bluuebunny/crossref_metadata_2025 huggingface dataset:
The effect of internal waves on mid-frequency underwater acoustic communications
Authors: Yang T. C.
Published: September 2013

Internal waves are abundant in coastal waters and are known to have a significant impact on acoustic signal propagation. The effect includes significant signal fading to loss of signal coherence depending on the acoustic signal frequencies. This paper presents the analysis results using binary phase-shifted keying signals for underwater acoustic communications during the ASCOT01 experiment. We report the measurements of temporal coherence time as a function of time and relate it to the performance of the decision feedback equalizer. The interpacket coherence was measured using the LFM signals in each packet which were separated by ∼2 min. We see that the intensity and arrival time of the multipath arrivals vary significantly from packet to packets. The signal coherence within a packet (intrapacket) is measured using consecutive m-sequences. We find that at mid (2–5 kHz) frequencies, the acoustic environment, while randomly changing with time on the scale of minutes, presents an instantaneous deterministic environment for each packet; the coherence time is much longer than the packet length. The environmental impact can be mitigated by adequate sampling of the channel impulse response function. [Work supported by the ONR.]
Characterization of seismic signals and background seismic noise
Authors: Uhrhammer Robert
Published: September 2013

A variety of signal processing methodologies have been utilized in the analysis and characterization of the seismic signals and the background noise recorded by modern seismic instrumentation at the UC Berkeley Seismological Laboratory. The capabilities of seismic instrumentation have vastly improved over the past couple of decades since the advent of microcomputer and related technologies. Modern seismic instrumentation consists primarily of force fedback broadband seismometers (to record weak ground motions) and strong motion accelerometers (to record large local shocks) coupled to high resolution digital (24-bit+integer) data loggers. We discuss the characteristics and capabilities of the various seismic sensors and the digital recording systems. The earth’s background noise and the instrumental self-noise characteristics are discussed. Various weak signals (both coherent and transient) detection and analysis methodologies are also discussed along with some interesting cases of detecting and characterizing unusual natural and artificial seismic sources.
Techniques for estimating sediment attenuation from chirp sonar data
Authors: Turgut Altan
Published: September 2013

Recent broadband (50–500 Hz) matched-field inversions of geoacoustic parameters indicated that sediment attenuation is difficult to estimate especially at short source-receiver distances. On the other hand, geoacoustic inversions become less practical at long source-receiver distances due to the range-dependency of both ocean bottom and water column. Several inversion techniques are investigated to estimate in-situ sediment attenuation from normal incidence reflection data, collected by chirp sonars. Both frequency-shift and attenuation roll-off measurement techniques showed nonlinear frequency dependency of attenuation for sandy sediments at the 2-12-kHz frequency band. Estimated attenuation values are extrapolated to lower frequencies by using the Biot theory. The extrapolated results are consistent with those of the matched-field inversions. Several advantages and limitations of the frequency-shift and attenuation roll-off measurement techniques are also discussed. [Work supported by ONR.]
A theoretical method to extract target information between two vertical arrays
Authors: Sanders William , Werby Michael
Published: September 2013

For a known source and suitable environmental information one may determine the presence (or absence) of a target ensonified by a guided wave initiated by the source by using vertical arrays that sandwich a region of interest. We show first that one can represent the ensonified object as a source term in the solution of an inhomogeneous Sturm–Louiville problem where the homogeneous solution corresponds to the solution in the absence of a target. This allows one to choose a bounded region about the object and evaluate the equivalent of Huygens’ integral (over the boundary that includes the vertical arrays), which enables one to trace the signal back to the object. This determines both the range and depth and even the target strength. When this is extended to a pulse signal then the information leads to higher fidelity than the frequency results due to phase averaging of fluctuations. Thus we demonstrate this to be a good strategy for an experimental study that would lead to a robust detection method that may be implemented on a rapid basis.
The dependence of Lamb wave stimulation parameters on the impedance difference between upper and lower liquid
Authors: Declercq Nico F. , Van den Abeele Filip , Degrieck Joris , Leroy Oswald
Published: September 2013

It is known that inhomogeneous plane waves are better capable of stimulating Lamb waves in a solid plate than homogeneous plane waves. For each Lamb mode there exists an angle of incidence and an inhomogeneity parameter that maximizes its stimulation. If for example a solid plate separates two different liquids (a lower liquid and an upper liquid), then the angle of incidence, for stimulating the same Lamb mode, is unchanged, whereas the inhomogeneity differs compared with the situation where both liquids are equal to each other. If for a given upper liquid and for a given Lamb mode this inhomogeneity difference is plotted as a function of the impedance of the lower liquid, then a linear dependence is noticed. It is noticed that all physical liquids follow this linear tendency. Furthermore, this linearity depends on the generated mode and also on the properties of the solid plate. [Work supported by The Flemish Institute for the Encouragement of the Scientific and Technological Research in Industry (I.W.T.).]
Development of a submersible, baffled impedance tube for characterization of bubbly liquids
Authors: Wilbur Jed C. , Holmes Jason , Canney Michael , Carrey William M. , Roy Ronald A.
Published: September 2013

