bluuebunny/crossref_metadata_2025 huggingface dataset:
Can monaural temporal masking asymmetry explain the transient and/or ongoing precedence effect?
Authors: Freyman Richard L. , Morse-Fortier Charlotte , Griffin Amanda M. , Zurek Patrick M.
Published: May 2013

Investigations of the precedence effect show that interaural differences within the first of two pairs of brief binaural sounds contribute more to lateralization than those within the second pair. The present study asked whether this phenomenon could be explained by asymmetries in monaural masking, and compared the results to a second experiment investigating the same question with the “ongoing” precedence effect. Leading and lagging stimuli were binaural pairs of 1-ms frozen noise bursts, with a 2-ms delay between pairs, and had ITDs of +500 and −500 ms, respectively. Detection thresholds for lead or lag in the presence of the other were assessed monaurally using a 4AFC task. Results showed that threshold for the lead was 12 dB better than that for the lag, on average. Ongoing stimuli were created by repeating the transient stimuli 63 times, with a new noise token on each repeat. Although the leading burst in each binaural pair contributes more to lateralization than the lagging burst, monaural thresholds for the leading bursts were not better than those for the lagging bursts. The results suggest that the transient precedence effect, but not the ongoing precedence effect, might be explained by temporal masking asymmetry. [Work supported by NIH DC01625].
Frequency domain binaural model with front-back discrimination capability using artificial neural network
Authors: Usagawa Tsuyoshi , Tomita Takuro , Chisaki Yoshifumi
Published: May 2013

Conventional binaural model or binaural hearing assistance systems have a well known ambiguity in front-back discrimination, which is called as ”front-back confusion” or ”cone of confusion” in psychoacoustics. It is known that spectral cue of sound provides keys to solve this confusion in binaural listening condition and the peaks and notches of spectral components play main role to estimate the vertical angle in sagittal coordinate. In this paper, a frequency domain binaural model with front-back discrimination method using artificial neural model is proposed and examined for various HRTF catalogs. The performance of the model is examined for simulated conditions using various HRTF catalogs and the results show similar discrimination capability even if the learning process should be done for each catalogs. The experimental results using a dummy head is also provided.
Effect of distant-variant/invariant head-related transfer functions on perception of a proximal sound source in virtual auditory space
Authors: Otani Makoto , Hirata Fuminari , Itoh Kazunori , Hashimoto Masami , Kayama Mizue
Published: May 2013

A virtual auditory space can be presented to a listener based on binaural synthesis using head-related transfer functions (HRTFs) that are obtainable by measurements or numerical simulations. Due to hardware complexity, HRTF measurements are typically made for a fixed source distance though they are used in binaural synthesis for variable source distances. However, it is known that HRTFs depend on source distance especially for proximal sources for distance less than 1 m. So it is possible that binaural synthesis with HRTFs for a fixed source distance may result in degradations for proximal sound image perception. In this paper, experiments were performed to examine how the use of distant-variant or -invariant HRTFs affect the perception of a proximal sound source in a virtual auditory space in which the listener’s motion is compensated by head tracking. HRTFs for source distances up to 1 m, in 5 cm steps, are numerically simulated using the boundary element method. Results show the difference between presented and perceived source distances being significantly smaller when distance-variant HRTFs were used. This indicates that the use of HRTFs corresponding to actual sound source position leads to accurate perception of a proximal source.
The relation between the information delivered by head-related transfer function and human spatial hearing
Authors: Tourbabin Vladimir , Rafaely Boaz
Published: May 2013

The human auditory system is capable of performing various tasks related to spatial hearing including sound localization and source segregation. The performance of these tasks depends on many factors, including the complexity of the sound field; the way in which the information from the sound field is transferred to the ears; and the ability of the binaural hearing system to extract the required information. This study focuses on the role of the transfer system represented by the HRTFs that relate the sound field to the signals at the ears. Previously, a measure for the information delivered by the HRTFs was proposed, and the role of HRTFs in human sound localization in the horizontal and median planes was investigated. In the current study, the role of HRTFs in human sound localization is further investigated by analyzing localization in the entire three-dimensional space. Then, the proposed measure is used to investigate the role of HRTFs in source segregation as part of a spatial-release from masking task. The results show that the information delivered by the HRTFs can account for the increase in intelligibility as a function of the spatial separation between the desired source and the masker reported in the literature. However, the decrease in intelligibility with increasing number of spatially separated maskers seems to be related to the binaural hearing system, and cannot be explained by the information in the HRTFs.
An efficient finite-impulse-response filter model of head-related impulse response
Authors: Li Junfeng , Zhang Jian , Sakamoto Shuichi , Suzuki Yoiti , Yan Yonghong
Published: May 2013

