bluuebunny/crossref_metadata_2025 huggingface dataset:
Talker and speaking rate variation affect lexical neighborhoods
Authors: Sawusch James R. , Zimack Liza K. , Measer Kathleen M. , Newman Rochelle S. , Luce Paul A.
Published: October 2013

Words that are similar to a syllable can influence listeners perception of phonemes in the syllable. This influence of lexical neighborhood is part of how the words of a language influence speech perception. In previous studies (ASA, June 2000 and Dec 2000), this effect was shown to vary with talker voice, indicating that the similarity of words to nonsense syllables (lexical neighborhood) varies with talker. In a new series of studies, token variation in the nonsense syllables was introduced. Listeners identified the syllable initial stop in bowth-powth and powse-powse series. In these series, bowth and powse are similar to more real words. One group of listeners heard these series as spoken by four different talkers. The tokens of the different talkers were randomly intermixed. A second group heard tokens from one talker at three different speaking rates: fast, normal, and slow. Again, these were randomly intermixed. To the extent that token variation alters the influence of lexical neighborhood, factors other than simple acoustic–phonetic similarity must play a role in determining lexical neighborhood. The implications of these data for models of word recognition will be discussed. [Work supported by NIDCD Grant No. R01DC00219 to SUNY at Buffalo.]
Wavelet denoising of sidescan sonar images
Authors: Ioup Juliette W. , Gendron Marlin L. , Bourgeois Brian S. , Ioup George E.
Published: October 2013

One important application of wavelet transforms is for noise removal or denoising. The effectiveness of this technique is influenced by the choice of wavelet used, the decomposition level, and the threshold (both amplitude and type). Thirty different wavelets, several allowable decomposition levels, and a range of appropriate thresholds are tested. Preliminary results will be presented of wavelet denoising applied to 2-D acoustic backscatter imagery from a sidescan sonar in an attempt to improve the detection of bottom features. Comparisons with Fourier based filtering are also discussed. [Research supported in part by an NRL/ASEE Summer Faculty Fellowship and ONR.]
Simulations of rough boundary effects on shallow water propagation
Authors: Thorsos Eric I. , Williams Kevin L. , Henyey Frank S. , Reynolds Stephen A. , Elam W. T.
Published: October 2013

Results of PE simulations will be described for propagation in a shallow water waveguide with a rough sea surface and a flat seafloor. Pressure fields are simulated in two space dimensions and have been obtained using a wide-angle PE code developed by Rosenberg [A. D. Rosenberg, J. Acoust. Soc. Am. 105, 144–153 (1999)]. At a frequency near 3 kHz and with typical sea conditions, the behavior of the forward scattered field is found to depend dramatically on the incident grazing angle. A narrow incident beam at a 10° grazing angle is completely broken up, with energy scattered widely in angle. At a 1° angle, little energy is scattered out of the beam. Examples will also be shown for the more typical case of a relatively broad beam with a wide range of grazing angles at the surface. At low frequency, such as 100 Hz, it is useful to represent shallow water fields in terms of normal modes. Results of PE simulations using single mode initial fields and sinusoidal surfaces will be used to illustrate the effects of surface scattering on mode propagation and on mode couplings. [Work supported by ONR.]
Estimated reverberation room absorption coefficients from an empirically derived transformation of predicted coefficients of large flat absorbers
Authors: Godfrey Richard D.
Published: October 2013

ASTM C 423 specifies the industry accepted standard method to measure sound absorption coefficients by the reverberation room method. These coefficients are sometimes referred to as Sabin absorption coefficients. Because of the finite size of the sample used in these measurements, diffraction effects usually cause the apparent area of a specimen to be greater than the geometric area, thereby increasing the coefficients measured by this method over a sample of infinite lateral dimension which can be analyzed theoretically. In this paper, an empirically derived equation is present that transforms predicted absorption for large (infinite) flat absorbers to that which is measured by the reverberation room method. This empirically derived equation is based on an extensive database of measured absorption coefficients for A-mounted fiberglass materials with a wide range of flow resistivities and thickness. Estimated absorption is compared with measured data and infinite sample size predictions based on Mechel design charts [J. Acoust. Soc. Am. 83, 1002 (1988)]. Comparisons are also made between the estimate made with this empirical method, and an analytical method, which accounts for diffraction developed by Northwood [J. Acoust. Soc. Am. 31, 595 (1959); 35, 1173 (1963)].
Modeling the noise exposure of marine animals during seismic surveys
Authors: Duncan Alec J. , McCauley Robert D.
Published: October 2013

This paper describes a technique that has been developed to estimate the noise exposure of marine animals during seismic surveys. A seismic source model was developed to account for the directionality of many-element sources and was used to compute the source output as a function of frequency and azimuth. Sea bed, water column, and bathymetry properties were established for inputting to an appropriate sound propagation model, in this case the Range Dependent Acoustic Model (RAM). RAM was run for representative propagation paths at a number of frequencies spaced at octave intervals. The results were approximated by simple functions to allow rapid calculation of the received sound level for a large number of shot location/receiver location combinations. For a large number of receiver locations the number of shots were determined for which sound levels exceeded species dependent thresholds. These results were grided on an appropriate grid covering the entire survey area and presented in several different ways to highlight both spatial and temporal variations.
Pitches, amplitudes, and centroids: The PACT model of sound segmentation
Authors: Spiegelberg Scott
Published: October 2013