The ability to measure the acoustic properties of bubbly liquids in situ has long been a goal of the underwater acoustics community. An apparatus for characterizing these properties in the 1–16-kHz frequency range capable of submerged deployment in a controlled environment (large tank or test pond) has been developed. A water-filled, 2.5-cm-thick walled, baffled, stainless steel impedance tube was used to measure the complex reflection coefficient at the opening. The classical result for the radiation impedance of a baffled, open tube is verified, as is that for the radiated sound field along the tube axis. The test tank is then filled with air bubbles (order 500 micron mean radius and 0.1% vf) and the measurement repeated at the water/bubbly liquid interface. The acoustic properties of the bubbly liquid were obtained as a function of frequency by an inversion process using classical theory. Because of the high attenuation in bubbly liquids, the finite-sized baffle (61-cm outer diameter) is treated as being infinite in extent. The measured sound speeds and attenuations are compared with established theory. [Work supported by U.S. Navy ONR.]
An experimental comparative study of 20 Italian opera houses: Measurement techniques
Authors: Farina Angelo , Armelloni Enrico , Martignon Paolo
Published: September 2013

For ‘‘acoustical photography’’ we mean a set of measured impulse responses, which enable us to ‘‘listen’’ at the measured room by means of advanced auralization methods. Once these data sets have been measured, they can be employed in two different ways: objective analysis and listening test. In fact, it is possible to compute dozens of acoustical objective parameters, describing the temporal texture, the spatial effect and the frequency-domain coloring of each opera house. On the other hand, by means of the auralization technique, it becomes easy to conduct listening experiments with human subjects. This paper focuses principally on the development and specification of the measurement technique, which is the topic assigned to the research unit of Parma, to which the authors belong. It describes the hardware equipment, the software, the electro-acoustic transducers (microphones and loudspeakers), the measurement positions, the system for automatic displacement of the microphones and the conditions of the room during the measurements. Experimental results are reported about a couple of opera houses which were employed for testing the measurement procedure and showing the benefits of the new method against the previously employed ones.
Acoustical measurements in a 1:10 scale model of a multi-purpose hall
Authors: Jeon Jin Yong , Ryu Jong Kwan
Published: September 2013

Measurements were made in a 1:10 scale model of a 1500-seat multi-purpose hall to investigate the acoustic properties of its opera mode. The length from the stage front to the rear wall is 33 m and its maximum width is 28 m with a volume of 14300 m3. Reflecting surfaces were included to scatter early energy from the stage area onto the audience areas. Several settings of the sources were used on the platform and in the pit in order to determine how a lack of brilliance caused by diffraction occurred in the stalls audience. The advantages of the curved ceiling surface in front of the proscenium above the orchestra pit were exploited. In addition, the effect of some diffusing treatments on surfaces adjacent to the pit and on the side walls was also investigated.
Measurements of the self-to-other ratio in an opera chorus in performance
Authors: Ternstrom Sten , Davis Pamela
Published: September 2013

The objective of this study was to obtain measurements of the self-to-other ratio (SOR) in the opera chorus on stage. Four members of the opera chorus of Opera Australia volunteered to carry wireless binaural microphones during a dress rehearsal of Verdi’s The Masked Ball, in the Sydney Opera House. Conditions were those of actual performance in costume, with soloists and orchestra. Hence, the recordings are very realistic, if acoustically not so stringently controlled. The SOR was estimated from the recordings using M/S matrixing, complemented with L–R correlation by adaptive filtering. The results indicate how much louder the singers heard themselves relative to the rest of the ensemble, including the orchestra. The SOR varied with score and on-stage formation, but was 4–8 dB higher than in ordinary choirs (not opera), as might be expected from the podium acoustics. While each opera chorus artist can hear his or her own voice very well, they often hear less of the rest of the chorus and very little of the orchestra. This was borne out in informal listening; the orchestra often became inaudible once the choir entered its louder passages. [Work supported by the Australian Research Council under the SPIRT program.]
Acoustical measurement of the Salt Lake Mormon Tabernacle
Authors: Rollins Sarah , Leishman Timothy W.
Published: September 2013

An acoustical survey of the Salt Lake Mormon Tabernacle has been performed to assess the behavior of the hall in its current state. The tabernacle is a well-known historical building with a large elongated dome ceiling. This paper discusses the measurements used to characterize the hall. Several parameters derived from omnidirectional, directional, and binaural impulse response measurements are presented. Color maps of the parameters over audience seating areas reveal their spatial variations. These maps and the statistical properties of the parameters aid in clarifying the acoustical characteristics and anomalies of the hall.
New Amsterdam Theater
Authors: Scarbrough Paul , Greenberg David
Published: September 2013

a)Paul Scarbrough was principal-in-charge and David Greenberg was the project manager for Jaffe Holden Scarbrough Acoustics.
Digital signal processing at Bell Labs—Foundations for speech and acoustics research
Authors: Rabiner Lawrence R.
Published: September 2013

