bluuebunny/crossref_metadata_2025 huggingface dataset:
Development and achievement of hearing-aid fitting strategy based on loudness model
Authors: Lai Ying-Hui , Jhou Yang-Ming , Young Sheunn-Tsong
Published: September 2008

The purpose of this study was to develop a hearing-aid fitting strategy (named NYMU strategy), which aimed to compensate the loudness perception of hearing-impaired individuals as that of normal-hearing individuals. The loudness growth curve of normal-hearing individuals was derived by the equal loudness contour. The loudness growth curve of hearing-impaired individuals was constructed by fitting the curve through three points, which were the uncomfortable loudness level, the most comfortable loudness level, and the pure-tone threshold of hearing-impaired individuals for each frequency band. The NYMU strategy primarily calculated the hearing-aid gain for each frequency band according to the differences between the loudness growth curve of normal-hearing individuals and that of hearing-impaired individuals; the gain was set to guarantee the audibility of the lowest boundary of speech intensity with normal vocal effort. Furthermore, the NYMU strategy adopted an innovative approach, which adjusted the point of the most comfortable loudness level to fine-tuning the loudness growth curve, and it resulted in the fine-tuning gain of each band. The preliminary results reveal that the NYMU strategy provides more gain above 4,000 Hz and under 500 Hz, compared with the NAL-NL1 strategy. In addition, three of five subjects get higher word recognition scores when they were fitted by the NYUM strategy rather than the NAL-NL1 strategy.
Off-frequency listening: A population study
Authors: Eckstein Veronica , Berg Bruce
Published: September 2008

A pitch-matching paradigm was used to investigate the pitch of three-tone complexes in a college-age population. For different conditions, the central tone was 1,000 Hz with the side tones spaced by either 15 Hz or 45 Hz. The tones were equal intensity or a signal was added in phase to the central tone with a level of 5, 0, or 5 dB relative to the side tones. Listeners adjusted the frequency of a pure tone to match the pitch of the complex. According to a model known as the envelope-weighted average of the instantaneous frequency (EWAIF) [L. Feth, Percept. Psychopys. 15, 375–378 (1974)], the pitch should be approximately equal to the center frequency for all conditions. Contrary to this prediction, increasing the bandwidth or the decreasing the signal level systematically lowered the pitch. However, the results are consistent with EWAIF calculations if the stimuli are first passed through the skirt of an auditory filter. Results for the population were consistent with pitch matches for expert listeners. [Work supported by the Center for Hearing Research, UCI.]
Predicting the ability to learn complex auditory tasks
Authors: Watson Charles S. , Kidd Gary R.
Published: September 2008

The goal was to determine whether the ability to learn complex auditory discrimination and identification tasks could be predicted from performance on a screening test battery. The screening battery included auditory tasks chosen on the basis of earlier factor-analytic studies of individual differences [Kidd et al. JASA 122, 418–435 (2007)], visual acuity tasks, and a test of working memory. These tests were administered to 1,000 college students with normal audiograms. Based on screening test performance, 25 exceptional performers and 25 average performers were selected for prolonged training on three new auditory tasks. The training tasks included multi-tone pattern discrimination, identification of a vocabulary of sequences of three-tone syllables, and binary categorization of multi-dimensional profile-like stimuli. A broad range of individual differences in post-training performance was observed on each of these tasks, but differences were not well predicted by screening task performance or estimated general intelligence. Analysis of the screening task performance supported the previously reported independence of individual differences in spectral-temporal acuity and the recognition of familiar sounds (both speech and nonspeech). Intellectual abilities and visual processing abilities were also independent of each other and of auditory abilities. [Work supported by Grant No. N000140310644 from the Office of Naval Research.]
SPATS: Speech perception assessment and training system
Authors: Miller James D. , Watson Charles S. , Kewley-Port Diane , Sillings Roy , Mills William B. , Burleson Deborah F.
Published: September 2008

A software system (SPATS) to test and train important bottom-up and combined bottom-up/top-down speech-perception skills is described. Bottom-up skills are the abilities to identify the constituents of syllables: Onsets, nuclei, and codas in quiet and noise as produced by eight talkers. Top-down skills are the abilities to use knowledge of linguistic context to identify the words in spoken sentences. The sentence module in SPATS emphasizes combined bottom-up/top-down abilities in recognizing sentences in noise. The word-initial onsets, stressed nuclei, and word-final codas are ranked in importance and grouped into subsets, based on their importance. Testing utilizes random presentation of all the items included in a subset. Training in Quiet (SNR=40 dB) or in Noise (SNR=5 dB), is adaptively focused on individual listeners learnable items of intermediate difficulty. Alternatively, SNR-adaption training uses Kaerenbachs algorithm to find the SNR required for a target percent correct. The unique sentence module not only assesses and trains a listener’s ability to infer words in sentences based on linguistic cues, but also trains the ability to combine bottom-up with top-down abilities to identify words in spoken sentences in noise. Scoring in the sentence module is objective and automatic. SPATS demonstrations are included.
Examination of bone-conducted sound pathways
Authors: Brault Lynn M. , Lansing Charissa R. , Chambers Ron D. , O’Brien William D.
Published: September 2008