Head-related impulse responses (HRIRs) play a crucial role in binaural 3-D audio rendering. The HRIRs with a couple of hundred-sample lengths result in the high computation cost for the real-time 3-D audio applications especially when multiple sound sources are rendered simultaneously. To overcome this problem, various modeling approaches have been reported to reduce the number of parameters of HRIRs without sacrificing the quality of rendered sounds. In this research, an efficient finite-impulse-response (FIR) model is first studied, which is essentially based on the concept of the minimum-phase modeling technique. In this method, the measured HRIRs are represented by the interaural time delay (ITD) and the magnitude spectra that are approximated by two FIR filters. To investigate the accuracy dependence of this modeling approach on the order of FIR filter, two psychoacoustic experiments on sound localization and sound quality were conducted by comparing the synthesized stimuli with the measured HRIRs and those with the FIR models of different orders. Experimental results indicated that the measured hundred-sample-length HRIRs can be sufficiently modeled by the low-order (a dozen of coefficients) FIR model from the perceptual point of view. The derived low-order FIR model can be further applied to real-time 3-D audio applications.
Further evidences of the contribution of the ear canal to directional hearing: Design of a compensation filter
Authors: Martelloni Andrea , Mauro Davide A. , Mancuso Antonio
Published: May 2013

It has been proven, and it is well documented in literature, that the directional response in HRTFs comes largely from the effect of the pinnae. However, few studies have analyzed the contribution given by the remaining part of the external ear, particularly the ear canal. This work investigates the directionally dependent response of the modeled ear canal of a dummy head, assuming that the behavior of the external ear is sufficiently linear to be approximated by an LTI system. In order to extract the ear canal’s transfer function, two critical microphone placements (at the eardrum and at the beginning of the cavum conchae) have been used. The system has been evaluated in several positions, along the azimuth plane and at different degrees of elevation. The results point out a non-negligible directional dependence that is well within the normal hearing range; based on these findings, physical models of the ear canal have been analyzed and evaluated. We have also considered the practical application to binaural listening, and the coloration originated by the superimposition of the contribution of two ear canals (the listener’s and the dummy head’s). A compensating FIR filter with arbitrary frequency response is discussed as a possible fix.
Modeling of wave propagation in drill strings using acoustic transfer matrix method
Authors: Han Je-Heon , Kim Yong-Joe , Karkoub Mansour
Published: May 2013

In order to understand critical vibrations of a drill bit such as stick-slip and bit-bounce and their wave propagation characteristics through a drillstring system, it is critical to model the torsional, longitudinal, and flexural waves. The objective is to model these waves propagating through the drillstring in a computationally efficient way. Here, a modeling method based on an acoustic transfer matrix between two sets of wave variables at the ends of a cylindrical pipe is proposed. For a drillstring system with multiple pipe sections, the total acoustic transfer matrix is calculated by multiplying all individual matrices of which each is obtained for an individual pipe section. Since drillstring systems are typically extremely long, conventional numerical analysis methods such as FEM require a large number of meshes, which makes it difficult to analyze these drillstring systems. On the contrary, the “analytical” acoustic transfer matrix method requires significantly low computational costs. For the validation, experimental and numerical data are obtained from a laboratory measurement and by using a commercial FEM package, ANSYS, respectively. They are compared to the modeling results obtained by using the proposed method. It is shown that the modeling results are well matched with the experimental and numerical results.
Some research of mapped radiation modes and its application in analyzing the radiation surface of vibrating structures
Authors: Wu Haijun , Jiang Weikang , Pan Siwei
Published: May 2013

Acoustic radiation modes divide the velocity patterns into groups with effective and ineffective radiation efficiencies. It suggests a promising way in the computation of sound power and near field acoustic holography. However, obtaining the radiation modes based on eigenvalue analysis suffers from issues of CPU time and memory, especially for large-scale problems. Based on the idea of equivalent source method and boundary integral equation, it is mathematically and numerically proved that the spherical basis functions is a set of linearly independent patterns on an arbitrary surface. They are termed as mapped radiation modes. The analytical sound power of a structure vibrating in its mapped radiation modes is obtained. An efficient and accurate method to compute the sound power based on the mapped radiation modes is proposed. Based on the relationship between the mapped radiation modes of a vibrating structure and its field sound pressure distribution patterns on a sphere, an improved near field acoustic holography is also developed. It does not need the inverse process but requires microphones to locate at the integral points on a sphere. Numerical examples are presented to validate advantages of applying mapped radiation modes in the sound power evaluation and near field acoustic holography.
Experimental investigation into sound and vibration of a torpedo-shaped structure under axial force excitation
Authors: Leader James , Pan Jie , Dylejko Paul , Mathews David
Published: May 2013