Logical sound segmentation is an essential tool in musical timber and articulation perception research. Unfortunately, the three current models of sound segmentation—the Constant Time Interval model, the 50% rms amplitude model, and the Amplitude/Centroid Trajectory model—have major theoretical flaws, preventing any real progress in these fields. The PACT (Pitch/Amplitude/Centroid Trajectory) model presents a new way to divide a musical sound into its attack, transition, steady state, decay, and release segments. This new method uses discrete wavelet transforms, spectral centroid envelopes and rms amplitude envelopes to define each of these segments. The most radical change is in the definition of the attack by its lack of pitch. The PACT model is supported by accepted theories of sound propagation and sound perception, with very clear results. The utility of the PACT model will be demonstrated with acoustical analyses of a variety of trumpet and violin articulations.
Guidelines for locating permanent noise monitors along a shore line
Authors: Luz George A. , Estridge Kimberly
Published: October 2013

The performance of a noise monitoring system located along the shoreline of the upper Chesapeake Bay was analyzed by looking at measurements of a 3 lb. explosive charge detonated each work day morning at about 8:00 am. Monitors located (1) on a cliff (2) at the water’s edge were more likely to register the explosions than monitors located as little as 500 m inland from the water’s edge. These results were consistent with M.J. White’s simulations of low frequency sound propagation across a land–water boundary [J. Acoust. Soc. Am. 108, 2650 (2000)]. Comparison of measured levels with predictions from the Noise Assessment and Prediction System (NAPS) demonstrated a degradation of the NAPS predictions at the land–water boundary.
Nonlinear seismo-acoustic land mine detection: Field test
Authors: Donskoy Dimitri , Ekimov Alexander , Sedunov Nikolay , Tsionskiy Mikhail
Published: October 2013

The paper presents results of the field test of the nonlinear seismo-acoustic technique for detection and discrimination of land mines [D. Donskoy, SPIE 3392 (1998); 3710 (1999)]. The tests were conducted in summer 2001 at the U.S. Army outdoor testing facilities. Plastic antitank mines (M19, VS1.6, VS2.2) and plastic antipersonnel mines (M14, VS50, TS50) were confidently detected at their maximum burial depths in both gravel and dirt lanes. Mine M14 is one of the smallest mine and practically undetectable by the competitive techniques, including NQR, GPR, etc. The test proved that the developed nonlinear seismo-acoustic detection approach is very sensitive to AT and AP mines, while completely insensitive to false targets, such as rocks, chunks of metal or wood, thus promising to deliver high probability of detection with low false alarm rate. The results of the tests are in good agreement with the developed physical model of the seismo-acoustic detection. [Work supported by the U.S. Army Research Office under Grant Nos. DAAG55-98-1-0402 and DAAD19-00-1-0039.]
Angle-of-arrival estimates for passive sensor arrays operating in atmospheric turbulence: Theory and experiment
Authors: Collier Sandra L. , Wilson D. Keith
Published: October 2013

Angle-of-arrival (AOA) estimates are considered for a passive sensor array operating in atmospheric turbulence. The Cramer–Rao lower bounds (CRLBs) of the azimuth and elevation are calculated for a spherical wave assuming multiple unknown parameters, a line-of-sight propagation path, and a single monochromatic source. The received signal is modeled as a complex Gaussian random variable with a deterministic mean. Therefore, this theoretical treatment is valid for strong or weak scattering in the presence of strong diffraction. The experimental variances of the AOA estimates calculated using a spatial Fourier transform beamformer are compared to the theoretical CRLBs. The data were collected at a planar array under several distinct atmospheric conditions [D. K. Wilson et al., Acoust. Res. Lett. Online 106, L24–L29 (1999)]. The CRLBs are calculated for the meteorological conditions observed during the experiment. Models based on von Kármán’s spectrum are used to model the atmospheric turbulence. The observed experimental standard deviations are typically found to be two to three times larger than the theoretical calculations.
The equivalent impedance of a comblike grating
Authors: Zhu Wenhao , Stinson Michael R. , Daigle Gilles A.
Published: October 2013

The scattering of acoustic plane waves by a comblike grating surface has been solved analytically. Both rigid and impedance gratings are considered. For the impedance grating, acoustic leakage through the partition walls is introduced by a leakage specific impedance. At low frequencies the grating can be represented by an equivalent impedance plane. Numerical results for the equivalent impedance are given for different grating geometries. It is shown that the equivalent impedance plane is locally reacting when the following conditions are met: λ/a≫1 and l/a≫1, where λ is the incident wavelength, a is the grating period, and l is the depth of the slots. The leakage impedance contributes to the nonlocal reaction of the equivalent impedance. When the incident wavelength is comparable to the grating period, a surface mode can be strongly excited, resulting in net surface wave propagation.
Modeling of acoustic pulse propagation over topographic ground
Authors: Hole Lars R. , Lie Knut , Dyrdal Idar
Published: October 2013