Evidence of bone-conducted sound pathways through the skull have been explored with growth of loudness tests, auditory brainstem response, and otoacoustic emissions; however a comprehensive understanding of these pathways has not been achieved. Less is known about the usefulness of behavioral thresholds, sound pressure levels in the ear canal, and the auditory steady-state response (ASSR) as a function of ear canal occlusion in understanding propagation pathways of bone-conducted signals. Results acquired from multiple, repeated test sessions with a single-subject design, revealed lower behavioral thresholds and higher sound pressure level in the ear canal (Real Ear) in the occluded condition below 2,000 Hz; however, this trend was reversed above 2,000 Hz. The ASSR revealed lower thresholds in the occluded condition at all frequencies. Real Ear measured the bone-conducted signal while the behavioral thresholds show the product of the signal after following the traditional bone conduction pathway through the ear structures and the brain. The ASSR results indicate the responses recorded were suprathreshold, and are the product of bone conduction sound excitation at the level of the auditory nerve. Understanding bone-conducted sound pathways could provide guidance for designing enhanced hearing protection, new diagnostic protocols, and novel approaches for signal processing. [Work supported by AFOSR FA9550-06-0128.]
Infants’ and adults’ sensitivity to a tone masked by remote frequency tones or noise
Authors: Werner Lynne A. , Leibold Lori J.
Published: September 2008

It was previously reported that infants and adults exhibit masking of a tone by two tones of constant, remote frequencies. Neither informational nor energetic masking would be expected for adults under these conditions. In the present experiment, infants’ and adults’ detection of a tone masked by two constant frequency tones or two narrow bands of noise were compared. The target tone was 1,000-Hz, 300-ms duration with 16-ms rise/fall, at a level expected to produce rmd′=1 in the two-tone masker condition for each age group. The maskers were either tones of 581 and 2,920 Hz or 50-Hz-wide noise bands centered at 556 and 2,895 Hz. The duration of the maskers was 300 ms with 16-ms rise/fall. The maskers repeated at 600-ms intervals. The target tone was presented simultaneously with the maskers on signal trials. Sensitivity was measured using an observer-based procedure. Listeners received feedback whenever they correctly detected the target. Infants achieved a higher rmd′ in the two-noise-band condition than in the two-tone condition; there was no difference in rmd′ between these conditions for adults. Thus, introducing a timbre difference between the target and masker reduced the amount of masking for infants, but not for adults.
Psychoacoustic evaluation of broadband and conventional reversing alarms localization accuracy
Authors: Lakatos Stephen
Published: September 2008

Three psychoacoustic tests assessed whether a recently developed class of vehicle reversing alarms with broadband signal characteristics would be more accurately localized than conventional equivalents. Vehicle reversing conditions were simulated by mounting sample alarms on a linear slide whose velocity and displacement could be controlled by computer. Reversals of 4 m displacement were recorded at two velocities (1 m/s and 2 m/s) and at azimuthal orientations varying in 50 increments. The first two tests presented direct hit trajectories recorded with a binaural microphone (Crown SASS-P) and Kemar Mannequin, respectively; a third test presented near miss trajectories with the recording centerline perpendicular to the alarm trajectory and at linear displacements of 0.5 m and 1 m. Participants listened to trajectories over supra-aural headphones and selected the closest match from computer depictions of all possible trajectories. Results for all three tests revealed no significant advantage in localization acuity for the broadband alarm.
Effects of frequency shifts on the identification of vowels and words in sentences
Authors: Assmann Peter F. , Nearey Terrance M.
Published: September 2008

Studies of the effects of frequency shifts on vowel identification have shown a drop in accuracy when the spectrum envelope is shifted up or down, and when the fundamental frequency (F0) is raised or lowered. We have found an interaction between F0 and spectrum envelope shifts: Performance is better for vowels with matched shifts (both F0 and spectrum envelope shifted in the same direction) compared to mismatched shifts (F0 shifted up and spectrum envelope shifted down or vice versa). The aim of the present study was to determine the extent to which these effects persist in sentence context. The STRAIGHT vocoder was used to process a set of sentences from the HINT test using the same scale factors as in the vowel identification experiment. Word recognition scores generally followed the same pattern as vowel identification, with poorer performance for lowered F0 and raised spectrum envelope, and the lowest scores in conditions with high F0 and downward shifts in spectrum envelope, compared to the corresponding matched shifts. [Work supported by NSF and SSHRC.]
Effects of a response format modification on the diagnostic rhyme test
Authors: Goshorn Edward L.
Published: September 2008

