bluuebunny/crossref_metadata_2025 huggingface dataset:
Effect of musical training on static and dynamic measures of spectral-pattern discrimination
Authors: Sheft Stanley , Smayda Kirsten , Shafiro Valeriy , Maddox W. Todd , Chandrasekaran Bharath
Published: May 2013

Both behavioral and physiological studies have demonstrated enhanced processing of speech in challenging listening environments attributable to musical training. The relationship, however, of this benefit to auditory abilities as assessed by psychoacoustic measures remains unclear. Using tasks previously shown to relate to speech-in-noise perception, the present study evaluated discrimination ability for static and dynamic spectral patterns by 49 listeners grouped as either musicians or nonmusicians. The two static conditions measured the ability to detect a change in the phase of a logarithmic sinusoidal spectral ripple of wideband noise with ripple densities of 1.5 and 3.0 cycles per octave chosen to emphasize either timbre or pitch distinctions, respectively. The dynamic conditions assessed temporal-pattern discrimination of 1-kHz pure tones frequency modulated by different 5-Hz lowpass noise samples with thresholds estimated in terms of either stimulus duration or signal-to-noise ratio. Musicians performed significantly better than nonmusicians on all four tasks. Discriminant analysis showed that group membership was correctly predicted for 84% of the listeners with the structure coefficient of each measure greater than 0.46. Results suggest that enhanced processing of static and dynamic spectral patterns defined by low-rate modulation may contribute to the relationship between musical training and speech-in-noise perception. [Work supported by NIH.]
Toward a model of informational masking: The Simpson–Fitter metric of target-masker separation
Authors: Gilbertson Lynn , Chang An-Chieh , Stamas Jacob , Heo Inseok , Lutfi Robert
Published: May 2013

Informational masking (IM) is the term used to describe masking that appears to have its origin at some central level of the auditory nervous system beyond the cochlea. Supporting a central origin are the two major factors associated with IM: trial-by-trial uncertainty regarding the masker and perceived similarity of target and masker. Here preliminary evidence is provided suggesting these factors exert their influence through a single critical determinant of IM, the stochastic separation of target and masker given by Simpson–Fitter’s da [Lutfi et al., J. Acoust. Soc. Am. 132, EL109-113 (2012)]. Target and maskers were alternating sequences of tones or words with frequencies, F0s for words, selected at random on each presentation. The listener’s task was to discriminate a frequency-difference in the target tones or identify the target words. Performance in both tasks was found to be constant across conditions in which the mean difference (similarity), variance (uncertainty), or covariance (similarity) of target and masker frequencies were selected to yield the same value of da. The results are discussed in terms of their implications for the development of a model of IM that emphasizes the statistical properties of signals over loosely defined concepts of masker uncertainty and target-masker similarity.
Retention of gap length in normal-hearing listeners
Authors: Smith Meghan M. , Ries Dennis , Woods Audra
Published: May 2013

Listeners’ ability to retain information about gap length within a noise burst was studied. JND for gap length between the target and comparison stimuli was obtained using the single-interval adjustment-matrix procedure for retention intervals that were silent, included four noise bursts, or included four noise bursts with discrete gaps. JNDs for retention intervals containing either type of noise burst did not differ significantly from those obtained for the silent retention interval. This result differs from that found for retention of pitch and loudness. This might occur as more cortical resources are used for retention of auditory temporal information.
Dependency of tonality perception on frequency, bandwidth, and duration
Authors: Taghipour Armin , Edler Bernd , Amirpour Masoumeh , Herre Jürgen
Published: May 2013

Psychoacoustic studies show that a narrowband noise masker exhibits a stronger simultaneous masking effect than a tonal masker with the same signal power placed at the noise center frequency. Consequently, perceptual audio codecs commonly incorporate some sort of tonality estimation as part of their perceptual model. However, common tonality estimation techniques do not necessarily reflect the perception of tonality by human listeners. As long as the tone and narrowband noise signals are long enough, they are easily distinguishable for normal hearing listeners. However, if the stimulus duration decreases, both signal types approach the shape of impulses, and therefore, at some point become audibly identical. Consequently, at a given frequency and noise bandwidth, there is a duration threshold below which the signals cannot be distinguished. A series of so-called “2-AFC 3-step up-down” psychoacoustic tests are designed and carried out to investigate the frequency and bandwidth dependency of these duration thresholds. The test results, collected from 32 listeners, are statistically evaluated and confirm a decreasing threshold for increasing center frequency and bandwidth. These results can be used to improve psychoacoustic models for audio codecs by using tonality estimators with frequency and bandwidth adapted temporal resolution.
Reflection orders and auditory distance
Authors: Mendonça Catarina , Lamas João , Barker Tom , Campos Guilherme , Dias Paulo , Pulkki Ville , Silva C. , Santos Jorge
Published: May 2013