In this study, the sound radiation patterns and vibration characteristics of a torpedo-shaped structure are determined experimentally using a proof mass actuator to allow pure axial excitation of the model. Using this method, the second energy path found in previous designed structures is eliminated. Input power and driving forces are measured using four force transducers and four accelerometers, while the vibration response and mode shapes are measured using an array of accelerometers. The sound pressure and its directivity are captured by a spatially distributed microphone array inside an anechoic chamber. Motivations for this work are to investigate the effect of the complex boundary constraints: a semispherical head and conical tail on the two meter long model when compared to existing analytical solutions for simple geometries, and later the measurement will be performed in an underwater experiment to contrast the effect of fluid loading.
Results of an implementation of the dual surface method to treat the non-uniqueness in solving acoustic exterior problems using the boundary element method
Authors: Burgschweiger Ralf , Schäfer Ingo , Mohsen Adel , Piscoya Rafael , Ochmann Martin , Nolte Bodo
Published: May 2013

The problem of non-uniqueness (NU) of the solution of exterior acoustic problems when using the boundary element method (BEM) is well known. Methods like the Burton-Miller technique or the CHIEF method are used to solve this challenge at the expense of more complex procedures for handling hypersingular integrals and/or higher computing times due to higher complexity of the algorithm or additional equations. The dual surface method, commonly used for electromagnetic problems, was adapted for acoustic radiation and scattering problems. The basic principles of methods to solve the NU problem are outlined and results for different models and solution procedures are presented, taking into account quality, solution time, and the numerical advantages when using iterative solvers.
The peculiarities of the non-axisymmetric frequency spectra of finite elastic cylinders
Authors: Libov Dmytriy
Published: May 2013

A rigorous solution of three-dimensional boundary-value problem concerning the forced vibrations of a finite, elastic, isotropic cylinder is constructed analytically by means of the superposition method. With this solution, the resonances in non-propagating waves were investigated. Particularly, a survey of the frequency spectrum for an aluminum cylinder, vibrating with the circumferential order two, reveals the existence of a localized resonance, usually referred to as an end resonance, well below the cut-off frequency of the lowest real dispersion branch of an infinite cylinder. This phenomenon demonstrates the remarkable differences between the axisymmetric and non-axisymmetric end resonances of elastic cylinders. Comparison of the theoretical results with the experiments published elsewhere reveals an excellent agreement.
Sound generated by a wing with a flap interacting with an eddy
Authors: Manela Avshalom , Huang Lixi
Published: May 2013

Acoustic signature of a rigid wing, equipped with a movable downstream flap and interacting with a line vortex, is studied in a two-dimensional low-Mach number flow. The flap is attached to the airfoil via a torsion spring, and the coupled fluid-structure interaction problem is analyzed using thin-airfoil methodology and application of the Brown and Michael equation. It is found that incident vortex passage above the airfoil excites flap motion at the system natural frequency, amplified above all other frequencies contained in the forcing vortex. Far-field radiation is analyzed using Powell-Howe analogy, yielding the leading order dipole-type signature of the system. It is shown that direct flap motion has a negligible effect on total sound radiation. The characteristic acoustic signature of the system is dominated by vortex sound, consisting of relatively strong leading and trailing edge interactions of the airfoil with the incident vortex, together with late-time wake sound resulting from induced flap motion. In comparison with the counterpart rigid (non-flapped) configuration, it is found that the flap may act as sound amplifier or absorber, depending on the value of flap-fluid natural frequency. The study complements existing analyses examining sound radiation in static- and detached-flap configurations.
Biomechanical models of damage and healing processes for voice health
Authors: Granados Alba , Brunskog Jonas , Jacobsen Finn
Published: May 2013

In voice-loading occupations, employees are required to use their voice for continuous and large periods of time, which might lead to voice problems. In this work, anomalous vocal-fold vibrations due to long-time high voice-load are investigated. Laryngeal endoscopic high-speed images within the vocal-fold plane are available. These data are used to improve existing continuum biomechanical models of the vocal-folds by analyzing the injury processes. The project is expected to result in methods that objectively demonstrate the impact of high voice-load on voice. A detailed description of the currently developing work will be presented, including a rigorous analysis of the hypothesized injury processes of the vocal folds.
Perceptual consequences of changes in epilaryngeal area and shape
Authors: Samlan Robin A. , Kreiman Jody E.
Published: May 2013