Detection theory measures of listener performance, as well as distinctive feature scoring, have been shown to provide useful additional information to percent correct scores when testing word recognition. The diagnostic rhyme test (DRT) uses a two-item multiple choice (MC) response format to provide percent correct and distinctive feature information but does not measure d. This study examined the effects of modifications to the DRT response format that allow a measure of d. The format was modified to a single-option in which the listener designates yes or no (Y-N) that the single-option matches the stimulus. DRT words were mixed in white noise and presented in sound field at 3 and 6 dB signal-to-noise ratios at 50 dBHL to eight normal-hearing subjects. Results showed a significant difference between response formats for percent correct and a significant interaction between signal-to-noise ratios and feature errors across formats. At 3 dB, there were more temporal-based feature errors (sustension and compactness) for the Y-N response format, but no significant differences between formats for voicing, nasality, sibilation, and graveness. At 6 dB, there were significant differences only for sustension and graveness. The ds for signal-to-noise ratios were significantly different for the Y-N format.
Evaluation of the consonant-nucleus-consonant word test as spoken by a female talker
Authors: Fourakis Margaritis , Hawks John , Zingler Heather , Persak Lyndsey , Hawks John , Zingler Heather , Persak Lyndsey
Published: September 2008

The most common recording of CNC words used to test speech perception by persons with cochlear implants utilizes a single male talker. However, research has shown that CI recipients often have difficulty perceiving speech by persons with high F0 and formant frequencies (Loizou, P., Dorman, M., and Powell, V. 1998. J. Acoust. Soc. Am. 103: 1141–1149). Presented here are CNC word lists recorded using an adult female talker. Ten normal hearing adults identified these words with 99% accuracy. Eight CI subjects also listened to the original (male voice) and new (female voice) recordings. The two star subjects identified the female speakers words 15% less accurately than the male speakers words (Male voice: 82% correct; Female voice: 67% correct). For the other CI subjects (mean performance around 40%) there was no effect of voice on their performance. Results of the acoustic analysis of all the stimuli will also be presented. [Research supported by a subcontract from Washington University in St. Louis, NIDCD R01 DC000581.]
The response of a beam-like acoustic sensor to transient sound waves
Authors: Shepard W. Steve , Zhang Bill B.B.
Published: September 2008

An analytical approach is presented for studying the response of a beam-like structure being considered to act as an acoustic sensor under the excitation of a transient sound wave. First, the response of a Bernoulli-Euler beam is modeled for a multi-cycle transient wave excitation moving at a constant speed. For verification, the model is compared to other approaches by looking at extremes of the input. Then, to examine sensing requirements for the beams structure, the influence of various parameters on the ability to excite the sensor in the midfrequency region is investigated. The parameters considered include beam length, beam materials, amplitude of the transient pressure excitation, duration of the excitation, and the number of sinusoidal half-cycles present in the transient excitation. Potential effects of the superposition of incident and reflected waves caused by the boundary conditions on the sensors response are discussed. [Work supported by the National Science Foundation.]
S-matrices for waves in random structures, unitarization of diffuse field waveforms, and mesocopics
Authors: Weaver Richard
Published: September 2008

Diffuse field methods, such as reverberation room acoustics and statistical energy analysis, predict wave energy flow in complex structures. These methods are applied here to predict the root mean square s-matrix 1/2 for an undamped structure coupled to a single incoming and outgoing channel. Multiplying by a white noise random process produces a candidate s(t). The prediction is statistically exact if the internal scatterings are sufficiently phase incoherent, but it is not unitary and is thus inadmissible in the absence of such scatterings. Here it is shown that it can be made unitary in a minimally invasive manner by filtering with a minimum phase causal filter with modulus 1/|s(ω)|. This is illustrated with envelopes  predicated on a picture of the structure as a single reverberation room, as two coupled rooms, and as a quasi-one dimensional multiply scattering system. The resulting unitarized S matrices are found to exhibit a variety of familiar mesoscopic features and behaviors not present in the original diffuse . These include enhanced backscatter, quantum echo, power law tails, level repulsion, and Anderson localization. It is remarkable, and of potential practical interest, that quantitative mesoscopic features follow from simple principles. [Work supported by NSF CMS 05-28096.]
A simplified finite-element modeling approach for constrained layer damping
Authors: Xie Zhengchao , Shepard W. Steve
Published: September 2008

In formulating finite-element models for multilayer structures, designers usually use a detailed model that contains many degrees of freedom. In order to reduce the modeling complexity for structures containing constrained layer damping, some researchers have attempted to develop equivalent properties that are applied to regular single-layer finite elements. For example, the analytical formulation developed by Ross, Ungar, and Kerwin is sometimes used to find an equivalent bending stiffness. However, that stiffness is based on an infinite or simply-supported beam, which is often inappropriate. The research presented here uses the governing equations of a regular beam or plate to derive the frequency transfer function between two points on the structure. Curve fitting this transfer function to a reference value found using a more detailed model provides a complex bending stiffness at the first few peak frequencies. The goal is to develop an improved approach for extracting equivalent properties for the simplified modeling of multilayer structures. Numerical examples show that extracted properties can lead to good results when calculating the response of the structure. This approach may provide a means of reducing the modeling requirements for viscoelastically damped structures.
Dynamics of a hollow rubber-metal laminate structure
Authors: Smith Robert W. M.
Published: September 2008