The perception of sound distance has been sparsely studied so far. It is assumed to depend on familiar loudness, reverberation, sound spectrum, and parallax, but most of these factors have never been carefully addressed. Reverberation has been mostly analyzed in terms of ratio between direct and indirect sound, and total duration. Here we were interested in assessing the impact of each reflection order on distance localization. We compared sound source discrimination at an intermediate and at a distant location with direct sound only, one, two, three, and four reflection orders in a 2AFC task. At the intermediate distances, normalized psychophysical curves reveal no differentiation between direct sound and up to three reflection orders, but sounds with four reflection orders have significantly lower thresholds. For the distant sources, sounds with four reflection orders yielded the best discrimination slopes, but there was also a clear benefit for sounds with three reflection orders. We discuss the results in terms of direct-to-reflected ratio, reflection directionality, and spectral information.
Multi-scale acoustics of partially open cell poroelastic foams
Authors: Hoang Minh Tan , Bonnet Guy , Perrot Camille
Published: May 2013

The present paper reports on the modeling of linear elastic properties of acoustically insulating foams with unit cells containing solid films or membranes at the junction between interconnected pores from a numerical homogenization technique. It combines fluid-flow induced microstructure identification with simulations of the effective Young’s modulus and Poisson ratio from a mixture of routinely available laboratory measurements (porosity, permeability, cell size) and finite element calculations when the boundary conditions of the periodic unit cell take particular symmetric forms. This combination results in microstructural determination of the macroscopic coefficients entering into the Biot-Allard theory of wave propagation and dissipation through porous media. Precise control over pore morphology and mechanical properties of the base material renders this multi-scale approach particularly suitable for various advanced applications.
Full-band exact homogenization of one-dimensional elastic metamaterials
Authors: Yang Min , Yang Zhiyu , Sheng Ping
Published: May 2013

Metamaterials extend the realm of materials’ properties by carefully designed structural inclusions. By targeting the extraction of effective properties from composite materials, homogenization theory plays an important role for metamaterials in their design and characterization. However, conventional homogenization methods are limited to the long wavelength limit. Here, we introduce an exact homogenization scheme valid for one-dimensional metamaterials over the full frequency band. In this scheme, with the aid of eigenstates’ characterization, a set of explicit formulas for effective mass density and effective elastic modulus are obtained by matching the surface responses properties of a metamaterial’s single structural unit with a piece of effectively homogenized material. In the frequency regimes beyond the conventional homogenization theory, new features, such as the imaginary parts of the effective parameters, have been found. Applying this scheme on a layered structure, the predicted transport properties and displacement fields from the effective parameters show excellent agreement with numerical simulations.
Structural-borne sound mitigation in small wind turbines using constrained viscoelastic layer
Authors: Pletner Baruch , Strum Nic , Sampson David , Kheirabadi Ali
Published: May 2013

As the growing acceptance of small wind turbines operating in suburban and rural communities coincides with increasingly stringent regulations on the sound emitted by these turbines, the need for sound mitigation solutions becomes urgent. Small turbines need to be affordable for small business use, and thus, proposed solutions must be cost-effective and low maintenance. Easy retrofit to existing turbines is also desirable. Wind turbines generate sound via two main mechanisms: structural borne sound generated by the gearbox and generator and transmitted through the nacelle structure and aeroacoustic sound generated by the interaction of the airstream with the rotating blades and other turbine components. Current study focused on the mitigation of structural-borne sound in a 50 kW wind turbine using a constrained viscoelastic layer. The viscoelastic layer comprised of multiple tiles with normal force to the nacelle structure provided by ratcheting bands. Optimal value for the normal force was empirically determined, and the resulting reductions in generated sound were documented both in the laboratory and on a working turbine under a number of operating conditions. The result is a cost-effective solution with zero cost of ownership and easy installation on a wide range of small to medium-size wind turbines.
Modeling multi-level factors using linear mixed effects
Authors: Clopper Cynthia G.
Published: May 2013

Linear mixed-effects models of 2x2 designs are readily interpretable using treatment contrast coding, although their interpretation is not directly comparable to the interpretation of more traditional ANOVAs. In particular, the interpretation of the “main effect” term for one factor holds only for the baseline level of the other factor. Interpreting models of designs involving factors with more than two levels and/or interactions involving more than two factors is more complex and even less comparable to familiar interpretations of ANOVAs. Alternative methods for analyzing these more complex designs include using different contrast coding (e.g., sum, Helmert, or custom), selecting specific baseline levels for the factors, and running multiple models of the same data set with different baseline levels of comparison. These methods may return quite different results, however, such that a significant factor with treatment contrast coding may not be significant with sum contrast coding or a significant interaction term in one model may not be significant when the baseline levels of the relevant factors are changed. Thus, although linear mixed effects provide an opportunity to model complex designs with multiple sources of variability, this modeling requires careful consideration of model parameters to achieve the most appropriate interpretation of the data.
A likelihood ratio-based forensic voice comparison using formant trajectories of Thai diphthongs
Authors: Pingjai Supawan
Published: May 2013