Digital signal processing (DSP) is a fundamental tool for much of the research that has been carried out of Bell Labs in the areas of speech and acoustics research. The fundamental bases for DSP include the sampling theorem of Nyquist, the method for digitization of analog signals by Shannon et al., methods of spectral analysis by Tukey, the cepstrum by Bogert et al., and the FFT by Tukey (and Cooley of IBM). Essentially all of these early foundations of DSP came out of the Bell Labs Research Lab in the 1930s, 1940s, 1950s, and 1960s. This fundamental research was motivated by fundamental applications (mainly in the areas of speech, sonar, and acoustics) that led to novel design methods for digital filters (Kaiser, Golden, Rabiner, Schafer), spectrum analysis methods (Rabiner, Schafer, Allen, Crochiere), fast convolution methods based on the FFT (Helms, Bergland), and advanced digital systems used to implement telephony channel banks (Jackson, McDonald, Freeny, Tewksbury). This talk summarizes the key contributions to DSP made at Bell Labs, and illustrates how DSP was utilized in the areas of speech and acoustics research. It also shows the vast, worldwide impact of this DSP research on modern consumer electronics.
Multichannel signal processing at Bell Labs Acoustics Research—Sampled by a postdoc
Authors: Kellermann Walter
Published: September 2013

In the mid 1980’s, the first large microphone arrays for audio capture were designed and realized by Jim Flanagan and Gary Elko. After the author joined Bell Labs in 1989, the first real-time digital beamformer for teleconferencing applications was implemented and formed a starting point for the development of several novel beamforming techniques. In parallel, multichannel loudspeaker systems were already investigated and research on acoustic echo cancellation, small-aperture directional microphones, and sensor technology complemented the research scenario aiming at seamless hands-free acoustic communication. Arrays of many sensors and loudspeakers for sampling the spatial domain combined with advanced signal processing sparked new concepts that are still fueling ongoing research around the world—including the author’s research group. Here, robust adaptive beamforming has found its way from large-scale arrays into many applications using smaller apertures. Blind source separation algorithms allow for effective spatial filtering without a priori information on source positions. Full-duplex communication using multiple channels for both reproduction and recording is enabled by multichannel acoustic echo cancellation combined with beamforming. Recently, wave domain adaptive filtering, a new concept for handling many sensors and many loudspeakers, has been verified for arrays that may well remind some observers of former Bell Labs projects.
Information entropy analysis of leopard seal vocalization bouts
Authors: Buck John R. , Rogers Tracey L. , Cato Douglas H.
Published: September 2013

Leopard seals (Hydrurga leptonyx) are solitary pinnipeds who are vocally active during their brief breeding season. The seals produce vocal bouts consisting of a sequence of distinct sounds, with an average length of roughly ten sounds. The sequential structure of the bouts is thought to be individually distinctive. Bouts recorded from five leopard seals during 1992–1994 were analyzed using information theory. The first-order Markov model entropy estimates were substantially smaller than the independent, identically distributed model entropy estimates for all five seals, indicative of constraints on the sequential structure of each seal’s bouts. Each bout in the data set was classified using maximum-likelihood estimates from the first-order Markov model for each seal. This technique correctly classified 85% of the bouts, comparable to results in Rogers and Cato [Behaviour (2002)]. The relative entropies between the Markov models were found to be infinite in 18/20 possible cross-comparisons, indicating there is no probability of misclassifying the bouts in these 18 comparisons in the limit of long data sequences. One seal has sufficient data to compare a nonparametric entropy estimate with the Markov entropy estimate, finding only a small difference. This suggests that the first-order Markov model captures almost all the sequential structure in this seal’s bouts.
Dual-modality phased array systems for hyperthermia cancer therapy
Authors: McGough Robert , Samulski Thaddeus
Published: September 2013

External ultrasound (US) phased arrays, as applied to hyperthermia cancer therapy, are generally local heating devices, whereas external radio-frequency (RF) electromagnetic (EM) phased array applicators are primarily designed for regional heating. Clinical applications of external ultrasound heating devices are presently limited to smaller tumors, and the inclusion of an external RF/EM applicator increases the size of the heated volume. When these two modalities are combined in a hybrid RF/US phased array structure, uniform heating is facilitated in larger tumors. This is demonstrated in computer simulations of thermal therapy in the breast, where adding an external RF/EM component to an ultrasound phased array in a hybrid structure produces temperatures of 43°C in nearly half of a 9-cm-diam tumor volume. If applied alone, neither modality achieves 43°C in such a large volume without overheating intervening tissues. This result is demonstrated in two hybrid RF/US applicator prototypes designed for hyperthermia treatments of locally advanced breast cancer. In each example, hybrid RF/US phased arrays achieve higher temperatures in larger tumor volumes by selectively delivering power to different regions in the tumor with each modality.
Apparatus for studying wave motion and sound at the University of Nebraska–Lincoln’s ‘‘Historical Scientific Instrument Gallery.’’
Authors: Wang Lily M. , Rudd M. Eugene
Published: September 2013