Transverse frequencies of vibration of a hollow cylindrical rubber-metal laminate structure, subject to a mean pressure difference across the cylinder wall will be presented. Extending Timoshenko beam theory to model such a structure, we use two independent observations most often credited to J. A. Haringx: First, an analogy exists between an axially loaded Euler beam and a pipe without axial loading but subject to internal pressure, and second, for this type of rubber-metal laminate structure, the transverse component of the axial load acts perpendicular to the bending slope, rather than the deflection slope. When dynamics of the pressurized laminate are considered, theory predicts somewhat unexpected changes in the resonance frequency of transverse bending modes as a function of mean pressure difference across the structure. Experimental results illustrating static buckling behavior will be shown for both positive and negative pressure differentials, corresponding to the analogous axial loading cases of buckling in both compression and tension. In applications where oscillatory pressures are present across the structure, the unusual tuning curve has implications for prediction of parametric instabilities.
Radiation from a source embedded in a two-layer coating of a spherical shell
Authors: Ginsberg Jerry H.
Published: September 2008

Analysis of the acoustic field generated by a small source embedded in a viscoelastic layer encapsulating a spherical shell gives rise to several analytical difficulties not encountered in the classical problem of a point source in the surface of a rigid sphere. The present investigation addresses a double layer configuration in which the layer closest to the shell is acoustically soft, whereas the outer layer, in which the source is embedded, has an impedance that matches that of the surrounding fluid, water. The analysis is carried out by treating the layers as dissipative fluids, while classical Love theory describes the interior shell. The analysis employs spherical harmonic expansions for each medium, which is known as Mie series analysis in related studies of electromagnetic scattering from coated series. Like the electromagnetic problem, computational difficulties associated with the occurrence of spherical Bessel functions of complex argument arise, but the remedies used there require modification.
Tilted aluminum cylinder acoustic scattering properties and holographic and synthetic aperture sonar (SAS) images
Authors: Baik Kyungmin , Dudley Christopher , Marston Philip L.
Published: September 2008

A solid bluntly-truncated aluminum cylinder in water is a convenient target for exploring the relationship between free-field scattering properties of short tone-bursts and acoustic images. These experiments concern situations where the product ka of the acoustic wave number and the cylinder radius exceed about 12. There are several prominent features in the tilt-angle dependence of the time-resolved backscattering. Some of these features are also present in prior measurements for tilted stainless steel cylinders using relatively long tone bursts [K. Gipson and P. L. Marston, J. Acoust. Soc. Am. 106, 1673–1680 (1999); K. Gipson and P. L. Marston, J. Acoust. Soc. Am. 107, 112–117 (2000)]. Prominent features include responses caused by guided generalized Rayleigh waves associated with meridonal rays, helical rays, and face crossing rays. These features are easily observed for tilted aluminum cylinders in the scattering of short tone bursts for cylinders having length-to-diameter ratios of 3 and 5. Guided Rayleigh waves also influence the appearance of the imaged cylinders in holographic and synthetic aperture sonar (SAS) images constructed using far-field bistatic data. This influence on the images is explained using ray theory. [Supported by ONR.]
Scattering from penetrable prolate spheroids
Authors: Boisvert Jeffrey E. , Van Buren A. L.
Published: September 2008

The problem of acoustic scattering from a fluid-filled elastic layer (shell) is considered. The inner and outer surfaces of the elastic shell are confocal prolate spheroids with the outer surface having major-axis length L and minor-axis length D. A plane wave ensonifies the shell at an arbitrary angle of incidence. The 3-D equations of elasticity in prolate spheroidal coordinates are solved using a Helmholtz decomposition for the displacement vector. The displacement vector is expressed in terms of a scalar potential (dependent on the dilatational wave number) and a vector potential (dependent on the shear wave number). These wave potentials are solutions, respectively, of the scalar and vector wave equations cast in prolate spheroidal coordinates. An eigenfunction expansion of spheroidal wave functions represents the solution of the scalar wave equation. The solution of the vector wave equation is facilitated using expansions of prolate spheroidal vector wave functions. Nearfield and farfield scattering results are presented as a function of incident angle, spheroid shape L/D, and internal fluid (air-filled or water-filled). Comparisons with the rigid case (impenetrable spheroid) are also presented. [Work supported by the NAVSEA Newport ILIR Program.]
Benchmark solutions for low-frequency structural acoustic target scattering
Authors: Zampolli Mario , Tesei Alessandra , Jensen Finn B.
Published: September 2008