This study investigates the phonetic–acoustic properties of the three Thai diphthongs /i:a, ī:a, u:a/ within the context of forensic voice comparison. The likelihood-ratio approach is applied to the parameterized formant trajectories of each diphthong in order to evaluate their respective discriminatory power. The aim of this study is to assess to what extent such properties can be used to distinguish, in a probabilistic sense, two or more speech samples. Formant trajectories were fitted using both polynomial interpolation and the discrete cosine transform. Likelihood ratio values were derived by the multivariate kernel density (MVKD) estimation approach proposed by Aitken and Lucy (2004) and then calibrated by using the log-likelihood ratio cost function—Cllr (Brummer 2005) and the 95%-credible interval (Morrison et al., 2010). We have finished gathering all speech data for this study, and are currently processing the data using various computational tools. References: Aitken, C. G. G. and Lucy, D. “Evaluation of trace evidence in the form of multivariate data,” App. Stat. 54,109–122 (2004). Brümmer, N., FoCal Toolkit [software], 2005. See: http://www.dsp.sun.ac.za/~nbrummer/focal/. Morrison G. S., Thiruvaran T., and Epps J. (2010) “Estimating the likelihood-ratio output of a forensic-voice-comparison system,” The Speaker and Language Recognition Workshop, Brno.
Regularized reconstruction of sound fields with a spherical microphone array
Authors: Granados Alba , Jacobsen Finn
Published: May 2013

Spherical near field acoustic holography with microphones mounted on a rigid spherical surface is used to reconstruct the incident sound field. However, reconstruction outside the sphere is an ill-posed inverse problem, and since this is very sensitive to the measurement noise, straightforward implementation might lead to disastrous reconstructions. A large number of regularization tools based on singular value decomposition are available, and it has been found that the acoustic holography problem for certain geometries can be formulated in such a way that similarities to singular value decomposition become apparent. Hence, a number of regularization methods, including truncated singular value decomposition, standard Tikhonov, constrained Tikhonov, iterative Tikhonov, Landweber and Rutishauser, have been adapted for spherical near field acoustic holography. The accuracy of the methods is examined by means of simulations and measurements, which leads to practical recommendations on the use of regularization techniques regarding space and frequency.
Optimized two-dimensional imaging of transient sound fields using a hybrid transient acoustic holography
Authors: Pan Siwei , Jiang Weikang
Published: May 2013

A hybrid transient acoustic holography (HTAH) is presented to visualize the two-dimensional transient sound fields radiated from planar sources with unknown locations and dimensions, by combining the time reversal source localization with the near-field acoustic holography (NAH) based on the interpolated time domain equivalent source method (TDESM). Based on the near-field measurements with a microphone array, the time reversal source focusing algorithm is used to find out the hotspots of the sound sources on the equivalent source plane, which suggests the collocation of equivalent sources. The interpolated TDESM based NAH is then employed to reconstruct and image the transient sound field on the reconstruction plane. The proposed HTAH technique can reduce the elements number of microphone array by only collocating the equivalent sources in the vicinity of the “real” sound sources. The visualization of the transient sound fields radiated from single-planar-piston and dual-planar-piston model is studied by numerical simulations, respectively. The experiments are performed in a semi-anechoic chamber by using two baffled loudspeakers. Both the simulation and experimental results revealed that this hybrid scheme can realize a better two-dimensional imaging of transient sound fields than the original interpolated TDESM based NAH in the measurement using same amount of microphones.
Classifying sonar signals with varying signal-to-noise ratio and bandwidth
Authors: Murphy Stefan
Published: May 2013

An automatic aural classifier developed at Defence Research and Development Canada has demonstrated the ability to distinguish target echoes from clutter using perceptual-based features inspired by sonar operators. Initially, the classifier was tested with echoes from explosive sources, but more recent research involved transmitting broadband waveforms from transducer sources. In sonar transducer operation, there is a trade off between source level and bandwidth, and the goal of this paper is to study how these factors affect echo classification. Source level relates to signal-to-noise ratio (SNR), which inherently affects classification since signals with low enough SNR cannot be distinguished from noise, let alone other signals. The dependence of classification performance on bandwidth is less obvious; however, the aural classification technique is based on a sub-band type of processing that mimics the basilar membrane in the human auditory system, and this model is not well adapted for narrow bands. Performance of the aural classifier is therefore expected to degrade as bandwidth is decreased. In this paper, the effect of SNR and signal bandwidth on echo classification is examined using echoes of varying SNR, and in various bands selected using band-pass filters.
B-format for binaural listening of higher order Ambisonics
Authors: Nishimura Ryouichi , Sonoda Kotaro
Published: May 2013

