bluuebunny/crossref_metadata_2025 huggingface dataset:
High resolution radiation pattern measurements of a grand piano—The effect of attack velocity
Authors: Roginska Agnieszka , Mathew Justin , Anderson Jim , Case Alex U.
Published: May 2013

The sound radiation pattern of a grand piano is highly complex and depends on the shape of the soundboard, construction of the frame, reflections from the lid, and other parts of the instrument’s structure. The spectral energy generated by and emitted from the instrument is further complicated by the sound production mechanism (hammers, strings), the attack velocity, and results in independently complex behaviors depending on the register of the piano. This paper presents the acoustic measurements of the radiation pattern of a grand piano using a high spatial resolution measurement technique. Measurements of a Yamaha Diskclavier were taken using a 32-channel microphone array with a 2-in. spacing between capsules. The complex radiation patterns and overtone structure is analyzed for middle-C at three attack velocities—pianissimo, mezzo forte, and forte. Comparisons of the effect of attack strength on frequency response and radiation pattern are presented.
“Calibrating” the insertion depth of roll-down foam earplugs
Authors: Berger Elliott H.
Published: May 2013

Since introduction in 1972, roll-down slow-recovery foam earplugs have become nearly ubiquitous. They are used widely in industries and by consumers around the world. Their performance has been reported in numerous journal articles and they are often part of laboratory experiments, either as the object of the study or as a reference device that is used as a control or to assure exclusion of noise from the ear to facilitate data acquisition. As such it is important to be able to describe their performance since although they generally provide high levels of protection, the amount of protection and its spectral dependence is a function of insertion depth. Real-ear attenuation results will be presented for a range of insertions from that which caps the earcanal to full earcanal insertion past the second bend. The results will be compared to published data to demonstrate how they can be used to estimate the quality of fit that was likely achieved vs that which was reported. These data will be useful to researchers who wish to “calibrate” the quality of insertion they are achieving in their own studies going forward. [The author is an employee of 3M and the research was funded by 3M.]
Comparison of subjective and objective methods for the measurements of hearing protector devices attenuation and occlusion effect
Authors: Nélisse Hugues , Le Cocq Cécile , Boutin Jérôme , Voix Jérémie , Laville Frédéric
Published: May 2013

With the increase popularity of individual fit testing and miniaturization of electronic components, the field-microphone-in-real-ear approach (F-MIRE) is becoming more appealing and well suited for estimating hearing protection devices (HPD) attenuation both in laboratory and in “real world” occupational conditions. The approach utilizes two miniature microphones to simultaneously measure the sound pressure levels in the ear canal under the hearing protector, as well as outside of the protector. In this study, experiments on several human subjects were carried out in order to examine the various factors relating the subjective and objective attenuation values. The subjects were first instrumented on both ears with miniature microphones outside and underneath the protector. They were then asked to go through a series of subjective hearing threshold measurements followed by objective microphone recordings using high level diffuse field broadband noises. Earmuffs, earplugs, and double-protection were tested for each subject, and attenuation values were compared. Additionally, an objective scheme to measure the occlusion effect was developed and tested using subjects’ voice as the excitation and the same microphone setup. Results obtained for the attenuation values as well as the occlusion effect levels are presented and discussed.
First-principle simulation of the acoustic radiation force on microparticles in ultrasonic standing waves
Authors: Herring Jensen Mads Jakob , Bruus Henrik
Published: May 2013

The recent development in the field of microparticle acoustophoresis in microsystems has led to an increased need for more accurate theoretical predictions for the acoustic radiation force on a single microparticle in an ultrasonic standing wave. Increasingly detailed analytical solutions of this specific problem can be found in the literature [Settnes and Bruus, Phys. Rev. E 85, 016327 (2012), and references therein], but none have included the complete contribution from thermoviscous effects. Here, we solve this problem numerically by applying a finite-element method to solve directly the mass (continuity), momentum (Navier-Stokes), and energy conservation equations using perturbation theory to second order in the imposed time-harmonic ultrasound field. In a two-stage calculation, we first solve the first-order equations resolving the thermoviscous boundary layer surrounding the microparticle and with a perfectly matched layer as a non-reflecting boundary condition for the scattered waves. These first-order solutions are then used as source-terms for solving the time-averaged second-order equations [Muller et al., Lab Chip 12, 4617 (2012)] and in particular to determine the second-order time-averaged hydrodynamic stress on the particle surface. From this, we deduce the radiation force and compare it as a function of the physical parameters to existing analytical results.
Acoustic radiation force on a sphere without restriction to axisymmetric fields
Authors: Ilinskii Yurii A. , Zabolotskaya Evgenia A. , Hamilton Mark F.
Published: May 2013