The University of Nebraska–Lincoln’s ‘‘Historical Scientific Instrument Gallery,’’ compiled by the second author in 1998, contains approximately 700 inventoried items and may be visited on-line at http://physics.unl.edu/outreach/histinstr/. Amidst the collection are several acoustical instruments that were used in the early 1900s. These include equipment that demonstrate wave motion (traveling wave machine, mercury ripple dish, vibration microscope), wave interference (interference machine), resonance conditions (Helmholtz resonators, vibrating rods, singing flames, sonometer), and sound generation (Galton’s whistles, high-frequency tuning forks, large tuning forks, organ pipes, siren saw). A review of the equipment and the history of their use at the University of Nebraska are discussed. Much of the equipment was superbly manufactured by the Max Kohl/Chemnitz Company in Germany and Rudolph Koenig in France. Pages from the Max Kohl/Chemnitz equipment catalogs of 1910 and 1925 helped to characterize several of the pieces and are shown in this presentation.
A model for precalculus students to determine the resonance frequency of a trumpet mouthpiece
Authors: Chapman Robert C.
Published: September 2013

The trumpet mouthpiece as a Helmholtz resonator is used to show precalculus students a mathematical model for determining the approximate resonance frequency of the mouthpiece. The mathematics is limited to algebra and trigonometry. Using a system of mouthpieces that have interchangeable cups and backbores, students are introduced to the acoustics of this resonator. By gathering data on 51 different configurations of mouthpieces, the author modifies the existing Helmholtz resonator equation to account for both cup volumes and backbore configurations. Students then use this model for frequency predictions. Included are how to measure the different physical attributes of a trumpet mouthpiece at minimal cost. This includes methods for measuring cup volume, backbore volume, backbore length, throat area, etc. A portion of this phase is de-signed for students to become acquainted with some of the vocabulary of acoustics and the physics of sound.
Comparison of Russian and American educational programs in acoustics
Authors: Zagrai Nikolai , Zakharevich Vladislav
Published: September 2013

Recent progress in finding parallels between Russian and American professional education programs has shown that particular attention should be paid to the differences and similarities in engineering education. The efforts of the Taganrog State University of Radio-Engineering (TSURE) are directed towards accreditation of its engineering programs in ABET. The Russian Bachelor program in acoustics was compared to the ABET-accredited American program in electrical engineering. A direct comparison has shown that Russian program has: (a) more credit hours in the course work under the liberal arts section (factor of 2); (b) traditionally more credit hours in mathematics and fundamental science (factor of 1.5); (c) same number of credit hours for the general engineering classes; (d) less number of credit hours for the specialized engineering classes. However, by examining the actual class contents of the specialized engineering classes in the American program, we observed that some classes contain material which, in Russian program, is usually attributed to the general engineering. Recent revision of the acoustic education program at TSURE aimed to increase its flexibility, support students’ mobility, and facilitate ABET accreditation.
Retrieving the sources in historical sound recordings
Authors: Brock-Nannestad George
Published: September 2013

Broadly speaking, historical recordings (ca. 1880–1950) are considered low quality, and when they are reissued it is regarded as an improvement when noises are removed, or according to modern psychoacoustic criteria, pushed into frequency bands where they are masked. However, a recording is a report of an acoustic event, and some of the elements or features of the original sounds must have been available to contemporary listeners, or they would not have accepted the recordings in the first place. Similarly, we may want to retrieve these elements, and for that purpose it is important to identify them. The paper points out some of the fundamental elements of the sounds that have been recorded and discusses the degree to which they may be retrieved in modern replay, either directly from original recordings or via signal-processed transfers. Reference is made to work by D. C. Miller based on his early recognition that the recording process transforms the elements it is desired to retrieve. In a similar manner that Miller compensated his recordings for measurement purposes, we may today compensate early recordings for replay purposes. The presentation will be accompanied by very short comparative excerpts of historical sound recordings.
Source identification and manipulation in stereo music recordings using frequency-domain signal processing
Authors: Avendano Carlos
Published: September 2013

A short-time frequency domain framework for source identification, separation, and manipulation in stereo music recordings is presented. Using a simplified model of the stereo mix, a similarity measure between the short-time fourier transforms (STFTs) of the input signals is computed to identify time-frequency regions occupied by each source based on the panning coefficients assigned to it during the mix. Individual sources are identified and manipulated by clustering time-frequency components with a given panning coefficient and frequency range. After modification, an inverse STFT is used to synthesize a time-domain processed signal. Applications of the technique to source suppression, enhancement and repanning will be described, and audio demonstrations will be presented to illustrate the results.
Music processing above and below the fundamental frequency
Authors: Athineos Marios , Ellis Daniel P. W.
Published: September 2013