B-format is a four-channel signal capable of rendering a sound scene with spatial information. It can be regarded equivalent to first order ambisonics. Ambisonics requires a high order to contain precise spatial information, and higher order ambisonics requires an exponentially large amount of data. This limitation comes from the fact that the original aim of ambisonics is to reproduce the whole sound field. However, as mobile devices are prevalent, users often listen to sound media through earphones. Because nowadays users can hold sound contents individually, one can assume that sound contents could be produced adaptively to each user. Here we propose a way to make B-format signals more suitable for individual binaural listening. We assume that the production side can capture a sound scene with higher order ambisonics, because it may be processed for enterprise applications. Under this assumption, the binaural signal is once generated from the higher order ambisonics, and then its B-format signal is obtained by inversely processing the signal, assuming the first order ambisonics. Computer simulations show that interaural phase differences (IPDs) are improved at a frequency region where IPD dominantly affects sound localization. Results of hearing tests are also discussed.
Steering for listening area of reflective audio spot with parametric loudspeaker array
Authors: Masunaga Shohei , Ikefuji Daisuke , Nakayama Masato , Nishiura Takanobu
Published: May 2013

A parametric loudspeaker has a high directivity by utilizing an ultrasound wave as a carrier wave. Therefore, a parametric loudspeaker can form a specific listening spot called “audio spot.” Furthermore, a parametric loudspeaker can form the “reflective audio spot” by utilizing the reflected sound. The listeners in the listening area may perceive the acoustic sound image on the reflector. However, it has the problem that the reflective audio spot with a single parametric loudspeaker is narrow area. Therefore, it is difficult for several listeners to perceive the reflective audio spot at the same time. If we can expand the area of the reflective audio spot, several listeners can perceive the reflective audio spot at the same time. Thus. in this paper, we attempt to steer the area of the reflective audio spot with parametric loudspeaker array. We carried out objective and subjective evaluation experiments to confirm the effectiveness of the proposed system in a conference room. As a result, we confirmed the proposed system can expand the are of the reflective audio spot.
Fast Bayesian hierarchical inference via sparsity enforcing a priori for aeroacoustic source imaging
Authors: Chu Ning , Djafari Ali , Picheral José , Gac Nicolas
Published: May 2013

Aeroacoustic imaging is a technique for mapping the positions and powers of aeroacoustic sources. We propose a novel inverse solution by applying Bayesian hierarchical inference via sparsity enforcing a priori. We model the sparse prior of source powers by using the double exponential distribution, which can greatly improve the spatial resolution and robustness to background noise. Hyperparameters and source powers can be alternatively estimated based on the joint maximum a priori optimization. To accelerate the optimization, we improve the forward model of aeroacoustic power propagation by exploring the convolution operator. Finally, our approach is compared with some classical methods on simulated and real data. And our approach is feasible to apply for aeroacoustic imaging with the 2D non-uniform microphone array in wind tunnel tests, especially for near-field monopole and extended source imaging.
Security screening using ultrasound
Authors: Hutchins David , Davis Lee , Tsen Sheldon
Published: May 2013

This work will demonstrate that it is possible to produce images of hidden objects, using ultrasound transmitted through air. For example, it can be shown that a knife can be imaged, when hidden behind a layer of clothing fabric. To achieve this, it is necessary to use coded waveforms and signal recovery techniques, in order to retrieve small signals in the presence of a much larger reflection from the outer fabric surface. In addition, ultrasound can be used in through-transmission to detect hidden objects within thin packages. This and other examples of the use of air-coupled ultrasound for security work will be demonstrated.
A study on acoustic imaging based on beamformer to range spectra in the phase interference method
Authors: Miyake Ryota , Hayashida Kohei , Nakayama Masato , Nishiura Takanobu
Published: May 2013

Information on the distance to the target is very important to achieve the practical use of hands-free speech interfaces and nursing-care robots. Many distance measurement methods, which use the time-of-flight (TOF) of a reflected wave measured with reference to the transmitted wave, have been proposed. However, these methods cannot measure short distances because the transmitted wave, which has not attenuated sufficiently at the time of a reflected wave reception, suppresses reflected waves for short distances. We previously proposed an acoustic distance measurement method based on interference between the transmitted and reflected waves, which can be used for distance measurement over a short range using single microphone. This method is referred to the phase interference method. It can estimate the distance to target, but cannot estimate the direction of target. In the present paper, therefore, we propose to achieve acoustic imaging with the phase interference method by using microphone-array instead of single microphone. More specifically, we apply the beamformer to the range spectra calculated from observed signals at each microphone of microphone-array to obtain the spatial information. Finally, we confirm the effectiveness of the proposed method through evaluation experiments in real environments.
A detection of danger sounds based on variable-state hidden Markov models
Authors: Okamoto Asako , Hayashida Kohei , Nakayama Masato , Nishiura Takanobu
Published: May 2013