The analysis presented at the previous ASA meeting related to investigation of the acoustic radiation force on a sphere embedded in a soft elastic medium with shear modulus that is several orders of magnitude smaller than its bulk modulus. The acoustic field was assumed to be axisymmetric and the spherical scatterer to be located on the axis of the acoustic beam. When one of these conditions is violated, the problem loses its symmetry. In this talk, the acoustic radiation force is considered in the more general case of nonaxisymmetric fields. The calculation is performed in Lagrangian coordinates. All acoustic fields, incident as well as scattered, depend on all three spherical coordinates. The incident and scattered waves, which include both potential and solenoidal parts, are expanded with respect to spherical harmonics. An analytical expression for the acoustic radiation force derived in this investigation may contain as many spherical harmonics as needed. In limiting cases when the scatterer is in liquid and only two modes, monopole and dipole, remain in the scattered fields, the solution for the acoustic radiation force recovers the results reported by Gor’kov [Sov. Phys. Doklady 6, 773 (1962)]. [Work supported by NIH DK070618 and EB011603.]
Three-dimensional analysis of the acoustic radiation pressure: Application to single-beam acoustical tweezers
Authors: Baresch Diego , Marchiano Régis , Thomas Jean-Louis
Published: May 2013

Recent studies on the acoustic radiation forces exerted by sound impinging spherical objects suggest the use of structured wavefronts for particle entrapment and controlled manipulation. In the scope of understanding why it is made possible to trap and manipulate small particles with sound, we present a general model for the acoustic radiation forces in three dimensions. A first generalization comes from the extension of well known results for the radiation pressure of plane waves to incident wavefields having arbitrary wavefronts. Second, the elastic spherical target of any dimension is allowed to be arbitrarily located within the wavefield. Introducing a new class of “single-beam” acoustical tweezers, we discuss the capabilities of different acoustical beams to achieve particle trapping and manipulation tasks. In addition, using an efficient experimental setup, we report the propagation of a peculiar beam carrying orbital angular momentum, namely an acoustical vortex, which is our selected candidate to achieve the first three-dimensional acoustic trap for elastic particles.
Perturbation analysis of flow about spherically pulsating bubble at the velocity node of a standing wave
Authors: AlHamli Mohammad , Rednikov Alexey Y. , Sadhal Satwindar S.
Published: May 2013

An analysis using the singular perturbation method for a radially pulsating gas bubble at the velocity node of a standing wave was conducted with ε = U0/(aω)≪1 as a small parameter and ωa2/ν≫1 as a large parameter. Here,a, U0, ω, and ν are length scale, velocity scale, frequency, and kinematic viscosity, respectively. While the mean oscillatory flow around the gas bubble has no net time-averaged flow component, viscous steady streaming arises due to the nonlinearity of the flow dynamics. However, with bubble surface being considered shear-free, the vorticity generation in the system is quite weak as compared with what would result from a solid boundary. Not surprisingly, the steady streaming is also weak. As already known, the steady streaming would not arise with purely radial pulsations of a bubble in an otherwise quiescent liquid. For the case of a non-pulsating bubble at the velocity node, streaming is seen at O(ε2). However, as seen with the case of a radially pulsating bubble at the velocity antinode, interaction of two oscillatory fields creates streaming at lower order. The phase difference between radial and lateral oscillations was found to play a significant role in both the streaming direction and intensity.
Acoustic bubble sorting of ultrasound contrast agents
Authors: Versluis Michel
Published: May 2013