Because of the centrality of pitch to music, most music signal processing has been focused on pitch-related timescales, e.g., short-time analysis over 30–50-ms windows. But there is, of course, important information at other scales as well. By looking at ways to model and extract very fine time-scale information one can extract attributes that lead to the perception of ‘‘texture’’ in unpitched instruments such as maracas. At the other extreme, interesting effects can be revealed by investigating the effect of ‘‘modulation-domain’’ processing, such as filtering subband energy envelopes in the 0.1–10-Hz range. A single representation, frequency-domain linear prediction, gives convenient access to both these scales: its use in music analysis/synthesis will be illustrated by examples.
On the possible role of MOC efferents in speech reception in noise
Authors: Ghitza Oded
Published: September 2013

The hypothesis explored in this study is that the MOC efferent system plays an important role in speech reception in the presence of sustained background noise. This talk describes efforts to assess this hypothesis using a test of initial consonant reception (the Diagnostic Rhyme Test) performed by subjects with normal hearing. Activation of selected parts of the efferent system was attempted by presenting speech and noise in various configurations (gated/continuous, monaural/binaural). Initial results of these experiments show a gated/continuous difference analogous to the ‘‘masking overshoot’’ in tone detection. These results are interpreted to support the hypothesis of a significant efferent contribution to initial phone discrimination in noise. [Work supported by AFOSR.]
New developments in transit noise and vibration criteria
Authors: Hanson Carl E.
Published: September 2013

Federal Transit Administration (FTA) noise and vibration impact criteria were developed in the early 1990’s. Noise criteria are ambient-based, developed from the Schultz curve and fundamental research performed by the U.S. Environmental Protection Agency in the 1970’s. Vibration criteria are single-value rms vibration velocity levels. After 10 years of experience applying the criteria in assessments of new transit projects throughout the United States, FTA is updating its methods. Approach to assessment of new projects in existing high-noise environments will be clarified. Method for assessing noise impacts due to horn blowing at grade crossings will be provided. Vibration criteria will be expanded to include spectral information. This paper summarizes the background of the current criteria, discusses examples where existing methods are lacking, and describes the planned remedies to improve criteria and methods.
Syllable-final fricatives: Dutch and English listeners’ processing of voicing
Authors: Broersma Mirjam
Published: September 2013

English and Dutch have both voiced and voiceless fricatives, and, in English, both occur in either syllable-initial or -final position. In Dutch, however, only voiceless fricatives can occur in syllable-final position. A categorization experiment investigated the processing of the voicing distinction in English fricatives by Dutch and English listeners, and in particular whether Dutch listeners can distinguish voicing contrasts in syllable-final position, and whether preceding vowel length informs their voicing judgments. Dutch and English participants heard tokens from an 11-step /v-f/ continuum, at the end of a single nonword. As the nonword context was kept constant within each block, vowel length could not serve as a cue. Half of the participants heard nonword contexts which were originally pronounced with a final /v/, the other half with final /f/. Therefore, in both cases a mismatch occurred for a subset of the items between vowel length and the other information in the signal. This mismatch hampered the performance of the English but not the Dutch listeners, so that the Dutch listeners outperformed the English in mismatching cases. This suggests that the Dutch did not rely on vowel length as a cue to voicing as strongly as the English.
Robust adaptive decision feedback equalization using residual prediction errors
Authors: Preisig James C.
Published: September 2013

Channel estimated based adaptive decision feedback equalization (DFE) relies on estimates of the channel impulse response and interfering noise field to calculate optimal equalizer filter coefficients. While the DFE offers superior performance to either passive time-reversal or minimum mean squared error (mmse) linear equalization when good quality channel estimates are available, the DFE is more sensitive than the other two techniques to channel estimation errors. A technique incorporating the residual prediction error of the channel estimation algorithm is presented. This algorithm, referred to as the residual prediction error DFE (RPE-DFE), maintains the superior performance that is characteristic of the DFE operating with good quality channel estimates while improving the robustness of the DFE to channel estimation errors. Experimental data is presented that allows for a direct comparison of the RPE-DFE performance with the performance of the traditional DFE, the passive time reversal and mmse linear equalizers, and robust DFEs proposed earlier [M. Stojanovic, J. Proakis, and J. Catipovic, IEEE T-Comm. 43, 877–886 (1995)]. [Work supported by ONR Ocean Acoustics.]
Doppler compensated underwater acoustic communication system
Authors: Raj Anand , George Binu , Supiya M. H. , Kurian James , Pillai P. R. Saseendran
Published: September 2013