Decreasing epilaryngeal area has been shown to increase glottal flow pulse skewing and harmonic amplitudes [Titze, J. Acoust. Soc. Am. 123, 2733 (2008)]. It is not known, however, whether listeners perceive voice quality changes when epilaryngeal area is altered, or if perceived quality is different if the area change occurs at the ventricular folds or aryepiglottic (AE) folds. In this study, a kinematic vocal tract model was used to create five epilaryngeal cavity shapes resulting from constriction and retraction of the ventricular and AE folds. Four voice sources simulating varying degrees of vocal deviation were filtered through the five shapes for a total of 20 stimuli. Fourteen listeners completed a sort and rate task. Results were analyzed using multidimensional scaling (MDS). Altering the epilaryngeal cavity shape resulted in voice quality differences, and perceptual distances differed by voice source. AE fold constriction was perceived most differently from other shapes for all talkers. Ventricular fold constriction was perceived most similar to AE constriction for 3 of the 4 voice sources. Glottal flow and acoustic differences for each epilaryngeal shape will be described and related to the perceived differences in voice quality.
Variation of maximal Lyapunov exponent with voice disorders of pilots
Authors: Ruiz Robert , Plantin de Hugues Philippe , Legros Claude
Published: May 2013

The maximal Lyapunov exponent λ is a signature of chaos in the field of nonlinear dynamics. Analysis of vowels uttered by a troubled speaker can show irregularities and instabilities due to nonlinearities of the phonatory system. Therefore λ can be studied for the research of voice acoustic features able to present variations due to psychophysiological disturbances. These ones belong to the aeronautical context. Two pilots’ voices have been recorded at stopovers during short-haul rotations on a day. A day of driving was used as an experimental material to study a similar workload. After being woken up in a laboratory sleep inertia experiment, another pilot is recorded. Finally, the Cockpit Voice Recorder of a crashed airplane provided a real-case corpus for the study. Vowels are extracted in all recordings and their maximal Lyapunov exponent is estimated. Regardless of whether or not the chaotic behavior of the voice, results show a large dispersion, little variations with different directions from the normal state of the speaker to the end of the recordings. On the basis of these experiments, the number of speakers involved, the choice of the calculation parameters, the phonetic material used, λ has a low sensitivity to the aeronautical psychophysiological disturbances.
Vowel synthesis using a vocal tract mapping interface and simulation study of inverse mapping
Authors: Ogata Kohichi , Hayakawa Tomohiro
Published: May 2013

We have developed a vocal tract mapping interface to produce vowel sounds. The interface provides an easy and effective setting of the vocal tract shape with a simple mouse click on its interface window. A vocal tract shape at the mouse position on the interface window is calculated by interpolation based on the prepared vocal tract shapes. In this paper, the features and advantages of the interface are shown through examples of the generated vowel sounds and vocal tract shapes. In addition, the inverse estimation of the vocal tract shape from formant frequencies is studied as one of its applications. In this method, the vocal tract shape is estimated by searching the point inside the region after determining a possible region that includes the solution. The possible region on the interface window is determined based on the changes in the formant frequencies. The usefulness of the proposed method is shown through simulation results.
Evaluation of bone-conducted ultrasonic hearing-aid regarding transmission of phonetic features
Authors: Kagomiya Takayuki , Nakagawa Seiji
Published: May 2013

Human listeners can perceive speech signals in a voice modulated ultrasonic carrier from a bone-conduction stimulator, even if the listeners are patients with sensorineural hearing loss. Considering this fact, we have been developing a bone-conducted ultrasonic hearing aid (BCUHA). The purpose of this study is to evaluate the ability of BCUHA to transmit Japanese distinctive features to the recipient. For this purpose, a series of mono-syllable intelligibility experiments was conducted. A series of sequential information transfer analyses (SINFA) were carried out to analyze what kind of articulatory features were well transmitted. Results of the SINFA showed that: in vowel perception, “openness” and “frontness” were well transmitted, while in consonant perception, Japanese “you-on” (palatalized sound) feature was well transmitted; however, transmission of other features like articulatory position or manner was limited. These results indicated that the BCUHA has sufficient frequency resolution to transmit vowel information, while some signals are masked by carrier sound. To improve this problem, further investigation and development is required.
Spatial sound pick-up with a low number of microphones
Authors: Palacino Julian D. , Nicol Rozenn
Published: May 2013

Portable audio devices have become more and more popular during the last decade. Recent advances in audio compression and electronic miniaturization allow people to keep all their music and movies at hand, at all times. Nowadays people are using their smartphones everywhere as photo or video cameras. In order to provide a consumer possibility of 3D audio recording adapted to these kinds of devices, we developed a compact microphone array able to pick-up a full 3D sound scene, using less than four microphones. To compensate for the low number of microphones, which results in poor spatial selectivity, spatial post processing is applied to the microphone signals and improves the sound source localization. Another advantage is that the post processing makes the sound reproduction flexible. The 3D audio scene can be converted into any format to be rendered to any equipment and any device. The paper will describe various recording set-ups in combination with the associated post processing. As a first assessment, their performances in terms of sound localization accuracy will be compared using a new set of objective criteria descriptors.
Heading and hydrophone data fusion for towed array shape estimation
Authors: Odom Jonathan , Krolik Jeffrey
Published: May 2013