Low-frequency scattering problems, typically in the range up to 10 s of kHz for O(1 m) size targets, can be solved by a variety of numerical techniques, each of which has different tradeoffs. In September of 2006, a Target Strength Modeling Workshop was held at NURC with the purpose of establishing a set of benchmark solutions for problems of significance to the underwater structural acoustics modeling community. Problems for low-frequency scattering from spherical and cylindrical objects located in the water column or proud or buried in the seabed are presented. Solution techniques ranging from fully 3-D finite-element/boundary integral techniques, to Fourier decomposition axisymmetric finite-element techniques and semi-analytical methods have been applied to the problem set. The reported solutions show benchmark quality agreement in most cases, and hence could be used as reference solutions for testing low-frequency structural acoustic modeling tools in the future. The results and techniques presented reach beyond the realm of underwater acoustics, and should be of interest to researchers working in the broader field of structural acoustics.
Suppression of responses to pitch perturbations during Mandarin speech
Authors: Larson Charles R , Liu Hanjun , Xu Yi
Published: September 2008

Previous research on the task dependency of voice fundamental frequency (F0) responses to pitch-shifted feedback has shown them to be larger during speech than in a vowel task. The present study investigated how responses to pitch-shift stimuli vary with the intonation of a speech utterance and the timing of the pitch shift stimuli. Mandarin speakers produced a four-word phrase in which the F0 was held relatively steady and then either increased or was held steady on the final word. Pitch shift stimuli (100 cents, 200 ms) were presented 160, 240, or 340 ms after the onset of vocalization. F0 response magnitudes for the 340 ms onset condition were significantly smaller than in the 160 or 240 onset conditions when the F0 level on the final syllable was rising. When the F0 on the final syllable was held constant, there was no difference in response magnitudes between the three onset conditions. The decrease in response magnitude in the 340 ms, F0 rising condition, corresponding to approximately 660 ms before the final syllable, suggests vocal responses can be modulated both by the timing of stimulus presentation and the speech intonation pattern. Possible explanations for this finding will be discussed.
Role of voice F0 production accuracy and pitch perception ability in modifying compensatory voice F0 responses during sustained vowel productions
Authors: Bauer Jay , Gelfer Marylou , Bashford James
Published: September 2008

The purpose of the present study is to assess whether subject characteristics in terms of voice F0 production accuracy and pitch perception ability are related to the ability to compensate in real-time for altered auditory feedback. To assess the relationship of vocal production, pitch-perception, and voice F0 error-correction, a series of three tasks were completed by typical college-age participants: (1) A pitch-matching task where participants vocally matched the pitch of audible complex tones, (2) two-tone forced choice pitch-discrimination tasks where participants decided if complex tone pairs (200 ms duration for each tone) sounded the same or different, and (3) pitch-shifting tasks where participants produced sustained vowel phonations, while listening to experimentally altered auditory feedback (brief 200 ms pitch perturbations of +/−20, 60, or 100 cents were introduced into the feedback). Dependent measures of pitch-matching accuracy, pitch-discrimination sensitivity, and variations in the latency, stability, and magnitude of compensatory pitch-shift reflexes were assessed in all participants. Pilot data (N=3) indicate that production accuracy and discrimination sensitivity does not play a role in compensatory control of the voice. Poor pitch-matchers and/or discriminators actually respond to very small magnitude perturbations (20 cents) in a manner similar to accurate pitch-matchers and/or perceivers.
Modification of acoustic-vocal mappings for fundamental frequency control in singers and nonsingers
Authors: Jones Jeffery A. , Keough Dwayne
Published: September 2008

Singing requires accurate control of the fundamental frequency (F0) of the voice. Previous work has demonstrated that F0 control involves an interplay between closed- and open-loop control. A frequency-altered auditory feedback (FAF) paradigm was used to examine the acoustic-motor representation of the mapping between F0 feedback and the vocal production system in singers and nonsingers. Participants sang a note while hearing their F0 shifted down one semitone. Both singers and nonsingers compensated for the F0 perturbations by increasing their F0. However, the magnitude of compensations was initially smaller in singers than nonsingers. Moreover, after participants heard their feedback returned to normal, the aftereffects observed for singers were larger than those observed for nonsingers. This pattern of aftereffects generalized to the production of another note different than the one participants produced while hearing FAF. Combined, these observations suggest that singers rely more on an internal model for F0 production during singing than nonsingers.
Motor and sensory adaptation following auditory perturbation of /s/ production
Authors: Shiller Douglas M. , Sato Marc , Gracco Vincent L. , Baum Shari R.
Published: September 2008