A hybrid coupled wave-number integration model and a time domain finite difference (TDFD) model have been used to predict propagation of low-frequency (below 500 Hz) acoustic pulses above ground with a hill present. The former model is also applied to study propagation along an upward slope. Ground conditions are described by poroelastic and viscoelastic theory. Air–ground interactions as well as the effect of important parameters such as permeability and thickness of poroelastic layers are discussed. Results are also compared with data from full-scale experiments.
Acoustical crack detection of the structure: Wigner–Ville distribution versus short time Fourier transform
Authors: Gelman Leonid M. , Giurgiutiu Victor , Cuc Adrian
Published: October 2013

The innovative acoustical forced oscillation method with nonstationary excitation for fatigue crack diagnostics of the structure is considered. Because of the nonstationarity of the acoustical structure response, the modern time-frequency technique is used for diagnostics. The numerical simulation is carried out to investigate the diagnostic capabilities of the time-frequency signal processing technique based on Wigner–Ville distribution. Results from the Wigner–Ville procedure are compared to results from the short time Fourier transform. [Financial support of the U.S. National Research Council Twinning Program with Ukraine (Senior Program Officer Kelly Robbins) is thankfully acknowledged.]
Effects of false vocal fold width on translaryngeal flow resistance
Authors: Agarwal Meena , Scherer Ronald C.
Published: October 2013

Our purpose in this study was to determine the effects of the false vocal fold width (FVFW) on laryngeal flow resistance. False vocal folds were placed within a Plexiglas model of the larynx. The dimensions and placement of the false vocal folds were based on an earlier anatomical study of laminagrams during phonation. The ratio of the FVFW and the minimum glottal diameter (Dg) used in this study ranged from 0.60 to 21.5. Results suggest that for large FVFW/Dg values, flow resistance is not affected. For ratios near 1 and lower, the resistance is greater than for the true folds alone. For intermediate ratio values, the flow resistance is less than for the true folds alone. These intermediate values should be prevalent during maximal glottal opening. The results obtained so far are dependent upon glottal diameter, with a greater reduction in flow resistance for larger values of glottal diameter. The results suggest that the false vocal fold width may help control laryngeal airflow differently throughout the phonatory cycle, and should be considered in models of phonation.
Empirical measurements of vocal fold viscoelasticity at relatively high frequencies
Authors: Chan Roger W.
Published: October 2013

Viscoelastic shear properties of human vocal fold tissues have been quantified in details in a series of previous experiments [see, e.g., Chan and Titze, J. Acoust. Soc. Am. 106, 2008–2021 (1999)]. However, due to mechanical limitations of the rheometer used for empirical viscoelastic measurements, data have only been obtained at a relatively low frequency (up to 15 Hz). These low-frequency data are an order of magnitude below the physiological frequency range of phonation (on the order of 100–1000 Hz), requiring data extrapolation based on various theoretical approaches. The only truly valid approach, however, is direct empirical measurement of viscoelastic data at high enough frequency. Using a controlled strain rotational rheometer free of the inertial problems present in controlled stress rheometers, this paper reports preliminary measurements of dynamic shear data made at relatively high frequencies (up to 80 Hz), close to the frequency range of phonation in male speech.
Analysis of factors affecting vocal fold impact stress using a mechanical model
Authors: Gunter Heather E. , Howe Robert D. , Hillman Robert E. , Stevens Kenneth N.
Published: October 2013

High mechanical stress levels in the vocal folds likely contribute to pathological changes to the tissue such as vocal nodules. Vocal fold collision during phonation was modeled as a dynamic contact problem in order to calculate stress levels during impact. The model incorporated a 3-D, linear elastic, finite element representation of a single vocal fold, a rigid midline surface, and a simplified aerodynamic waveform. The effects of fold geometry, material properties, and sub-glottal pressure on peak impact stress were calculated. Impact stress time course and the relationship between sub-glottal pressure and peak impact stress agree with published experimental measurements [Jiang and Titze, J. Voice 8, 132–144 (1994)]. Identification of high-stress configurations will promote better understanding of the etiology, persistence, and treatment of certain pathological voice changes. [Work supported by the Whitaker Foundation.]
The effects of fluctuation based processing on waterfall displays
Authors: Dalrymple Dale B.
Published: October 2013

Fluctuation based processing is a nonlinear method of processing time series data. Passive sonar waterfall displays are time-frequency plots where sequential data in each bin is a time series. Fluctuation based processing enhances the detection of low-fluctuation amplitude signals when processed against a background of high fluctuation noise or interferers. This paper examines the impact of fluctuation based processing on the signal processing load required for the generation of waterfall displays and examines alternatives to traditional waterfall normalization methods that apply to low-fluctuation amplitude signals.
Autocorrelation estimation using a traveling-wave filterbank
Authors: Wilson Timothy A.
Published: October 2013