This paper addresses the problem of towed array shape estimation for passive, horizontal sonar arrays. Beamforming and localization techniques significantly degrade when an assumed linear array bends due to tow platform maneuvers or ocean currents. In this paper, heading sensors along the array and acoustic hydrophone data are jointly used to estimate the shape of the array. Previously, heading data have been filtered using a dynamical motion model to reduce noise during turns. In recent work, a time-varying noise field directionality estimate that incorporates a dynamical model for the acoustic field provides a second, albeit biased, estimate of the array shape. In this paper, these two estimates are combined via adaptive weights to obtain improved shape estimates during maneuvers. A multi-source simulation is used to demonstrate the robustness of the combined array shape estimate when compared to the separate heading or acoustic sensor based techniques.
Signal processing for hemispherical measurement data
Authors: Müller-Trapet Markus , Vorländer Michael
Published: May 2013

To realistically model the sound propagation in rooms, a detailed knowledge of the reflection properties of the surrounding surfaces is required. In this context, the reflection properties include both the sound absorption as well as scattering. In order to be able to measure the angle-dependent reflection properties of surfaces in-situ, a hemispherical microphone array was recently designed and built. For a reduction of the required hardware an efficient, rotationally symmetric sampling was chosen, so that 28 microphones on two concentric semicircles are employed to measure a total of over 3000 positions on a hemisphere in just over 15 minutes. This contribution will give an overview over the required signal processing steps to process the measurement data from such a microphone array. Special emphasis will be placed on the determination of the microphone positions and the special case of data available on a hemispherical surface. Also, the sound field model used to determine the impedance on the surface will be explained. Preliminary results will be presented.
Fly-over aircraft noise measurement campaign at Montreal-Trudeau airport using a microphone array
Authors: Blais Jean-Francois , Camier Cédric , Patenaude-Dufour Mathieu , Lapointe Robby , Provencher Jonathan , Padois Thomas , Gauthier Philippe-Aubert , Berry Alain
Published: May 2013

Engines being quieter due to high by-pass ratios, the airframe noise, produced for instance by landing gears or high-lift devices, has become a significant contributor to the total noise radiated by aircraft during approach and landing. As part of the investigations carried out to understand noise generation mechanisms, the beamforming techniques developed over the last decade and applied to microphone array measurements have shown to be effective tools for localization and quantification of these aerodynamic noise sources. In order to validate their in-house beamforming softwares, Bombardier Aerospace and the Groupe d’Acoustique de l’Université de Sherbrooke have conducted a 5-day measurement campaign in June 2012. The 95-microphone array was located on the roof of a building next to the Montreal-Trudeau airport. Aircraft position was determined by two high-definition cameras, both synchronized with the microphone array by inter-range instrumentation group time codes generators. This paper summarizes the measurement campaign. The aircraft tracking tool and the beamforming algorithms used to characterize the noise sources are presented. Several Bombardier CRJ fly-overs were recorded during this test. Beamforming results obtained for different airlines are compared in order to evaluate the repeatability of the method.
Accuracy of head-related transfer functions synthesized with spherical microphone arrays
Authors: Salvador Cesar , Sakamoto Shuichi , Trevino Jorge , Li Junfeng , Yan Yonghong , Suzuki Yôiti
Published: May 2013

The spherical harmonic decomposition can be applied to present realistically localized sound sources over headphones. The acoustic field, measured by a spherical microphone array, is first decomposed into a weighted sum of spherical harmonics evaluated at the microphone positions. The resulting decomposition is used to generate a set of virtual sources at various angles. The virtual sources are thus binaurally presented by applying the corresponding head-related transfer functions (HRTF). Reproduction accuracy is heavily dependent on the spatial distribution of microphones and virtual sources. Nearly uniform sphere samplings are used in positioning the microphones so as to improve spatial accuracy. However, no previous studies have looked into the optimal arrangement for the virtual sources. We evaluate the effects of the virtual source distribution on the accuracy of the synthesized HRTF. Furthermore, our study considers the impact of spatial aliasing for a 252-channel spherical microphone array. The microphone's body is modeled as a human-head-sized rigid sphere. We evaluate the synthesis error by comparison with the target HRTF using the logarithmic spectral distance. Our study finds that 362 virtual sources, distributed on an icosahedral grid, can synthesize the HRTF in the horizontal plane up to 9 kHz with a log-spectral distance below 5 dB.
Using warping processing to range bowhead whale sounds from a single receiver
Authors: Bonnel Julien , Thode Aaron
Published: May 2013