Studies of motor adaptation have provided valuable insights into the role of auditory and orosensory feedback in speech production. While perturbations to the structure of the vocal tract (e.g., pseudopalate), which alter both orosensory and auditory feedback, have been used to investigate the production of a range of consonants and vowels, studies of short-term speech adaptation utilizing purely auditory feedback manipulations (e.g., spectral shifting) have been mostly limited to investigations of vowel production. These studies have revealed that motor plans underlying vowel production are flexible, and that subjects will readily alter their motor output in order to achieve auditory perceptual targets. In the present study, we have extended this work by spectrally shifting the acoustic signal for the sibilant /s/ during the production of short /s/-initial words. Evidence of motor adaptation was observed following a brief, intensive period of practice under altered auditory conditions, indicating a role for auditory feedback in sibilant production. In addition to changes in speech motor output, subjects showed changes to their auditory-sensory representations of /s/ and /∫/ following the manipulation, suggesting that adaptation was not limited to the motor domain. These results will be discussed in the context of current models of speech motor control.
A turbo approach to distributed detection and estimation
Authors: Roan Michael
Published: September 2008

A framework, based on the turbo principle, for distributed detection and estimation (DDE) using data from multiple sonar sensors is developed. Through the use of a Bayesian framework for DDE, an ideal observer is derived. This distributed observer makes optimal decisions, interpretations, or inferences based on measurements made at multiple sensors that observe the same scene. In this approach, a posteriori estimates at one sensor become prior probabilities for the next sensor. The posterior probability is proportional to the product of the probability of each possible state of the environment before observation (prior density) and the probability of the measurements given the state of the environment. It is the posteriors and the likelihoods representing knowledge about the environment that are shared among sensors. This iterative approach is mathematically similar to turbo coding in communications applications. This talk provides the background on the turbo DDE engine, and approaches for both practical implementation and performance quantification. Examples using Gaussian assumptions on the signal and noise distributions are given using simulations. The optimum approach is shown to be very good under ideal conditions, but not robust to sensor positioning errors. Sub-optimum approaches to improve robustness are introduced.
The impact of acoustic signal models on time-difference-of-arrival/frequency-difference-of-arrival estimation
Authors: Fowler Mark L. , Hu Xi
Published: September 2008

Much research has been done in the area of estimating time-difference-of-arrival (TDOA) and frequency-difference-of-arrival (FDOA) and their use in locating a radiating source. Early work in this area focused on locating acoustic sources using passive sonar processing. Only later was TDOA/FDOA-based location considered for the case of passively locating electromagnetic sources. As a result of this, it is tempting to use results derived for the acoustic case when answering questions about the electromagnetic case. This paper shows that such borrowing can lead to incorrect results. The key factor that drives the significant differences between these two cases is the difference between the signal model assumptions for the two cases: WSS Gaussian process is widely used as a valid model in the acoustic case, but is not usually appropriate in the electromagnetic case. Although the received signal equations may look identical (showing delay and Doppler shift), the resulting Fisher information, Cramer-Rao bound, and maximum likelihood estimator (MLE) are fundamentally different for the two signal scenarios. Thus, we show that there are distinct structures in processing and bounds that arise specifically due to using models appropriate for acoustic signals.
A local particle-like approach to wave propagation
Authors: Cohen Leon
Published: September 2008

Waves and particles are the main constituents of nature, but waves are our main method of communication, whether biological or technological. Historically, the study of waves has been separated into the stationary (standing wave) and nonstationary (pulse) case. Pulses are of fundamental consideration in radar, sonar, acoustics, fiber optics, among many other areas. A pulse has been called by many terms—transient, wave group, progressive wave, wave packet, nonrecurrent wave, traveling wave, nonperiodic wave—all to describe basically the same thing. The fact that so many names exist is an indication that there is no full understanding of their behavior. We present a simple particle-like view of pulse propagation, which makes pulses easier to understand and deal with intuitively. We have derived exact expressions for many physical quantities such as the spreading of a pulse, the conditions as to when a pulse contracts, the contraction time, and other important physical quantities. What emerges from these considerations is not only a clearer view of how pulses propagate, but a practical means to obtain their propagation in a dispersive medium.
Improving spatial resolution of a time-reversed focus using frequency-domain signal processing
Authors: Anderson Brian E. , Griffa Michele , Ulrich T. J. , Johnson Paul A. , Guyer Robert A.
Published: September 2008

In the typical application of time reversed acoustics, the spatial resolution of a time-reversed focus is set by the diffraction limit, i.e., a focal spot size cannot be made smaller than a half of a wavelength. A frequency-domain signal processing method is introduced in which the information in the received signal of the forward propagation stage is shifted to higher frequency. The resulting time domain signal is time reversed and rebroadcast to form a focal spot size that is diffraction limited at the higher frequency (shorter) wavelength. This method requires neither a priori information about the original source location nor nearfield signal detection. A theoretical discussion of the problem, along with numerical and experimental results, will be presented. The limitations and potential applications will also be presented. [This work is supported by Institutional Support (LDRD) at Los Alamos National Laboratory.]
Selective source reduction to identify masked sources using time-reversed acoustics
Authors: Anderson Brian E. , Ulrich T. J. , Griffa Michele , Johnson Paul A. , Scalerandi Marco , Gliozzi Antonio S.
Published: September 2008