A traveling-wave filterbank was implemented from a causal linear-phase unimodal-magnitude prototype by a time-scale operation similar to the design of a wavelet filterbank. The channels of the resulting filterbank are delayed relative to each other in a monotonic fashion, creating the traveling-wave effect. The scale varied in an exponential manner with channel number (place). Outputs at two places were multiplied by each other. Were the filterbank simply a nonuniform delay line and the input a stationary random process, each such product would be an unbiased estimator of the input autocorrelation at a lag corresponding the difference between the delays (relative to the input) of each channel. Considering the indices of two channels as coordinates in a Cartesian system, loci of constant relative delay were identified. Products of outputs along those loci were averaged to estimate the autocorrelation at the lag corresponding to that relative delay. Lags were chosen to be exponentially distributed over a certain range, so the collection of estimates form a log–lag autocorrelation. Estimates and their performance are compared to those of correlogram-based methods.
Recent advances in structural processing: Resolution of short transients of unknown parameters
Authors: Plotkin Eugene I. , Swamy M. N. S.
Published: October 2013

Resolution of transients of identical structures arises in such areas as machinery vibration, seismic and underwater acoustics. These processes are characterized by highly correlated nonstationary nature over a short interval. In extreme case, where the transients have the same structure and the record-length is short, even the most powerful adaptive methods become inefficient. Composite parameter-free modeling (CPFM) [E. I. Plotkin and M. N. S. Swamy, Int. J. Acoust. Vib. 4, 159–164 (1999)] is one of the effective techniques for improving the resolution of closely spaced (in time and frequency) short transients of unknown parameters. The model presented is a nested-form composition of null filters; the inner building blocks (variable-frame matched filters) are used to suppress one of the transients, while the outer nonlinear structure nullifies the second transient. The proposed model permits linear estimation of the target transient corrupted by almost identical targetlike interference. This model exhibits clear advantage in reconstructing a target signal in the presence of a powerful multi-tone transient of unknown parameters. The estimated envelope and phase of the target rapidly converge to their steady-state values, while the conventional approach produces prolonged lags with extensive fluctuations, precluding reliable reconstruction of the target signal. [Work supported by the NSERC.]
Short-time fractional Fourier methods for signal representation in the time-frequency plane
Authors: Capus Chris , Brown Keith
Published: October 2013

The analysis of multicomponent chirp signals is of interest in sonar, geophysical, bioacoustic, ultrasonic, and radar applications. This paper examines the interpretation of the fractional Fourier transform (FrFT) of such signals in terms of their representation in the time-frequency plane. The FrFT has been shown to be of value in analyzing and filtering linear chirp signals with various characteristics. The desirability of a short-time variant has been suggested by several authors and this paper introduces new short-time fractional Fourier methods for the piecewise description and parameterization of nonlinear chirp signals. Understanding of a signal’s FrFT as the projection of the joint time-frequency distribution onto a line at angle α to the time axis permits development of these short-time methods. Parameter α provides a means for transform optimization with reference to the entire signal or on a windowed basis. While providing much improved time-frequency resolution over the spectrogram, these techniques do not suffer from cross-term interference seen in Wigner distributions and related methods. Representation of highly nonlinear chirps in the time-frequency plane is demonstrated through analysis of a variety of signals including bat and dolphin chirps, heart sounds, and synthetic signals constructed to highlight particular differences between the techniques.
The development and use of an implantable radio whale tag from 1970 to the present
Authors: Wartzok Douglas , Maiefski Romaine R.
Published: October 2013

William A. Watkins, Woods Hole Oceanographic Institution (WHOI), led the development and refinement of the WHOI Radio Whale Tag. The initial tags were developed in the 1960s at WHOI, improved in the 1970s and 1980s by Ocean Applied Research, Inc., and adapted to satellite telemetry in the 1990s by Microwave Telemetry, Inc. Through this entire period, the tag delivery and attachment methods were continually refined. The same delivery and attachment methods were used with acoustic tags, radio tags, combined radio and acoustic tags, and satellite tags. Scientific studies using these tags have been conducted on finback, humpback, Brydes, bowhead, and sperm whales. These field studies have ranged from Venezuela to the Canadian arctic and Alaska, and from the Pacific to the Mediterranean. Every tagging experiment led to new information about the animals, and in most cases the information from radio and acoustic tagged whales showed that the best available data based on surface observations of untagged animals were deficient. Highlights from some of these findings will be presented.
Bill Watkins and pinniped bioacoustics
Authors: Carleton Ray G.
Published: October 2013