To detect hazardous situations with danger sounds, the acoustic surveillance system is an ideal candidate. The conventional systems recognize environmental sounds with hidden Markov model (HMM) in order to detect danger sounds. It is however difficult to accurately recognize the environmental sounds, because the optimum HMM parameters for environmental sounds have not been identified. It is important factor for accurately recognizing them to ideally determine the number of states, one of the HMM parameter. On the other hand, environmental sounds which include danger sounds have a wider characteristic as the structure, the complexity, the length, etc. The variable states should be therefore an optimum HMM structure to detect the danger sounds. We thus propose the danger sound detection based on variable-state HMMs corresponding to a number of inflection points with the delta power of environmental sounds. We first investigate the recognition performance of environmental sounds including danger sounds with various states of HMM. We then investigate the relationship between the recognition performance and a number of inflection points with the delta power of various environmental sounds. As a result of evaluation experiments, we designed an optimum variable-state HMM for environmental sounds and confirmed the effectiveness of the proposed method.
Chain architecture: An efficient hardware solution for a large microphone array system
Authors: Zotkin Dmitry N. , Adelman Ross , O'Donovan Adam E. , Duraiswami Ramani
Published: May 2013

A typical microphone array system consists of a number of microphones connected to the digitization hardware and central processing unit in a parallel fashion. Such radial, hub-and-spoke architecture has multiple points of failure, suffers from electromagnetic interference, and does not scale well. In this paper, an alternative, chain-like architecture is described. In such setup, the microphones in a system are organized in a single chain. Each individual microphone board has an ADC chip and is connected to the previous and to the next microphones in the chain with short multi-wire cables carrying digital signals. A buffer board at the end of the chain converts the digital data stream into the industry-standard USB 2.0 format. In this way, the individual microphone board becomes the building block for quick and easy arbitrary-configuration microphone array assembly with minimal amount of wiring involved. A hardware implementation of the chain architecture was developed and is described. Accompanying drivers and software allow the user to perform on-the-fly data acquisition and processing in C and in MATLAB. As an example, a 64-microphone array was built, and several source localization and beamforming algorithms were implemented in MATLAB. Experimental results using the data gathered from the array are presented.
Resolution analysis of the inverse problem for geoacoustic experiment design
Authors: Ganse Andrew A.
Published: May 2013

This work explores effects of experiment geometry and array configuration on the resolving power of a continuous ocean bottom geoacoustic inverse problem in a shallow water environment. The uncertainty and resolution of this problem, in which ocean bottom P-wave velocities as a function of depth are estimated from noisy acoustic pressure waveforms received on vertical and horizontal line arrays in the water, can be investigated before the experiment is conducted, allowing one to improve or optimize the problem parameters to best configure an experiment during its planning phase. In this work, the resolution results of complete synthetic geoacoustic inversions at varying geometries and array configurations are compared with resolution results at various candidate seabottom profiles, initially using standard techniques of linearized inverse theory. Singular value decomposition is used to interpret the tie between geometry and regularization in the inverse problem, which directly affects the resolution. Then, additional comparisons and analysis address the nonlinearity of the problem, which causes a dependence of the resolution results on the bottom profile being solved for—which is unknown, and Monte Carlo analysis is used to show where the linearity approximation breaks down in the resolution results. [Work partially funded by ONR.]
Rough interface acoustic scattering from layered sediments using finite elements
Authors: Isakson Marcia J. , Chotiros Nicholas
Published: May 2013

Quantifying acoustic scattering from rough interfaces is important for reverberation modeling, acoustic sediment characterization, and propagation modeling. Most models of interface scattering on layered surfaces rely on approximations to the Helmholtz/Kirchhoff integral. These models generally make such assumptions as neglecting the local angle for reflection and disregarding multiple scattering between rough layers. In this study, a mixed boundary element/finite element model is used to calculate rough interface scattering from sediment layers including elastic solids. The finite element method, based on the Helmholtz equation, is exact within the limits of the discretization density; reflections are calculated locally and all orders of scattering among layers are included. Using this model, bottom loss and backscattering predictions will be calculated for several cases. These predictions will be compared with more traditional scattering models and finally to a full finite element model of a point source reflection from the same sediment bottom. [Work supported by ONR, Ocean Acoustics.]
Active acoustics and sound reinforcement at TUI Operettenhaus, Hamburg: A case study
Authors: Schwenke Roger W.
Published: May 2013

TUI Operettenhaus is a proscenium theater with one balcony, which is host to drama, musical theater, and concerts. The venue hosts different sound reinforcement systems for different shows, and now has a permanent active acoustic system. The physical acoustics are very dry as is appropriate for modern theater with spatial sound reinforcement, and the active acoustic system allows the reverberation time to be extended as appropriate for different performances. The active acoustic system can also pass through signals to its speakers for spatial surround reproduction. The installation of the active acoustic system in an older building posed many challenges. This case study presents the challenges that were overcome during installation, the integration of the active acoustic system with sound reproduction, and the measured performance of the system.
The raw and the cooked in architectural acoustics
Authors: Martens William L. , Reinhardt Dagmar
Published: May 2013