This paper describes a time-reversed acoustics (TRA) method of spatially illuminating a source signal, which has been masked by another source signal. The selective source reduction (SSR-TRA) method employs a subtraction technique where one focus is selectively reduced to illuminate the masked focus. In this paper, numerical and experimental results are presented to demonstrate to what degree the SSR-TRA method is successful. The SSR-TRA method is demonstrated for two elastic wave pulses emitted simultaneously from two spatially separated sources of differing amplitudes. Results are presented from experiments conducted with two different solid samples: Aluminum and doped silica glass. Applying the SSR-TRA method, a stronger source, up to 13 times stronger than a weaker one, may be reduced to reveal information about the weaker source. Spatial and/or temporal characteristics of multiple close proximity sources can be resolved with the use of the SSR-TRA method. The results show that the SSR-TRA methods’ limitations are chiefly due to imperfect reconstruction of the source function in the time-reversed focal signal. [This work was supported by Institutional Support (LDRD) at Los Alamos National Laboratory.]
Cued beamforming of passive arrays in the context of active-passive data fusion
Authors: Yudichak T. W. , Le Minh
Published: September 2008

Active-passive data fusion seeks to exploit the complementary information measured simultaneously by active and passive sonar sensors: For instance, active sensors can provide good range estimates for a contact, while passive sensors can provide information on both the bearing of and the spectrum radiated by that contact. One aspect of this subject is the use of feedback, in which prior information is collected about the state of a contact in the data fusion framework, and then exploited to enhance the signal processing of data measured in that framework. Cued beamforming is one such method, in which a dense set of passive beams are steered toward regions of high contact probability, as determined by the result of the fusion of initial active and initial passive state estimates. The denser cued beams may then provide a refinement of the position estimate of the contact, as well as better isolation of its spectral signature. Comparisons of the effect of using cued beams are compared to the effect of using standard beams (those used without prior knowledge) on the state estimate of contacts in various scenarios using simulated data. [Work supported by ONR.]
Scattering models for Reverberation Modeling Workshop I problems
Authors: Thorsos Eric I.
Published: September 2008

For many of the problems posed for Reverberation Modeling Workshop I, the boundary roughness was presented in terms of roughness spectra. Models and/or tables were also supplied for the bistatic scattering cross section and for the coherent reflection loss. Integral equation simulations were used to verify the accuracy of the bistatic scattering cross sections, and rough boundary PE simulations were used to verify the accuracy of the coherent reflection losses. The results of this work will be briefly summarized. The issue also arises for reverberation simulations whether the coherent reflection loss is appropriate for the boundary loss, or whether some other loss model is more appropriate, or whether the loss due to roughness should be simply ignored. This issue will be discussed in light of rough boundary PE simulations. [Work supported by ONR.]
Reverberation Modeling Workshop problem definitions
Authors: LePage Kevin D.
Published: September 2008

In November of 2006 the first Reverberation Modeling Workshop was held in Austin, Texas. In this talk, the problems defined for the workshop are presented, along with some selected collective results to give an idea of the level of agreement achieved between the participants. In all, 20 problems were defined, ranging from monostatic 2-dimensional range independent problems up through fully three-dimensional bistatic problems. [Work supported by ONR and PEO C41 and Space (PMW 180).]
Parabolic equation techniques for reverberation modeling
Authors: Lingevitch Joseph F. , Fromm David M. , LePage Kevin D.
Published: September 2008

Recent developments of parabolic equations for modeling reverberation will be presented. This method is applicable to the 2-D reverberation problems with general sound speed profiles, sediment layering, and interface roughness. The environment is sub-divided into range-independent regions, and the forward and back-scattered fields are coupled at each interface with a single-scattering approximation. This method incorporates a cumulative scattering loss into the forward field, but neglects the non-local coupling terms between the forward and backward scattered fields. The back-scattered field is stored at each interface on a forward sweep through the environment and back-propagated on backward sweep. Reverberation time series are Fourier synthesized from the frequency domain solutions. This technique is fundamentally different than scattering kernel based approaches and provides an independent method for obtaining the forward and back-scattered solutions for reverberation problems. As currently implemented, this model does not include multiple scattering effects, but the algorithm may be generalized to include multiple scattering with iterative sweeps through the propagation domain. The method applied to 2-D reverberation problems from the recent Reverberation Modeling Workshop (November 2006) and results are compared with ray and normal-mode based models. [Work supported by ONR.]
An investigation into the validity of the Born approximation in reverberation calculations
Authors: Knobles David P. , Stotts Steven A. , Isakson Marcia J. , Wilson Preston S.
Published: September 2008