Bill’s work on pinnipeds began with a 1963 paper by Schevill, Watkins, and myself in Science on the underwater sounds of pinnipeds, which resulted from his recordings in the highly unnatural waters of the New York Aquarium. It was written partly in reaction to assumptions being made that seals, like cetaceans, possessed sonar. Following that mode, there followed a paper by the same team in the same location on the remarkable bell tones and knockings of captive walruses. Recording from natural situations, with Schevill in McMurdo Sound, Southern Ocean, resulted in the first analysis of Weddell seal sounds. In 1969, he provided the first technical description of a marine mammal song (in the behavioral meaning of that term), that of the bearded seal, and in 1975 a detailed description of walrus sounds in association with their reproductive behavior. Sounds of other polar pinnipeds, ribbon seals of the Arctic and Ross seals of the Southern Ocean, bring the history up to date. Pinnipeds possess a suite of underwater sounds that rivals any other group. They make up for apparent lack of echo-location ability by musical quality of voice.
How to label resonances from elastic shells
Authors: Uberall H. , Werby M. F.
Published: October 2013

A large class of resonances is observed from the excitation of elastic shells. A review of the type of resonances is presented. It is then shown how to label the resonances depending on material properties, ambient fluid, shape of object, and evacuation or fluid filling.
Vibration of in vacuo hemiprolate spheroidal shells
Authors: Hayek Sabih I. , Boisvert Jeffrey E.
Published: October 2013

The equations of motion for nonaxisymmetric vibration of hemiprolate spheroidal shells of constant thickness were derived using Hamilton’s principle. The thin shell theory used in this derivation includes shear deformations and rotatory inertias. The shell is clamped at the equator and is excited by mechanical surface force fields. The displacements and rotations were expanded in infinite series of comparison functions. These include associated Legendre functions in terms of the prolate spheroidal angular coordinate and circular functions in the azimuthal angle coordinate. Five-branched frequency spectra were computed for several shell thickness-to-length ratios ranging from 0.005 to 0.1, and for various diameter-to-length ratios, including the limiting case of a spherical shell. Numerical results were obtained for the frequency spectra of the driving and transfer mobilities of these shells due to surface force excitations. Some comparisons of the dynamic response of hemiprolate and full prolate spheroidal shells are presented. [Work supported by ONR and the Navy/ASEE Summer Faculty Program.]
A case study in optimizing TTS for Web content
Authors: Hagstrom Cynthia , Lin Qiguang
Published: October 2013

This is a case study of linguistic optimization of the Speechify text-to-speech (TTS) engine for use in the VocalPoint voice-browser and applications. To browse a website by phone the TTS engine must be able to handle two types of text, static system messages and dynamic content. The static messages include basic browser prompts such as the Default prompt: ‘‘What would you like?’’ or Help messages. In addition, the system must also handle dynamic content from the websites that are being browsed. The criteria of intelligibility, naturalness, and context-sensitivity were used to evaluate all the TTS output. Every static message was tested using a Speechify dialog tool. The TTS quality for dynamic content was evaluated through pronunciation bugs reported whilse browsing websites such as My Yahoo. Our study found that the text normalization module needs to be expanded to cover the vast variety of text on the Web. The study also suggested that three aditional functionalities would improve the caller experience: (1) make it possible to manipulate vowel length through punctuation or tagging; (2) make it possible to change intonation by manipulating pitch and phrase accents and tone breaks; and (3) add another phoneme to the sound inventory to allow flaps.
The unwarranted effectiveness of smoothness as a criterion for estimating vocal-tract length from speech
Authors: Neuburg Edward
Published: October 2013

Unambiguous quantitative characterization of speech sounds would be a useful thing. This talk will be about experiments done in an attempt to find such a characterization, by studying corpora of spoken vowel tokens all of which ‘‘sound alike.’’ Using Linear Predictive Coding (LPC) of order 10, 4 damped sinusoids (‘‘formants’’) were recovered from the tokens, and their bandwidths were algorithmically fixed as a function of frequency. (Other formant finders have also been tried.) Hypothesizing that a token is composed of 4 damped sinusoids is equivalent to hypothesizing that it was produced by a ‘‘vocal tract’’ with 8 sections; the shape (8 areas) of that tract is a function of its length, which is a free parameter of the formant-to-tract transformation. The experiments show that over a corpus of tokens of a given speech sound, simply choosing the vocal-tract length that produces the ‘‘smoothest’’ tract will make for shapes that are very nearly identical over all tokens, for all talkers (in particular for men and women), and are physiologically reasonable. Such 8-parameter shapes might be candidates for the desired characterization.
Sonar backscattering enhancements for small targets caused by caustics resulting from reflecting ocean surfaces
Authors: Dzikowicz Ben , Marston Philip L.
Published: October 2013

The reflection of sound by curved top or bottom ocean surfaces can produce caustics that enhance the sound level incident on a small target. Similarly, the scattering by the target reflected off a curved surface can be focused back onto the source transducer. These effects are simulated in a laboratory water tank experiment using a concave, cylindrical, Styrofoam surface and a small spherical target. The echo structure is simplest when the target is located at the cusp of the reflection caustic. A pulse can travel directly to the target and return or it can reflect off the surface on its way there, its way back, or both. For each reflection off the concave surface there is an observed enhancement so that the latest arriving pulse has a magnitude much larger than the earlier returns because it is doubly focused. The second return (though weaker than the doubly focused return) is also stronger than the direct echo. Application of a model for caustic wave fields [K. L. Williams et al., J. Acoust. Soc. Am. 96, 1687–1702 (1994)] enables simple predictions for the scaling of the relative amplitudes of the echoes with frequency. [Work supported by ONR.]
Broadband acoustic scattering from a bubbly liquid-filled compliant cylinder with known bubble population statistics
Authors: Wilson Preston S. , McCormick Ryan D. , Park Eun-Joo , Roy Ronald A. , Carey William M.
Published: October 2013