Whereas the “raw” experience of live sound events is often quite far removed from the “cooked” auditory imagery that is presented when live acoustical events are amplified by a sound reinforcement system, there are many audio signal processing tools that can be applied in the attempt to simulate the more natural auditory characteristics of live (unplugged?) musical performances. This paper builds a discussion of perceptual results of modern sound reinforcement technology based upon the Lévi-Strauss notion regarding what modern culture does to the “raw” to make it “cooked”). A key concept in evaluating the quality of a sound reinforcement system is that of the standard of reference against which the perceptual results can be compared. As there is no shared opinion nor well-established optimal acoustical character for a space upon which some consensus could be built, the question presents itself again and again. This paper will address related issues of reference, preference, and adequacy in sound reinforcement.
E-Venue—Affordable electronic acoustic enhancement for small venues
Authors: Barbar Steve
Published: May 2013

The advent of modern digital signal processing made altering the acoustical conditions in a venue using electro-acoustic tools practical. Such systems have been in constant use for many years in concert halls, opera houses performance spaces, houses of worship, and a variety of other spaces and applications throughout the world. However, the cost associated with specialized nature of these systems has put them out of the reach of many small venues that stand to benefit most from use of this technology. This paper describes a new low cost, integrated, electro-acoustic system designed specifically for use in small venues—including but not limited to; performance venues, recital halls, rehearsal spaces, houses of worship, etc.
Interaction between critical listening environment acoustics and listener reverberation preference
Authors: Leonard Brett , King Richard , Sikora Grzegorz
Published: May 2013

Reverberation is a central effect in many modern music productions. In the case of classical music, it may even be the only effect used. There is, however, minimal literature concerning the interaction between reverberation preference and the listening environment used during critical mixing tasks. In order to explore this critical interaction, a group of highly trained subjects are tasked with adding reverberation to dry, premixed stereo program material in two different acoustic environments: a recording studio control room and a highly reflective room. The control room is representative of most studios, with an RT of approximately 200 ms. The reflective environment more closely approximates an untreated residential room, with an RT of over 350 ms, with a marked increase in lateral energy. Somewhat predictably, the mean preferred reverberation level is higher in a less reverberant environment, but the distributions of reverberation level preference are shown to be narrower for the more reflective mixing environment. The time it takes for subjects to reach a decision is similar in both environments, but the reflective environment shows a longer period of adaptation at the beginning of each trial set.
Structures of resonators in a cavity for improving a sound insulation of a thin double-leaf panel
Authors: Nakanishi Shinsuke
Published: May 2013

The specific acoustic problem of a double-leaf panel is a less sound insulation caused by a mass-air-mass resonance. For improving the sound insulation, many studies have suggested Helmholtz resonators in the cavity, which are tuned at the resonant frequency. They have measured and analyzed this problem of double-walls spaced with 100 mm thickness of air gap. They have suggested that the resonators improve the sound insulation to the resonant transmission, and discussed its optimization for a gain by the resonators and structures set in the cavity. But it is unclear that those results can apply to sound insulation by a double grassing with 5 mm thickness of air gap, which is often seen even as a thermal insulated window, and whose air gap is quite thinner than that of the walls. Then, this study measured effects of various resonators in the cavity for improving the sound insulation of thin double-leaf panels, and discusses effects of structures and perforation ratio to the sound insulation. Moreover, for analyzing the effects of resonators, this study discusses measured results with theoretical studies of sound absorption models for resonators.
Bat species identification from zero crossing and full spectrum echolocation calls using Hidden Markov Models, Fisher scores, unsupervised clustering and balanced winnow pairwise classifiers
Authors: Agranat Ian
Published: May 2013

A new classification technique for the identification of bats to species from their echolocation calls is presented. Three different datasets are compiled and split in half for training and testing classifiers. Combined, the data include 9014 files (bat passes) with 226,432 candidate calls (pulses or extraneous noise) representing 22 different species of bats found in North America and the United Kingdom. Some files are of high quality consisting of hand-selected search phase calls of tagged free flying bats while others are from a variety of field conditions including both active (attended) and passive (unattended) recordings made with a variety of zero crossing and full spectrum recording equipment from multiple vendors. Average correct classification rates for the three datasets on test data are 100.0, 97.9, and 88.8%, respectively, with an average of 92.5, 72.2, and 39.9% of all files identified to species. Most importantly, classifiers in the third dataset for two species of U.S. endangered bats, Myotis sodalis (MYSO) and Myotis grisescens (MYGR) have a correct classification rate of 100 and 98.6%, respectively, and identify 67.4% and 93.8% of all files to species suggesting that the classifiers are well suited to the accurate detection of these endangered bats.
Robustness of perceptual features used for automatic aural classification to propagation effects
Authors: Binder Carolyn M. , Hines Paul C. , Pecknold Sean P. , Scrutton Jeff
Published: May 2013