Ultrasound contrast agents are coated microbubbles with radii ranging from 1 to 10 μm. Medical transducers typically operate at a single frequency; therefore, only a small selection of bubbles will contribute to the nonlinear contrast through resonance. Thus, the sensitivity of contrast-enhanced ultrasound can be improved by narrowing down the size distribution. Monodisperse bubble can be formed in a flow-focusing geometry. However, it requires extensive skills in microfluidics technology and in surface chemistry. Here, we present a simple lab-on-a-chip technique to sort microbubbles on-line in a traveling ultrasound wave. A broad range of the parameter space of bubble size and frequency has been characterized to provide physical input parameters for a simple force balance model. We find good agreement for the modeled displacement as a function of the bubble radius for a range of sizes in the unbounded fluid. Within the confinement of the sorting chip, we find good agreement for the resonance behavior and overall with a smaller displacement than predicted as a result of bubble–wall interactions.This novel sorting strategy may lead to an overall improvement of the sensitivity of contrast echo of at least one order of magnitude.
Connecting cues to signals in auditory attention
Authors: Hafter Ervin
Published: May 2013

Among his many fields of study, Bert Scharf played a major role in our understanding the role of auditory attention, especially at the level of basic psychophysics. Of special importance to this talk is the profound influence that he had on research in our lab (myself, Bert Schlauch, Joyce Tang, Kourosh Saberi, and Poppy Crum) through his work on signal uncertainty in masking and its alleviation by specific informational cues. Scharf’s resurrection of the probe-signal method led us to examine the effects of uncertainty on both the means and variances of effective bandwidths used by listeners in a detection task. Also shown will be cases where we used different kinds of cues to study detection at various levels of processing including judgments based on: specific spectral components, complex pitches derived from sets of harmonics and locations in frequency reliant on mentally tracking an FM stimulus through a period of occlusion.
Tuning in the time domain revealed through detection of auditory signals of unexpected duration or presentation time
Authors: Wright Beverly , Dai Huanping
Published: May 2013

Bert Scharf was interested in how expectation affects auditory performance. He explored this question using the probe-signal method of Greenberg and Larkin, in which the listener is led to expect a particular stimulus but is occasionally presented with an unexpected but equally detectable one. The detectability of unexpected stimuli provides insight into the listener’s template for the expected stimulus. Bert’s expectation research, which focused on the frequency domain, inspired us to extend the inquiry to the time domain. We have seen that signal detection can be quite poor for signals of unexpected duration as well as for signals presented at unexpected times, indicating that listeners attend selectively to these two temporal aspects of sound. However, this temporal tuning is much broader for starting time (hundreds of milliseconds) than for signal duration (can be <25 ms). Thus, it appears that listeners can select the template for signal detection with considerable accuracy, but do not apply the selected template strictly to the expected starting time of the signal. We are grateful to Bert for his mentorship and keen interest in this topic. [Work supported by NIH/NIDCD.]
Inspiration from Bertram Scharf’s work
Authors: Zeng Fan-Gang
Published: May 2013

In 1990, Bertram Scharf and I discussed about me doing a post doc in his laboratory. The opportunity to work with Bertram did not materialize, but his work in loudness, efferents, and attention has been a continuing inspiration not only for my own research but also for auditory perception, physiology, and audio engineering in general. For example, a Google Scholar search of “Bertram Scharf” on November 8, 2012, produced 1517 citations for his top 10 papers, with 6 on loudness and critical bands, 2 on efferents, and 2 on attention. Here I highlight two recent projects that have been inspired by Scharf’s work. The first project showed significant loudness adaptation in patients with auditory neuropathy, particularly those with otoferlin deficits. This result directly supports Scharf’s proposition that simple loudness adaptation is due to a sensory process, which in this case can be pinned down to transmitter release and replenishment in the hair cell and nerve synapse. The second project extended Scharf’s theoretical work in efferents and attention to improving feedback control in cochlear implant users and tinnitus sufferers. This line of work could improve cochlear implant speech perception in noise and reduce internal gain to alleviate tinnitus.
Tunneling effect on the sound transmission loss of a flat structure coupled with a porous material
Authors: Sgard Franck C. , Atalla Noureddine , Gholami Mohammad , Nelisse Hugues
Published: May 2013

It is well known that when measuring sound transmission loss (STL) in a laboratory, among all test conditions, the location of a specimen in an aperture affects the results, due to the tunneling effect. Previous studies have considered this effect for flat single panels and double walls but the case of a panel with attached sound package seems to have received very little attention. This paper deals with the application of a modal approach to study the STL of a rectangular plate coupled with a porous material located inside a tunnel. The sound absorbing material is supposed to be described by a transfer matrix calculated using a Transfer Matrix Method, which relates interstitial pressure and total normal stress on both sides of the material. The model is validated by comparison with finite element/boundary element computations. Numerical results are shown to illustrate the validity of the proposed hybrid modal-TMM methodology and its use to investigate the niche effect in the presence of a sound absorbing material.
A finite element solution strategy based on Padé approximants for fast multiple frequency sweeps of coupled elastic, poroelastic, and internal acoustic problems
Authors: Rumpler Romain , Göransson Peter
Published: May 2013