In certain shallow water environments, the acoustic propagation of low-frequency marine mammal calls can be well-modeled as a discrete set of normal modes. Each mode propagates with a different group velocity, and thus in principle the range of the call can be inferred by comparing relative arrival times of the modal arrivals. Traditionally, several time-synchronized hydrophones are required to spatially filter out individual modes in order to measure relative arrival times. In this presentation a nonlinear signal processing method classed “warping” is used to identify individual mode arrival times on a single receiver, even when the mode arrivals are overlapping in time. Warping processing is limited to frequency-modulated sources with monotonic increases or decreases of frequency with time. It is thus applicable to whale calls that consist of simple frequency-modulated upsweeps or downsweeps. Once the modes are separated, the source range can be estimated using conventional modal dispersion techniques. This method is applied on several bowhead whale vocalizations recorded near Kaktovik (Alaska) in 2012. Bowhead whale calls are ranged up to 35 km under median ambient noise conditions. These single-receiver range estimates are consistent with estimated ranges previously obtained via other methods [Work supported by North Pacific Research Board and Shell Exploration and Production Company.]
Source signature characterization and feature based detection of open-circuit SCUBA regulators
Authors: Gemba Kay L. , Nosal Eva-Marie
Published: May 2013

A goal of the Center for Island, Maritime, and Extreme Environment Security is to develop passive acoustic methods to monitor harbor and near-shore environments. To study detection of open circuit SCUBA divers, several regulator configurations were recorded and characterized under ideal conditions in a pool environment. Sound Pressure Levels (SPL) and Sound Spectral Levels were calculated over the 4 kHz to 80 kHz band. Results include SPL range over all recordings along with variation of SPL due to change in SCUBA tank pressure and regulator flow rate. Regulator broad-band signatures were used to test three diver detection algorithms under varying synthetic and real ambient noise conditions. The first detector is a broad band energy detector exploiting a priori knowledge of the underlying signal length. The other two detectors are an envelope and a cepstral detector which estimate the divers’ fundamental breathing frequency and require at least two breaths for a positive result. In order to maximize SNR, regulator signatures were band pass filtered to exploit their respective dominant features. Receiver operating characteristic curves were calculated to compare detector performances. [Work funded by the U.S. Department of Homeland Security through the Center for Island, Maritime, and Extreme Environment Security.]
A numerically stable rational approximant for the split-step Padé propagator
Authors: Roberts David E. , Thomson David J.
Published: May 2013

The split-step Padé algorithm due to Collins [J. Acoust. Soc. Am. 93, 1736–1742 (1993)] provides a fast and accurate method for solving the parabolic equation (PE). The formal solution to the PE propagator, which involves the pseudo-differential operator (1+X)1/2, is replaced by an [n/n]-Padé rational approximant. This approximant can be expanded either as a product or a sum of rational-linear terms, each term leading to a tridiagonal system of equations in X which is readily solved numerically. To ensure adequate suppression of undesirable contributions from the evanescent part of the spectrum (X < −1), stability constraints must be imposed. In contrast to this approach, we follow the suggestion of Lu and Ho [Optics Lett. 27, 683–685 (2002)] and examine the use of an [n−1/n]-Padé approximant that inherently dampens these evanescent components of the spectrum. An algorithm for generating the necessary coefficients is described. Transmission losses computed using both rational approximants are compared for a typical shallow-water configuration.
Applicability of two-dimensional boundary scattering models as a proxy for three-dimensional models
Authors: Tran Bryant M. , Joshi Sumedh M. , Isakson Marcia J.
Published: May 2013

Three-dimensional numerical models offer unique insight into the nature of scattering from rough surfaces. However, use of these models is computationally prohibitive for any application more time-sensitive than basic research. This work seeks to determine a proxy for full three-dimensional rough boundary scatting models using appropriate two-dimensional models. Specifically, a Monte Carlo Kirchhoff approximation model in 2D with a derived proxy relationship applied is compared to a similar model in 3D. The region of validity of the proxy will be explored. The usage of the proxy function when applied to an finite element method model will also be discussed. [Work supported by ONR Ocean Acoustics.]
Modeling (non-)uniform scattering distributions in geometrical acoustics
Authors: Schröder Dirk , Pohl Alexander
Published: May 2013

In most cases, a surface is not ideally smooth. It rather contains regular and irregular dents, bumps, and other textures that influence the reflection of the incident wave. A reflection on such a corrugated surface causes a frequency-dependent redirection of the incident sound energy outside the specular direction, called scattering. While the computation of the specular part is well elaborated today, a model that thoroughly captures the wave phenomenon of scattering is still under discussion. Here, the most common assumption is that scattered energy follows a uniform Lambert distribution, which has proven to be a good approximation, especially in room acoustical applications. In this contribution, we will discuss Lambert-based scattering models (specular/diffuse sound field decomposition and vector mixing) and their implementations in methods of Geometrical Acoustics. We will analyze benefits and flaws of the respective models and investigate possibilities to introduce angle-dependent scattering for use cases where the uniform Lambertian distribution becomes invalid.
Numerical models for predicting absorption/insulation performance of acoustic elements
Authors: Inoue Naohisa , Sakuma Tetsuya
Published: May 2013