Spread spectrum methods are used in communication systems to provide a low probability of intercept in hostile environments and multiple access capability in systems shared by many users as well as to provide high processing gain in channels where the transmitted signal is distorted by multipath effects. Such systems serve to be an effective tool for underwater telemetry environments, where multipath propagation effect and Doppler spreading is seen to be more predominant. This paper describes the implementation of a Doppler compensated underwater telemetry system based on CDMA technique. The system consists of multiple CDMA transmitters and a phase locked loop based carrier recoverable CDMA receiver. The effects of the Doppler shift can be compensated by the carrier recovery subsystem in the demodulator, based on PLL technique, which extracts the carrier frequency/phase and simultaneously demodulates the signal. The decision device in the receiver consists of a PN sequence generator as well as a bank of correlators, which are used to determine the data transmitted. The system simulation has been implemented in MATLAB. The advantage of this system is that multiple transmitting stations can transmit simultaneously to a central receiver, thereby increasing the system throughput.
Testing of ambient noise forecasting algorithms with LADC noise data
Authors: Eller Anthony I. , Ioup George E. , Ioup Juliette W. , Larue James P.
Published: September 2013

Long-term, omnidirectional underwater acoustic noise was measured by three buoys in the northern Gulf of Mexico over a 36-day period during the summer of 2001 as part of the Littoral Acoustic Demonstration Center project, phase I. This extensive data set is used to develop and evaluate algorithms for near-term temporal prediction of noise, based on immediate in situ data. Noise time series data in one-third-octave bands from 10 to 5000 Hz are processed to extract characterizing parameters such as mean level, variance, and temporal coherence across all frequency bands. The fluctuation spectrum is analyzed to determine how the noise variability is distributed with temporal period, and eigenvectors based on the fluctuation covariance matrix are used to identify underlying basic properties of noise variation. These tools are then used to predict future values of noise based on measurements at preceding times. [Research supported by ONR.]
An iso-deviant strategy for efficient ambient noise predictions with EAGLE
Authors: Rike Erik R. , DelBalzo Donald R.
Published: September 2013

Transmission loss (TL) computations in littoral areas require a dense spatial and azimuthal grid to achieve acceptable accuracy and detail, which is very slow. This problem of accuracy versus speed led to a new concept, OGRES (Objective Grid/Radials using Environmentally sensitive Selection), which produces sparse, irregular acoustic grids, with controlled accuracy. Recent work to further increase accuracy and efficiency with better metrics and interpolation led to EAGLE (Efficient Acoustic Gridder for Littoral Environments). On each iteration, EAGLE produces an acoustic field with approximately constant spatial uncertainty (hence, iso-deviance), yielding TL predictions with ever-increasing resolution and accuracy. This work adapts EAGLE to ambient noise computations. The Dynamic Ambient Noise Model (DANM) allows accurate, detailed estimation of the mean and variance of ambient noise in both space and time, but its TL computations are too slow for many applications. In the present work, a series of EAGLE acoustic field predictions was used by DANM (and compared to the dense full-grid solution) to determine the relationship between transmission loss uncertainty and noise-field uncertainty for a complex littoral area. An example is presented where approximately an order of magnitude efficiency improvement (over regular grids) is demonstrated. [Work sponsored by ONR under the LADC project.]
Ambient noise analysis of deep ocean measurements
Authors: Gaul Roy D. , Knobles David P. , Wittenborn A. F.
Published: September 2013

In October 1975 a measurement exercise designated CHURCH OPAL was done in the northeast Pacific Ocean to assess undersea acoustic noise and propagation phenomena. In 2003 the 10 days of deep ocean multiple hydrophone recordings during CHURCH OPAL were recovered and digitized. This paper presents results previously reported but unavailable for general distribution. The earlier work is augmented with more complete and detailed analyses using modern analytical techniques. Particular attention is given to statistical characterization of ambient noise within and beneath the deep sound channel in relation to distant shipping and local wind speed. [Work supported by ONR.]
Initial study of ambient noise modeling for 2001 ASIAEX SCS Experiment
Authors: Wei Ruey-Chang , Chen Chi-Fang , Chan Hsiang-Chih
Published: September 2013

Although ambient noise has been an object of study for a long time, there is little modeling done for real ocean environment to reflect oceanographic changes. The numerical simulation in this study is based on PE code, with environmental and bathymetry data from the 2001 ASIAEX SCS experiment. Simple sources were distributed near sea surface in the model, and source levels in significant range were estimated by the wave height data of experiment. It is well known that strong internal waves often occur in the South China Sea, which can cause significant change in water temperature distribution, thus the sound propagation. Moreover, the roughness of sea surface and bottom are also important factors in noise energy distribution. In this initial study, the fluctuation of noise energy would be evaluated by the varying spatial ocean environment. The computational noise levels would be compared with measurements on vertical line array in ASIAEX SCS experiment for adjusting the parameters and source levels.
Acoustical analysis and multiple source auralizations of charismatic worship spaces
Authors: Lee Richard W.
Published: September 2013