Analyses involving structural-acoustic finite element models including three-dimensional modeling of porous media are, in general, computationally costly. While being the most commonly used predictive tool in the context of noise and vibrations reduction, efficient solution strategies enabling the handling of large-size multiphysics industrial problems are still lacking, particularly in the context where multiple frequency response estimations are required, e.g., for topology optimization, multiple load cases analysis, etc. In this work, an original solution strategy is presented for the solution of multi-frequency structural-acoustic problems including poroelastic damping. Based on the use of Padé approximants, very accurate interpolations of multiple frequency sweeps are performed, allowing for substantial improvements in terms of computational resources, i.e., time and memory allocation. The method is validated and demonstrated for its potential on 3D applications involving coupled elastic, poroelastic, and internal acoustic domains.
Numerical simulation of acoustic waves in air and poroelastic media using the partition of unity finite element method
Authors: Chazot Jean-Daniel , Nennig Benoit , Perrey-Debain Emmanuel
Published: May 2013

Foams and fibrous materials are used in a large range of applications such as automotive or building acoustics. Their properties can be described with either a poroelastic model or an equivalent fluid model. These models, also used in geophysics, are now widely spread and are also available in some commercial finite element software applications. However, the discretization level required to achieve reasonable accuracy is not always acceptable in the mid-frequency range. In such case, the Partition of Unity Finite Element Method (PUFEM) using plane wave functions seems appropriate to avoid this limitation. The PUFEM has been recently applied to rigid frame materials with and without coupling with an acoustic cavity. It has also been applied efficiently to poroelastic materials. The present work focuses on the coupling between a poroelastic material, i.e., described with Biot's equations, and an air cavity. Some practical examples are tested to demonstrate the efficiency of the PUFEM for solving noise control problems at medium frequency, but also to underline the precautions that must be taken when dealing with an air-porous interface.
A method for measuring the acoustic properties of a porous sample mounted in a rigid ring in acoustic tubes
Authors: Dupont Thomas , Leclaire Philippe , Panneton Raymond , Verdière Kévin , Elkoun Saïd
Published: May 2013

This study presents a method to measure the acoustic properties of a homogeneous porous material with a support or a reduction element in an acoustic tube. Some materials tested have a lateral size much smaller than the tube’s diameter, as they cannot be produced in the correct dimensions without corrupting the material; this also permits the testing of the same samples in a large frequency bandwidth by using different section tubes. Moreover, the acoustic leaks on the material boundaries can significantly change the transmission loss measured in tubes. To rectify these problems, rings can be placed on each material surface. The presence of these rings can influence the acoustic indicator measurement; while this effect is negligible for tubes with a large cross section, it is not for tubes with a small cross section. To correct, or remove, the influence of the rings, we propose to use an application of the parallel assembly process of the transfer matrix method, which has recently been proposed by Panneton et al. [Proceeding Internoise, New York (2012)]. Measurements on classical porous materials with and without reductions are proposed and compared to simulated results. The ring’s effects and the proposed corrections are discussed for different materials.
Distinguishing between science and pseudoscience in forensic acoustics
Authors: Morrison Geoffrey Stewart
Published: May 2013

In this presentation, I argue that one should not attempt to directly assess whether a forensic analysis technique is scientifically acceptable. Rather one should first specify what one considers to be appropriate scientific principles governing acceptable practice, then consider any particular approach in light of those principles. I focus on one principle: The validity and reliability of an approach should be empirically tested under conditions reflecting those of the case under investigation using test data taken from the relevant population. Versions of this principle have been key elements in several reports on forensic science, including forensic voice comparison, published over the last four-and-a-half decades. I consider the aural-spectrographic approach to forensic voice comparison (also known as “voiceprint” or “voicegram” examination) in light of this principle, and also the currently widely practiced auditory-acoustic-phonetic approach (these two approaches do not appear to be mutually exclusive). Finally, I challenge the audience members to consider what each of them thinks constitutes the relevant scientific principles regarding acceptable practice, and then consider their own approach to forensic-acoustic analysis in light of those principles.
A Canadian perspective on forensic science versus pseudoscience
Authors: Ostrum Brent
Published: May 2013