With a great improvement of computer resource availability, numerical analysis is widely used to investigate acoustic characteristics of various materials. A further expectation will be to predict absorption/insulation performance of acoustic elements used for buildings, automobile and so on, which can be the alternative to the actual measurements. This paper presents general numerical models for predicting the absorption coefficient and the transmission loss of acoustic elements with arbitrary shape and material composition. The features of the models are: (1) A test sample is mounted in the cavity or aperture on a thick rigid baffle: (2) FEM is employed for the materials and the air in the cavity, and coupled with sound fields out of the baffle by BEM: (3) Acoustical indices are calculated from the incidence power and the absorption/transmission power on the interfaces. Numerical simulation demonstrates the oblique incidence absorption coefficients and transmission losses of single- and multi-layered materials, and the influence of the sample's area is discussed in comparison with theoretical values for the infinite area. Additionally, the influence of the thickness of the cavity/aperture is examined, which is so-called “niche-effect.”
Effects of flooring, topping, and underlayment on impact sound insulation of wood-joisted floor-ceiling assemblies
Authors: Hu Lin , Omeranovic Anes , Dufour Richard
Published: May 2013

Footstep impact noise transmission through floor-ceiling assemblies is a major source of complaints in wood-framed multi-family buildings. Experience shows that adding a finishing-topping-underlayment sandwich on a base floor-ceiling assembly significantly affects the noise transmission. So far, no reliable tool has been developed for designing the proper sandwich and most proposed solutions rely basically on trial and error. FPInnovations has launched a major research project to develop such tools. First phase of the project is focused on understanding the effects of flooring, topping and underlayment on the impact sound insulation. A mock-up assembly simulating a pair of typical stacked rooms was built. Standard field impact sound transmission tests were conducted on floors topped with a 1.2 m by 1.2 m four-layer sandwich patch. By varying the combination of flooring, topping and underlayment, over 50 patches were tested to better understand the effects of the type of materials in each layer on the assembly’s overall impact sound insulation. Based on a large number of tests conducted so far, it is evident that proper combination of flooring, topping and underlayment produces satisfactory impact sound insulation on wood-joisted floor-ceiling assembly. Verification testing on the assembly fully covered by the 4-layer samples is under way.
A study of a real world transmission loss chamber and the Kinetics UniBrace-L technology
Authors: Hulteen Scott , McGowan Eric , Chéenne Dominique J.
Published: May 2013

This study performed at Columbia College Chicago (CCC) had two purposes: The first was to test the functionality of the recently developed real-world transmission loss (RWTL) chamber, while the second was to evaluate the performance of the Kinetics UniBrace-L product on a double wall construction assembly. CCC’s RWTL chamber is designed to illustrate issues that have influence when testing partitions in the field. It is smaller in volume than a certified STC chamber, which results in modal effects on both sides of the chamber. Numerous microphone positions are available and are used to display the modal effects of the rooms and to determine an average sound pressure level in both spaces during testing. Absorption values are also substantially different between the sending and the receiving spaces (each side can be switched as either sending or receiving room) and diffuse-field conditions are not achieved in either side of the chamber. As such, the RWTL chamber will typically yield a lower STC value than a certified STC Chamber and will also yield lower values than what may be experienced when performing a test that would follow the standard ASTM E-336 or associated procedures.
Calling in gray treefrog choruses: Modifications and mysteries
Authors: Bee Mark A. , Schwartz Joshua J.
Published: May 2013

Frogs are well known model systems in the study of communication for investigating the influences of noise on both signaling behavior and auditory processing. The best-studied frogs in this regard are two sister-species in the Hyla versicolor species complex (H. versicolor and H. chrysoscelis). Males of both species produce loud, pulsatile advertisement calls that function to attract females. In the competitive social environment of a breeding chorus, males commonly shift to producing longer calls (with more pulses) at slower rates when the level of competition increases. These behavioral modifications can be evoked in controlled laboratory experiments using playbacks of calls and chorus-shaped noise. In contrast to birds and mammals, however, there is no evidence that males increase the amplitude of their vocalizations (the Lombard Effect) in response to increasing noise levels. In addition, current evidence suggests that males do not necessarily profit significantly from producing longer calls at slower rates in terms of increasing their overall attractiveness to females, overcoming interference by overlapping calls, or increasing the detectability of their calls in noise. Despite the robust and directional nature of call modifications in noise, the evolutionary function of these modifications remains obscure.
Modification of humpback whale social sound repertoire and vocal source levels with increased noise
Authors: Dunlop Rebecca , Noad Michael , Cato Douglas
Published: May 2013