Previous effort has shown that a prototype aural classifier developed at Defence R&D Canada can be used to reduce false alarm rates and successfully discriminate cetacean vocalizations from several species. The aural classifier achieves accurate results by using perceptual signal features that model the features employed by the human auditory system. Current work focuses on determining the robustness of the perceptual features to propagation effects for two of the cetacean species studied previously—bowhead and humpback whales. To this end, classification results are compared for the original vocalizations to classification results obtained after the vocalizations were re-transmitted underwater over ranges of 2 to 10 km. Additional insight into the propagation effects is gained from transmission of synthetic bowhead and humpback vocalizations, designed to have features similar to the most important aural features for classification of bowhead and humpback vocalizations. Each perceptual feature is examined individually to determine its robustness to propagation effects compared to the other aural features. To gain further understanding of propagation effects on the features, preliminary propagation modeling results are presented in addition to experimental data.
Complexity index and proportional variability to study dolphin whistles
Authors: Bazúa Durán Carmen , Sarmiento Ponce E. Julieta , González Leal Brenda P. , Rodríguez Bohorquez Camila
Published: May 2013

Dolphin whistles are emitted especially during social interactions and feeding activities involving group cohesion, individual recognition, and recruitment. This paper presents a new methodology to describe and compare the whistle repertoire of dolphins. It consists on first extracting the whistle contour using Matlab BELUGA, then classifying whistles into whistle types using Matlab ArtWARP, next classifying whistle types into four general categories (high complexity, low complexity, linear long, and linear short), and finally computing a complexity index and a proportional variability of the whistle repertoire. The method was tested with whistles from captive and wild bottlenose dolphins, Tursiops truncatus, and from wild Guyana dolphins, Sotalia guianensis. Results obtained showed that this very simple method is useful to describe the whistle repertoire and to compare it according to the general behavioral state of dolphins, and between species. It is necessary to implement new methodologies like this one to better understand how dolphins are using whistles, since acoustic communication is the most important sense in dolphin species. This is specially important in areas where dolphins are exposed to humans, and where underwater visibility is limited, like Laguna de Términos, a Marine Protected Area in Mexico. [Work supported by PAPIIT-UNAM.]
Uncertainty of transmission loss due to small scale fluctuations of sound speed in two environments
Authors: Fabre Josette P. , Wood Warren
Published: May 2013

Seismic oceanography techniques reveal detection of small scale variations in sound speed not detectable via conventional oceanographic means, i.e., frequent XBT or CTD casts). Due to computational and practical limitations, such small scale spatial and temporal detail that exists in a real ocean environment is not typically included in acoustic ocean models. However, such measurements can provide insight to the small scale variability (uncertainty) that exists in the ocean but is not predicted by mesoscale ocean models. We show acoustic predictions made with the Range Dependent Acoustic Model (RAM) using measured seismic oceanography and CTD data at two locations in significantly different environments. Additionally, the CTD measurements are smoothed to a resolution comparable to that provided by a dynamic ocean model and acoustic predictions are computed. The Uncertainty Band (UBAND) algorithm (UBAND) [Zingarelli,”A mode-based technique for estimating uncertainty in range-averaged transmission loss results from underwater acoustic calculations,” J. Acoust. Soc. Am. 124(4) (2008)] is applied to the smoothed oceanographic data using estimates of sound speed uncertainty calculated from the high resolution measurements. We find reasonable estimates of uncertainty due to the small scale oceanography that is not characterized by mesoscale ocean models.
Seismic oceanography imaging of thermal intrusions in strong frontal regions
Authors: Book Jeffrey W. , Wood Warren T. , Rice Ana E. , Carniel Sandro , Hobbs Richard , Ansorge Isabelle , Fischer Tim , Prandke Hartmut
Published: May 2013

The Naval Research Laboratory and collaborating partners carried out two dedicated seismic oceanography field experiments in two very different strong frontal regions. Adriaseismic took seismic oceanography measurements at the confluence of North Adriatic Dense Water advected along the Western Adriatic Current and Modified Levantine Intermediate Water advected around the topographic rim of the Southern Adriatic basin. ARC12 took seismic oceanography measurements in and around the Agulhas Return Current as it curved northward past the Agulhas Plateau and interacted with a large anticyclone that collided with the current. Despite one study focused on coastal boundary currents and the other focused on a major Western Boundary Current extension, the complex horizontal structures seen through seismic imaging are tied to the processes of thermal intrusions and interleaving in both systems. Seismic Oceanography provides a unique capability of tracking the fine-scale horizontal extent of these intrusions. In both systems they occur primarily along isopycnals and are largely density compensating. The formation of these structures is associated with advective processes rather than diffusive processes, despite gradients favorable for double diffusion mixing. Results from these studies also show that submesoscale eddies are playing an important role in the formation of thermal intrusions near these strong fronts.
Mapping non-local turbulent breakdown of oceanic lee waves offshore Costa Rica through seismic oceanography
Authors: Fortin Will F. , Holbrook W. Steven , Schmitt Ray , Book Jeff , Smith Scott
Published: May 2013