In previous experiments, acoustic scattering from a bubbly liquid-filled latex tube was investigated in the 5–20 kHz frequency range. The mean bubble resonance frequency was 113 kHz, and a Wood limit effective medium theory was found to quantitatively describe the scattering throughout the measurement range, using a void fraction as the model’s only fit parameter [Wilson et al., Acoust. Res. Lett. Online (in press)]. New measurements will be presented that extend the frequency range to 150 kHz and include in-situ void fraction measurement. A preliminary analysis of the new data shows qualitative agreement with the effective medium theory up to about 35–40 kHz. Above this range, the qualitative trends predicted by the model are no longer apparent, and an increased ping-to-ping variability in the received echo is observed. As the frequency approaches individual bubble resonance frequency, one proceeds from a continuum response where the bubble distribution is not important to a region where individual bubbles and groups of bubbles play a role. Moreover, as the frequency goes up, losses in the latex rubber tube wall should become increasingly important. An analysis of the data, which includes bubble resonance effects and tube wall losses, will be presented. [Work supported by ONR.]
The three-dimensional Fredholm equation of the first kind and its application to the inverse problem of submerged objects in a waveguide
Authors: Werby M. F.
Published: October 2013

Assume one has knowledge of the Green’s function in some ocean waveguide and a measurable signal scattered from a source. This may be treated by solving the Fredholm equation of the first kind (FEFK). The task is to determine any features from the measured field by solving the FEFK. We use synthetic data and produce measurents and determine what information may be recovered by solving this equation.
Velocity dispersion in granular marine sediment
Authors: Kimura Masao
Published: October 2013

Acoustic wave velocity dispersion in granular marine sediment is important to the study of sediment acoustics. The reported dispersion results cannot be explained with Biot’s model using constant Biot parameters. In this study, the acoustic wave velocities in water-saturated glass beads and beach sands with different grain sizes are measured at different frequencies. The velocities in air-saturated same media and in same media in vacuum are also measured. Next the frame moduli for water-saturated media, air-saturated media, and media in vacuum are obtained from these measured velocities. From these results, it is seen that the frame bulk modulus has dependence on the grain size and the frequency. It is considered that one possibility of the velocity dispersion in water-saturated granular marine sediment is grain size and frequency-dependent frame bulk modulus due to the gap stiffness between the grains.
Effects of sound source directivity on auralizations
Authors: Sheets Nathan W. , Wang Lily M.
Published: October 2013

Auralization, the process of rendering audible the sound field in a simulated space, is a useful tool in the design of acoustically sensitive spaces. The auralization depends on the calculation of an impulse response between a source and a receiver which have certain directional behavior. Many auralizations created to date have used omnidirectional sources; the effects of source directivity on auralizations is a relatively unexplored area. To examine if and how the directivity of a sound source affects the acoustical results obtained from a room, we used directivity data for three sources in a room acoustic modeling program called Odeon. The three sources are: violin, piano, and human voice. The results from using directional data are compared to those obtained using omnidirectional source behavior, both through objective measure calculations and subjective listening tests.
Pilot study: Exposure and materiality of the secondary room and its impact in the impulse response of coupled-volume concert halls
Authors: Ermann Michael , Johnson Marty E.
Published: October 2013

What does one room sound like when it is partially exposed to another (acoustically coupled)? More specifically, this research aims to quantify how operational and design decisions impact aural impressions in the design of concert halls with acoustical coupling. By adding a second room to a concert hall, and designing doors to control the sonic transparency between the two rooms, designers can create a new, coupled acoustic. Concert halls use coupling to achieve a variable, longer, and distinct reverberant quality for their musicians and listeners. For this study, a coupled-volume shoebox concert hall was conceived with a fixed geometric volume, form, and primary-room sound absorption. Aperture size and secondary-room sound-absorption levels were established as variables. Statistical analysis of sound decay in this simulated hall suggests a highly sensitive relationship between the double-sloped condition and (1) Architectural composition, as defined by the aperture size exposing the chamber and (2) Materiality, as defined by the sound absorbance in the coupled volume. Preliminary calculations indicate that the double-sloped sound decay condition only appears when the total aperture area is less than 1.5% of the total shoebox surface area and the average absorption coefficient of the coupled volume is less than 0.07.
Computer modeling of loudspeaker arrays in rooms
Authors: Schwenke Roger
Published: October 2013