In acoustic communication, high background noise is an important obstacle in successful receiver signal detection and perception of an intended acoustic signal. To overcome this problem, many animals modify acoustic signals by increasing the repetition rate, duration, amplitude, or frequency range of the signal. Humpback whales are the most vocal of the baleen species in that they use a wide and varied catalog of social sounds. More than 36 different sound types (vocal sounds and surface-generated sounds from energetic surface behaviors) were found during a three year study on migrating humpback whales. During periods of high wind noise (where there were no audible boats or singing whales in the area), humpback whales modify both their acoustic repertoire as well as vocal signal properties. We found that humpback whale groups gradually switched from primarily vocal to primarily surface-generated communication in increasing wind speeds and background noise levels, but kept both signal types in their repertoire. We also found evidence of the Lombard effect, in that in increased wind-dominated background noise levels, humpback whale groups tended to increase the amplitude of their vocalizations. Determining how whales modify their vocal behavior in increasing levels of background noise will give us an important insight into how they might cope with increasing levels of anthropogenic noise.
Variation in the vocal behavior of southern right whales (Eubalaena australis) in coastal Brazilian waters
Authors: Parks Susan , Groch Karina , Flores Paulo A. C. , Sousa-Lima Renata S. , Urazghildiiev Ildar R.
Published: May 2013

Currently, there are three recognized species of right whales. The largest population is the southern right whale (Eubalaena australis), with circumpolar distribution in the southern hemisphere. One calving area for this population is in Brazilian waters, where increasing numbers of right whales have been sighted over the past decade along with an increase in anthropogenic activities including shipping traffic and fishing. The goals of this study were to describe the vocal behavior of southern right whales in Brazilian waters, assess the difference in vocalizations between areas with low and high human activity, and compare these results to studies conducted with North Atlantic right whales (Eubalaena glacialis) in the Western North Atlantic. Bottom-mounted archival acoustic recorders were deployed in October and November 2011 in two coastal locations in central Santa Catarina State, southern Brazil. One recorder was placed off Gamboa (27056'S and 48039'W, low traffic) and a second off Ribanceira (28011'S and 48037'W, high traffic). Automated detectors and noise statistic analysis tools developed for North Atlantic right whale upcalls were utilized to analyze the dataset. Calls produced by Brazilian right whales were significantly lower in fundamental frequency than North Atlantic right whale calls and the implications for these results will be discussed.
Complimentary ultrasound methods for the estimation of sound speed in macroalgae
Authors: Randall Jo , Hermand Jean-Pierre , Arnould Marie-Elise , Ross Jeff , Johnson Craig
Published: May 2013

Temperate kelp forests are among the most productive ecosystems in the world. However, there is mounting evidence that these habitats are in decline, both in range and productivity. Acoustic propagation modeling has been used to identify primary productivity in seagrass beds, and work is ongoing in development as a method of providing large scale measurements of productivity in macroalgae forests. Acoustic predictive models require knowledge of the material properties of interest, yet little is known about the acoustic properties of seaweed species. As a preliminary step towards acoustic modeling of seaweed systems, this study investigates the acoustic properties of Ecklonia radiata, a key species in temperate Australian marine systems. Measuring sound speed in macroalgae, as with other biological material, provides unique challenges due to their intrinsic morphological and anatomical characteristics. Using a range of frequencies between 2 and 10 MHz different methods are proposed to measure sound speed both directly and indirectly. The measurements show a consistent result, with variation according to tissue type. This research provides an important first step toward the development of acoustic propagation models in kelp forest ecosystems.
Modeling the acoustic scattering from large fish schools using the Bloch-Floquet theorem
Authors: Kulpe Jason A. , Leamy Michael J. , Sabra Karim G.
Published: May 2013

Scattering from large fish schools can significantly contribute to volume reverberation in the open ocean measured by mid-frequency long-range SONAR systems (1–10 kHz). This can potentially cause large false-alarms, especially if the resonance frequencies of the fish’ air-filled swim bladder is excited. Hence, to ultimately improve the detection performance of long-range SONAR systems, we seek an efficient modeling technique for the acoustic scattering created by large school of fish, which readily accounts for the fish bio-acoustic properties, school’s spatial configuration and multiple scattering effects. We exploit here a key observation to simplify our problem: fish in larger schools tend to swim in a periodic arrangement whereby we approximate the large school as an infinite system with a periodic collection of fish’ air filled swimbladders. Thus, the Bloch-Floquet theorem, governing waves in periodic media, allows predictions of the acoustic field in an infinite media by simply modeling the dynamic response of a single unit cell only (containing one fish). This approach allows one to rapidly predict the frequency dependent reflection and transmission coefficient on a semi-infinite fish school for various incident waves. Good agreement was found with the results obtained from finite element modeling of realistic, finite sized fish schools.