muscles on the lower jaw and vice versa. Therefore, a dolphin’s ultrasonic receiving system may be modeled by a multi-input/two-output system in which each right or left ear, respectively, is a multi-input/single-output system.
The ultrasonic receiving system of a Pacific white-sided dolphin: Part 2. Multi-input/two-output system
Authors: Takagi Shigeru , Takemura Akira , Koido Takashi , Yoshizumi Kazuhiro
Published: February 2014

According to anatomical investigations, measurements of the receiving sensitivity distribution and investigation of ultrasonic sound wave propagation paths, each ear may be constructed as a multi-input/single-output system in the head. These two multi-input/single-output systems may be considered to act together as one multi-input/two-output system. Here, the input points are highly sensitive areas lying just inside of the lower mandibula. The two output points are the tympanic bones in each mandibula. All input points are connected acoustically to both tympanic bones by the the digastrici, mylohyoidei and sternohyoid muscles. Mathematical simulations were carried out using this multi-input/single-output system, applying impulse response functions previously measured from the right tympanic bone to each of the measuring points on the dolphin’s head surface. From the results of these simulations, it is considered that the highly sensitive areas located in lines along both mandibula act as line arrays, so the beamwidth of the receiving system can be narrow. Furthermore, the spectrum of the echo signal is changed by the target direction, and human beings can recognize this difference when frequency components of the echo signal are shifted down to the range of human hearing.