Loudspeakers present a special challenge to computational modeling of rooms. When modeling a collection of noncorrelated sound sources, such as a group of musicians, coarse resolution power spectrum and directivities are sufficient. In contrast, a typical loudspeaker array consists of many speakers driven with the same signal, and are therefore almost completely correlated. This can lead to a quite complicated, but stable, pattern of spatial nulls and lobes which depends sensitively on frequency. It has been shown that, to model these interactions accurately, one must have loudspeaker data with 1 deg spatial resolution, 1/24 octave frequency resolution including phase. It will be shown that computer models at such a high resolution can in fact inform design decisions of loudspeaker arrays.
An investigation of platonic polyhedron loudspeaker systems as omnidirectional sources of sound
Authors: Leishman Timothy , Dix Gordon , Smith Heather , Tygerson Angela
Published: October 2013

For reasons of consistency, many room acoustics measurements (e.g., ISO 140-4 and ISO 3382) require the excitation of sound fields via omnidirectional sources. For years, acousticians have used dodecahedron loudspeaker enclosures with individual loudspeakers mounted on each face as conventional ‘‘omnidirectional’’ sources. However, frequency-dependent interference between loudspeakers, diffraction effects, and inherent directivities of individual loudspeakers must contribute to a departure from the omnidirectional ideal. As a result, criteria have been established to limit the maximum allowable deviations from omnidirectional radiation for qualified measurements. Despite such concessions, acousticians should consider an important question: is anything extraordinary about the dodecahedron loudspeaker configuration? The dodecahedron is merely one member of a family of five platonic polyhedrons. Other members of the family include the tetrahedron, hexahedron, octahedron, and icosahedron. How do the omnidirectional characteristics of loudspeaker systems based on these latter geometries compare to those based on the dodecahedron? This paper discusses several theoretical, numerical, and experimental results that provide answers to this question.
Trumpet synthesis using context-dependent envelopes
Authors: Dannenberg Roger B.
Published: October 2013

Synthesizing trumpet music in a natural-sounding way requires careful control. Even when synthesis is achieved by splicing together actual recorded trumpet tones, the result can sound artificial and unnatural. This is because natural notes are not played in isolation and are therefore influenced by neighboring notes and the musical context. In fact, a succession of notes played on the trumpet is likely to be a continuous sound with no separating silences. Improved synthesis can be obtained by calculating amplitude and frequency control envelopes that take context into consideration. In the combined spectral interpolation synthesis (CSIS) method, the spectrum is controlled by instantaneous frequency and rms amplitude. These, in turn, are controlled by envelopes computed by a rule-based system. To reduce the high dimensionality of envelopes (typically a vector of 200 samples per second), envelopes are specified by about ten parameters. This reduced set of parameters is computed from note attributes, most importantly, the duration and pitches of the current and preceding notes, and whether or not the notes are tongued or slurred. This procedure is described in detail, and synthesis results will be demonstrated.
Additivity of forward masking at low and high frequencies
Authors: Plack Christopher J. , O’Hanlon Catherine G.
Published: October 2013

Recent physiological and psychophysical measurements suggest that the response of the cochlea is compressive at high frequencies, but much more linear at frequencies below around 500 Hz. In this experiment, the dependence of compression on frequency was investigated using the additivity of forward masking. Two equally effective forward maskers should produce 3 dB of additional masking when they are combined assuming linearity with respect to intensity. A combined effect greater than this (excess masking) indicates compression. The signal was a 10-ms pure tone presented 20 ms after a 200-ms narrow-band noise masker and/or immediately after a 20-ms narrow-band noise masker. The signal frequency was 250, 500, or 4000 Hz. The signal threshold produced by adding two equally effective maskers was measured. At all three frequencies, no excess masking was observed for a signal 10 dB above absolute threshold, indicating linear additivity (no compression). At signal levels 30 dB above absolute threshold, excess masking was observed at all three frequencies. The estimated compression exponent was 0.31 at 250 Hz, 0.35 at 500 Hz, and 0.17 at 4000 Hz. Although the results suggest less compression at low frequencies than high, the estimated compression at low frequencies was greater than that reported previously.
Acoustical components of the Orpheum Theatre renovation
Authors: Conant David A.
Published: October 2013

The before and after acoustically-important listening conditions and measurements are described for a $14.2 million restoration and renovation of this 1,400-seat grand 1929 Movie Palace—at one time the tallest building in Phoenix. Great care was taken to restore all the acoustically good parts of the original design and to subtly modify the acoustically-troublesome parts (including severely focusing dome and sidewalls) so they looked the same but performed properly. A beautiful playhouse was achieved in 1997 with fine, fixed acoustics and conventional audio reinforcement, a surrounding Italian village, plenty of gilt, moving clouds and sunsets overhead. Today, the venue successfully hosts shows from performance art to ballet to jazz. It was not intended (acoustically) to serve classical music but does so on occasion.
Use of bubble cloud attenuation measurements to determine the bubble size distribution and comparison with backscatter measurements
Authors: Dumbrell Hugh
Published: October 2013