Data analysis techniques in Seismic Oceanography are rapidly becoming more complex. Beyond first-order observation of oceanic structures, it is possible to extract quantifiable information about internal wave energies and turbulent dissipation rates. We use two co-located seismic surveys taken one day apart to estimate turbulent diffusivities of lee wave breakdown with emphasis on the mid-water turbulence generated. Through a horizontal wavenumber spectral analysis of our seismic images, we estimate turbulent dissipation rates throughout the cross-section. By integrating horizontal seismic slope spectra in the wavenumber domain over the turbulent subrange, we obtain relative turbulent energies across the survey. To resolve absolute turbulent diffusivities, we scale the relative measures to known absolute energies from tracked seismic reflectors (isopycnals). The analysis section spans 22 km laterally and full ocean depth with a resolution on turbulent diffusivity of 10 m vertically by 400 m laterally. We focus on the region of elevated turbulent diffusivity caused by the breakdown of the lee wave above its generating site. We find the turbulent diffusivities related to the lee wave breakdown to be about five times greater than surrounding waters and 15 times greater than average open ocean diffusivities. We also see increased turbulent diffusivity around the rough bathymetry.
A method for desalination and water remediation by hydrodynamic cavitation
Authors: Crum Larry , Skinner Michael , Zeilinger Scott
Published: May 2013

Water is becoming an increasingly valuable commodity, with population growth demanding more and more amounts of this limited resource. Increased efforts are directed toward recycling and remediation, as well as desalination of the large quantities of seawater available. Dr. Bertwin Langenecker was a pioneer in utilizing hydrodynamic cavitation in a variety of applications that would remove dissolved solids from water and other liquids. His combination of intense cavitation using a rotor-stator combination, as well as simultaneously adding an adsorbent, demonstrated impressive results in desalination and waste water remediation. In this presentation, a description will be given of Dr. Langenecker’s technology as well as a sampling of some of his most impressive results. Speculations as to why this approach works as well as it does will be presented.
Inertial cavitation at the nanoscale
Authors: Kwan James J. , Graham Susan , Coussios Constantin
Published: May 2013

Our group has recently developed novel nano-sized drug carriers that spatially target a tumor and release their payload in the presence of ultrasound-induced inertial cavitation. To maximize drug release and distribution within the tumor, co-localization of the drug carrier and cavitation nuclei is necessary. We have recently demonstrated that rough-patterned silica nanoparticles can reduce inertial cavitation thresholds to clinically relevant levels, and will extravasate in tumors alongside the liposomes by virtue of their size. We now report on the underlying mechanisms that these nanoparticles, which are orders of magnitude smaller than the acoustic wavelength, can instigate inertial cavitation. The rough surface of the nanoparticle is modelled as a plane with a crevasse that traps a nanobubble. Using this model, we predict the motion of a gas bubble as it emerges from the cavity in response to the compressional and rarefactional ultrasonic pressures. We show that cavitation occurs when the nanobubble breaks free from the surface, growing unstably before collapsing during the compressional half cycle of the acoustic wave. Calculations show that a nanoscaled cavity greatly reduces the cavitation threshold across all frequencies and geometries studied. In addition, cavitation thresholds nonlinearly decrease with increasing cavity size.
Ultrasonic atomization: A mechanism of tissue fractionation
Authors: Simon Julianna C. , Sapozhnikov Oleg A. , Khokhlova Vera A. , Wang Yak-Nam , Crum Lawrence A. , Bailey Michael R.
Published: May 2013

High intensity focused ultrasound (HIFU) can be used to atomize liquid by creating a fountain on the surface exposed to air. The mechanism of atomization can be most accurately described by the cavitation-wave hypothesis wherein a combination of capillary waves excited on the liquid surface with cavitation beneath the surface produces a fine spray. Here, we show experimentally that a free tissue surface can also be atomized resulting in erosion of tissue from the surface. A 2-MHz spherically focused transducer operating at linearly predicted in situ intensities up to 14,000 W/cm2 was focused at ex vivo bovine liver and in vivo porcine liver tissue surfaces without the capsule. The end result for both in vivo and ex vivo tissues was erosion from the surface. In bovine liver at the maximum intensity, the erosion volume reached 25.7 ± 10.9 mm3 using 300 10-ms pulses repeated at 1 Hz. Jet velocities for all tissues tested here were on the order of 10 m/s. Besides providing a mechanism for how HIFU can mechanically disrupt tissue, atomization may also explain how tissue is fractionated in boiling histotripsy. [Work supported by NIH EB007643, NIH DK43881, and NSBRI through NASA NCC 9-58.]