Because of the spontaneity and high level of call and response, many charismatic churches have verbal and musical communication problems that stem from highly reverberant sound fields, poor speech intelligibility, and muddy music. This research looks at the subjective dimensions of room acoustics perception that affect a charismatic worship space, which is summarized using the acronym RISCS (reverberation, intimacy, strength, coloration, and spaciousness). The method of research is to obtain acoustical measurements for three worship spaces in order to analyze the objective parameters associated with the RISCS subjective dimensions. For the same spaces, binaural room impulse response (BRIR) measurements are done for different receiver positions in order to create an auralization for each position. The subjective descriptors of RISCS are analyzed through the use of listening tests of the three auralized spaces. The results from the measurements and listening tests are analyzed to determine if listeners’ perceptions correlate with the objective parameter results, the appropriateness of the subjective parameters for the use of the space, and which parameters seem to take precedent. A comparison of the multi-source auralization to a conventional single-source auralization was done with the mixed down version of the synchronized multi-track anechoic signals.
Sensitivity of room acoustic parameters to changes in scattering coefficients
Authors: Rathsam Jonathan , Wang Lily M.
Published: September 2013

This project uses the room acoustics computer modeling program, ODEON, to investigate the sensitivity of room acoustic parameters to changes in scattering coefficients. Particularly, the study is interested in determining if the results from certain room models are more sensitive to scattering coefficients than from other models, due to their geometry or absorption characteristics. If so, how can one quantify a model’s susceptibility to being sensitive to scattering? Various models of three real spaces in Omaha, Nebraska are tested. The predicted reverberation, clarity, and spaciousness parameters are compared at various receiver locations, while the scattering coefficient of all surfaces is varied from 0 to 0.1, 0.3, 0.5, and 0.8. The resulting data are analyzed by frequency according to the (1) average absorption of the room; (2) magnitude variation of absorption within the room; (3) spatial distribution of absorption within the room; and (4) level of model detail. Initial results indicate that parameters studied may show more sensitivity to scattering coefficients in models that have a wider range of absorption values, more disparate distribution of absorption, and lower detail level. Various schemes that include these aspects are proposed for computing a model’s sensitivity to changes in scattering.
Biosonar signal processing of bats during flight observed by a telemetry microphone on the head
Authors: Riquimaroux Hiroshi , Watanabe Yoshiaki , Lin Liang-Kong
Published: September 2013

The purpose of the present study was to examine the daily consistency of constant frequency component (CF) of CF-FM bats and to examine biosonar sound characteristics of leaf-nosed bats (Hipposiderous terasensis) during flight by a telemetry microphone system (telemike) mounted on the bat’s head. Experiments were done within a steel-walled chamber. When at rest, the fundamental frequency of the bat was about 35 kHz. The second harmonic (70 kHz) was the most intense. The second harmonic of CF component (CF2) at rest and during flight varied significantly over several months. A systematical change in CF2 frequency during flight was observed through the telemike. Doppler-shift compensation in CF2 frequency, amplitude compensation, and pulse emission rate just before landing were confirmed. We will discuss time-sharing dual signal processing found during flight. [Research supported by Special Coordination Funds, and a grant to RCAST at Doshisha University from the Ministry of Education, Culture, Sports, Science and Technology of Japan.]
The ‘‘gunshot’’ sound produced by male North Atlantic right whales and its potential function in reproductive advertisement
Authors: Parks Susan E. , Hamilton Philip , Kraus Scott D. , Tyack Peter L.
Published: September 2013

North Atlantic right whales (Eubalaena glacialis) commonly use sound to mediate social interactions between individuals. Surface active groups (SAGs) are the most commonly observed social interaction on the summer feeding grounds. These groups are typically composed of an adult female with two or more males engaged in social behavior at the surface. Several distinct types of sounds have been recorded from these groups. One sound commonly recorded from these groups is a brief broadband sound, referred to as a gunshot sound because it sounds like a rifle being fired. This sound has been recorded in the Bay of Fundy, Canada from both lone whales (N=9) and social SAGs (N=49). Those lone whales producing gunshot sounds whose sex could be determined (N=9) were all mature males. In surface active groups, the rate of production of gunshot sounds was weakly correlated with the total number of males present in the group. Given the behavioral contexts of gunshot sound production by male whales, gunshots probably function in a reproductive context as an agonistic signal directed toward other males, an advertisement signal to attract females, or a combination of the two functions.
The beginning of Holliday’s underwater bioacoustics phase
Authors: Smith Paul E.
Published: September 2013

When Van Holliday entered graduate school, research on and management of coastal schooling fishes were in a crisis of disarray. Lasker’s sardine physiological work had yielded the unlikely result that the adult sardine had, at one time in the 1930s, consumed most of the production of the California Current. In the analysis of possible errors we had to know: (1) the swimming speed; (2) the size distribution; and (3) the depth distribution of fish in the schools. There were also opportunities. The new David Starr Jordan, a 50-m research stern trawler, had a Norwegian polished array of WWII tube-type acoustics with steerable powerful sonars at 11 and 30 kHz. This author had equipped the vessel with borrowed trawls. UCSD/SIO/MPL Professors Vic Anderson and Fred Spiess recommended that we add a research program to study the frequency content of acoustic signals. They also nominated a graduate student, Holliday, to design and lead this fisheries research. Van worked at a furious pace to assemble the sources and processors to carry out this work and then doubled and tripled that pace in the 20 months of sea work. Implementation of Holliday’s surveys was trapped in the technological dead zone of NOAA.
Critical scales for understanding the structure, dynamics, and impacts of zooplankton patches
Authors: Donaghay Percy
Published: September 2013