Ocean reverberation models for active sonar applications assume the validity of the Born or single scatter approximation. In many areas of physics, a Born series does not converge due to a combination of the scattering strength and the spatial scale over which the scattering occurs. In these cases, the validity of the Born approximation is at best questionable. It is natural to attempt to examine the validity of the Born approximation and the convergence of the Born series in general for well-defined scattering problems in ocean acoustics. The problems examined in this study involve scattering from specific realizations of bottom roughness for 2-D waveguides. This study examines several idealized problems from the 1st Reverberation Workshop in November 2006 for the purpose of comparing a Born approximation solution to an exact solution provided by a 2-way coupled-mode integral equation approach. The coupled integral equations are solved exactly to within the limitations of finite differences and numerical quadrature. The exact solutions are then compared to those that solve the fundamental integral equation using the Born approximation and to solutions obtained with a finite-element model. [Work supported by ARL:UT IR&D.]
A comparison of ray, normal-mode, and energy flux results for reverberation in a Pekeris waveguide
Authors: Ellis Dale D. , Ainslie M. A. , Harrison C. H.
Published: September 2008

A number of problems were developed for, and presented at, a 2006 Reverberation Modeling Workshop sponsored by the US Office of Naval Research. The simplest of these known to the participants as Problem 11, was a Pekeris waveguide (isospeed water over a flat bottom half-space) with Lambert bottom scattering. The water depth was 100 m and frequencies of 250, 1,000, and 3,500 Hz were specified. A number of source-receiver combinations were specified, but the reverberation predictions are quite insensitive to sensor depth except at 250 Hz. With some benefit from hindsight and the results from other models, we compare our results from ray, normal-mode, and energy-flux approaches. All three approaches agree at intermediate times, say 3 to 50 s. At short times, the steep-angle paths and fathometer returns cause the mode and energy-flux models to underpredict the reverberation. At longer times, the ray models run out of steam: i.e., there are too many contributing ray paths for them to handle so they underpredict the reverberation. By combining the model predictions together with analytical results from an energy-flux model, we propose a composite benchmark solution. [Work supported in part by ONR.]
Issues associated with solutions to a rough-bottom Pekeris waveguide
Authors: Stotts Steven A. , Knobles David P. , Koch Robert A.
Published: September 2008

The solution to reverberation time series in a 2-D shallow-water ocean waveguide with bottom roughness (c.f. Problem 1, Reverberation Workshop 2006) has been obtained using both plane wave Born approximation (PWBA) and two-way coupled-mode (CM) approaches. The source and receiver are range monostatic. The CM solutions obtained can be used to determine the accuracy of the Born approximation. Specified time intervals of the received field from the CM solution are correlated and compared to both the patch size and bottom roughness correlation length in the PWBA solution, thus connecting the two approaches. The minimum number of bottom interactions required for accuracy in the PWBA approach is determined. Propagation effects associated with time bin averaging in the PWBA approach are also discussed. The importance of including continuum components in the CM approach, via leaky modes evaluated in the complex horizontal wave-number plane, is also described. Convergence of the CM solution in an expansion of an orthogonal vector space is determined. [Work supported by ARL:UT IR&D.]
A Matlab and normal mode based reverberation model and some data/model comparisons
Authors: Preston John R. , Ellis Dale D.
Published: September 2008

A Matlab and normal mode based reverberation model has been developed that uses Ellis’ algorithm [J. Acoust. Soc. Am. 97, 2804–2814] for reverberation predictions and the ORCA normal mode model [Westwood et al., J. Acoust. Soc. Am. 100, 3631–3645] to compute the eigenvalues and mode functions. The model is currently range independent, but handles bistatic geometries, and towed array beam patterns. Group velocity corrections similar to LePage’s [J. Acoust. Soc. Am. 106, 3240–3254] have recently been added to this model. The matrix formulation in Matlab allows the bistatic calculations to be performed more efficiently. An overview of the model is presented that includes sample model-model comparisons for some problems from the 2006 ONR Reverberation Workshop. The model predictions are also compared with data obtained with the NURC (formerly the NATO Undersea Research Centre, SACLANTCEN) towed arrays during several reverberation experiments on the Malta Plateau. Select data from these experiments are used to compare with model predictions using best estimates of bottom properties in the area. [Work supported by US Office of Naval Research, Code 321OA, Grant No. N00014-05-1-0156.]
The importance of retaining inter-path coherence in reverberation prediction
Authors: LePage Kevin D.
Published: September 2008

The R-SNAP and BiStaR normal mode-based reverberation models are used to solve a selection of problems from the 1st Reverberation Modeling Workshop held in Austin, TX in November of 2006. R-SNAP and BiStaR are codes that include the effects of inter-mode decorrelation to predict the coherent aspects of reverberation, as well as allowing for the option to compute the ‘‘incoherent’’ reverberation. Comparisons between the incoherent R-SNAP and BiStaR predictions and model predictions obtained by other methods show good agreement. However, it is argued here that as defined, the problems require a coherent solution, and differences between the incoherent and coherent solutions can be significant. Comparisons with coherent 2-way PE results for problem I computed by Joe Lingevitch (NRL) show that coherent structure in reverberation exists, is predictable, and should therefore be an integral part of any reverberation calculation. [Work supported by ONR.]