This paper presents the method of using bubble cloud attenuation measurements to determine the bubble size distribution, taking into account the effect of time-dependent bubble scatter and absorption. It then shows that the results of attenuation measurements due to bubble clouds (generated by a ship’s wake) agree with backscatter measurements made at a much higher frequency.
The acoustics of bubbles and bubbly assemblages: a suite of laboratory studies
Authors: Roy Ronald A. , Wilson Preston S. , McCormick Ryan D. , Park Eun-Joo , Carey William M.
Published: October 2013

The in situ study of oceanic bubble acoustics is a challenging endeavor. One must impose a variety of a priori assumptions and constraints in order to extract information that unambiguously sheds light on the underlying physical mechanisms that govern the transmission and scattering of sound in oceanic bubble assemblages. Conversely, laboratory studies afford an added measure of physical control and observational access that permits one to focus on certain facets of bubble activity with accuracy and precision. We provide an overview of several laboratory-based bubble acoustics studies under way at Boston University. The scope of the work spans the spectrum from the quantitative measurement of acoustic dispersion through, and scattering from, high void fraction bubbly regions, to detailed studies of the linear dynamics of individual bubbles. [Work supported by ONR.]
Measurement of 3-D directional characteristics of reverberant sound fields using a spherical microphone array
Authors: Gover Bradford N.
Published: October 2013

A new measurement system has been used to analyze directional characteristics of reverberant sound fields in several small- and medium-sized rooms. The system employs a pair of 32-element spherical microphone arrays to obtain the room impulse response in each of 60 steering directions. The two different-sized arrays are used to cover the frequency range from 300–3300 Hz with an angular (directional) resolution of about 28 degrees. An analysis of these directional room impulse responses enables the variation of arriving sound energy with (3-D) direction and time to be studied. Specifically, the isotropy of the sound field at the receiving position can be assessed at any time instant, or over any time range desired. Directions from which significant or insignificant proportions of sound energy arrive are identifiable. Furthermore, the time and direction of incidence of the direct sound and discrete early reflections can be identified. This can be of use in localizing sources, finding reflections, and in decomposing the sound field into arriving components.
Live interactive computer music performance practice
Authors: Wessel David
Published: October 2013

A live-performance musical instrument can be assembled around current lap-top computer technology. One adds a controller such as a keyboard or other gestural input device, a sound diffusion system, some form of connectivity processor(s) providing for audio I/O and gestural controller input, and reactive real-time native signal processing software. A system consisting of a hand gesture controller; software for gesture analysis and mapping, machine listening, composition, and sound synthesis; and a controllable radiation pattern loudspeaker are described. Interactivity begins in the set up wherein the speaker–room combination is tuned with an LMS procedure. This system was designed for improvisation. It is argued that software suitable for carrying out an improvised musical dialog with another performer poses special challenges. The processes underlying the generation of musical material must be very adaptable, capable of rapid changes in musical direction. Machine listening techniques are used to help the performer adapt to new contexts. Machine learning can play an important role in the development of such systems. In the end, as with any musical instrument, human skill is essential. Practice is required not only for the development of musically appropriate human motor programs but for the adaptation of the computer-based instrument as well.
Including attenuation and dispersion in time domain modeling of broadband sound propagation in dispersive oceanic media
Authors: Norton Guy V. , Novarini Jorge C.
Published: October 2013

Shallow water sound propagation almost invariably involves paths through regions occupied by assemblages of microbubbles near the air/water interface and through the underlying sediment. Such an environment is intrinsically dispersive. Any reliable numerical model of acoustic propagation in this type of environment requires the inclusion of attenuation and its causal companion, dispersion. The time domain is the natural domain for numerical modeling of broadband pulse propagation. The inclusion of causal attenuation and dispersion within a numerical time domain model has in the past presented a challenge to modelers. For the case of propagation in a linear medium, Szabo [T. L. Szabo, J. Acoust. Soc. Am. 96, 491–500 (1994)], making use of generalized functions, introduced the concept of a convolutional propagation operator that plays the role of a causal propagation factor in the time domain. Utilizing a Finite Difference Time Domain (FDTD) model it is shown that the inclusion of the convolutional propagation operator correctly carries the information of attenuation and dispersion into the time domain. The causal FDTD model was exercised on environments having functional forms for the attenuation of the type encountered at sea, both in typical bubble clouds as well as the underlying sediment. [Work supported by ONR/NRL.]
Should visual speech cues (speechreading) be considered when fitting hearing aids?
Authors: Grant Ken
Published: October 2013

When talker and listener are face-to-face, visual speech cues become an important part of the communication environment, and yet, these cues are seldom considered when designing hearing aids. Models of auditory–visual speech recognition highlight the importance of complementary versus redundant speech information for predicting auditory–visual recognition performance. Thus, for hearing aids to work optimally when visual speech cues are present, it is important to know whether the cues provided by amplification and the cues provided by speechreading complement each other. In this talk, data will be reviewed that show nonmonotonicity between auditory-alone speech recognition and auditory–visual speech recognition, suggesting that efforts designed solely to improve auditory-alone recognition may not always result in improved auditory–visual recognition. Data will also be presented showing that one of the most important speech cues for enhancing auditory–visual speech recognition performance, voicing, is often the cue that benefits least from amplification.