The development and application of high resolution acoustic systems by Holliday and Greenlaw has played a key role in demonstrating that zooplankton could can form highly concentrated layers ranging in thickness from a few decimeters to a few meters. These thin layers are remarkable in that they persist longer and have higher spatial coherence than multi-meter thick patches that can be sampled by nets and conventional acoustics. These results are inconsistent with the classical theory that thicker patches are more persistent and thus more important. Simultaneous measurements of temporal and spatial changes in the finescale physical, optical, chemical structure have provided important insights into the mechanisms controlling not only the dynamics of thin zooplankton layers but also their impacts on marine ecosystems and the transmission of sound in the ocean. [Work supported by grants from the ONR Biology and Chemistry program.]
Computational evidence for a discrete-scatterer aberration model in medical ultrasound
Authors: Lacefield James C.
Published: September 2013

Many techniques for correcting ultrasound focus distortion model the aberrating properties of tissue with a single time-shift screen, but simulations and phantom studies suggest single-screen models are ineffective for transmit focus compensation. Extension of the models to include multiple parallel screens is a logical increment in complexity, but the number of screens must be manageable and readily determined to yield practical aberration correction methods. To assess the feasibility of multi-screen strategies, simulations were performed to search for a general form for the aberration profile of breast tissue. Two-dimensional propagation of 3-MHz planar wavefronts through digitized breast specimens was computed using a k-space method [Tabei et al., J. Acoust. Soc. Am. 111, 53–63 (2002)] and waveforms were sampled at 1-mm intervals along the propagation direction. Arrival time, amplitude, and coherence fluctuations were correlated with scattering from distinct structures. This observation was most apparent when the first derivatives of those parameters with respect to the propagation direction were compared with the connective tissue architecture in the specimens. The assumption underlying time-shift screen models that aberration arises from smooth fluctuations in the acoustic properties of tissue merits reexamination. [Research supported by an NSERC Discovery Grant.]
Dual transmission model of the fetal heart tone
Authors: Baker Donald A. , Zuckerwar Allan J.
Published: September 2013

Detection of the fetal heart tone by auscultation is sometimes easy, other times very difficult. In the model proposed here, the level of difficulty depends upon the position of the fetus within the maternal abdomen. If the fetus lies in the classical left/right occiput anterior position (head down, back against the maternal abdominal wall), detection by a sensor or stethoscope on the maternal abdominal surface is easy. In this mode, named here the ‘‘direct contact’’ mode, the heartbeat pushes the fetus against the detecting sensor. The motion generates pressure by impact and does not involve acoustic propagation at all. If the fetus lies in a persistent occiput posterior position (spine-to-spine, fetus facing forward), detection is difficult. In this, the ‘‘fluid propagation’’ mode, sound generated by the fetal heart and propagating across the amniotic fluid produces extremely weak signals at the maternal surface, typically 30 dB lower than those of the direct contact mode. This reduction in tone level can be compensated by judicious selection of detection frequency band and by exploiting the difference between the background noise levels of the two modes. Experimental clinical results, demonstrating the tones associated with the two respective modes, will be presented.
Ultrasound produced homogenization and liquefaction of brain tissue in vivo at pressures beyond the cavitation threshold
Authors: Vykhodtseva Natalia , McDannold Nathan , Hynynen Kullervo
Published: September 2013

Histological effects of short-pulsed focused ultrasound at pressure amplitudes above the cavitation threshold were studied in the brain. Sixty locations in the brains of 13 rabbits were sonicated at a depth of 4–10 mm (1.5 MHz, burst lengths, 1–100 ms, PRF 1–100 Hz, acoustic power, up to 950 W). MRI was used to aim the beam, monitor the temperature, and evaluate the effects. It was found that high-intensity short-pulsed ultrasound could effectively destroy targeted tissues. The temperature elevation measured during the sonication was below the threshold for tissue damage. The tissue effects were different from that which has been described for the thermal or cavitation mechanisms. The lesions appeared as homogeneous regions with no distinguishable cell debris. The central core of the lesion region consisted of a pudding-like mass, arranged in a helically coiled, layered fashion. In some cases, the core seemed to be completely liquefied. These findings appeared to be associated with the nonlinear ultrasound absorption and possibly cavitation-related effects. The spiral shape observed in some cases may indicate the existence of an axial radiation torque in the focal zone. The lesions’ appearance may point to new interactions between ultrasound and tissue, which could be exploited for new therapies.