This presentation will provide my personal observations regarding forensic science versus pseudoscience in the Canadian legal system. I am neither a lawyer nor a judge; rather, I am a forensic scientist with over 25 years of experience in the Canadian system. My presentation focuses on relevant criteria for expert evidence considered in Canadian courts. The key ruling in R. v. Mohan (1994) provides the start of the discussion with subsequent court rulings adding various elements. In Canada, we have had several judicial inquiries, such as the Kaufmann Commission, that can serve to guide experts. Select aspects of the 2009 NAS report “Strengthening Forensic Science in the United States: A Path Forward” will also be referenced. There are some common “criteria” often used by courts in different jurisdictions to assess expert evidence, including forensic acoustics. In other words, some basic expectations for all forms of expert evidence can be identified. I will attempt to show how select “sciences” have tried to fulfill those expectations. This will involve some commentary on issues of individual examiner competency, oversight at a system level (e.g., accreditation), and the need for proper and adequate method validation.
Assessing acoustic features in the speech of asylum seekers
Authors: Rosenhouse Judith K.
Published: May 2013

One of the areas of forensic linguistics concerns asylum seekers who speak languages which are foreign to the official language of the country where they apply for asylum. Identifying and verifying their real national background may be difficult if their speech manner reveals non-typical properties of their (real or alleged) native languages. Governments submit such asylum seekers’ speech samples for linguistic analysis on various levels, including phonetic acoustics. This aspect of forensic linguistics raises questions about the scientific merit of such an analysis. Our aim is to examine some of the questions which relate to segmental and supra-segmental features that are analyzed acoustically based on recorded samples of asylum seekers’ (alleged) native language and compared with the same features as known from the literature. We demonstrate such issues by examples from the speech of Arabic-speaking asylum seekers whose native tongue is (supposed to be) some local dialect but the recording includes various foreign features reflecting different dialects or languages. These questions involve sociolinguistic factors that affect individual speakers’ speech production due to a complex and unstable life-history. We suggest that the acoustic methods currently used in speech analysis in this context could be considered pseudo-science in many cases.
Speaker identification in reverberant environments
Authors: Korany Noha
Published: May 2013

The performance of speaker identification process degrades in reverberant environments, as reverberation leads to clear physical effects on the perceived signals. This paper investigates the effect of room reverberation on the identification rate. However, various reverberant environments are simulated, and the impulse response is convolved with dry speech signals. The reverberant speech database is used by the identification engine within the train and the test phases. Then, statistical identification technique using the Gaussian Mixture Model (GMM) is implemented. Three types of features, Mel-Frequency Cepstrum Coefficients (MFCC), Perceptual Linear Predictive Cepstrum Coefficients (PLPCC), and Relative Spectral Perceptual Linear Predictive Cepstrum Coefficients (RASTA-PLPCC) are extracted. Various types of features are integrated and used for the classification problem. Finally, the performance of the recognition process is evaluated while varying the duration of the train and the test signals, the features used for the classification problem, and the room reverberation. A series of physical measures that correlate with various attributes of the sound perceived in rooms, such as the reverberation time T60, the clarity index C80, the definition D, are calculated. Then, their effect on the identification rate is investigated.
Influence of amplification scheme and number of channels on aided speech-intelligibility performance
Authors: Amlani Amyn M. , Bharadwaj Sneha V. , Jivani Shirin J.
Published: May 2013

Modern hearing aids offer a wide range of channels (i.e., filters) and amplification schemes. Our previous work revealed that increasing the number of channels, in conjunction with a fast-fast amplification scheme, results in (a) the spectral flattening of the vowels /i, u, ^/ (Amlani et al., 2011), and (b) reduced consonant- and vowel-identification accuracy in impaired listeners (Amlani et al., 2012). In the present study, we assess the performance of impaired listeners and their normal-hearing controls on the perception of everyday speech using the Connected Speech Test (Cox et al., 1987, 1988). The stimuli were processed through a simulated hearing aid with varying amplification schemes (linear, compression [fast-fast, slow-slow, fast-slow]) and number of channels (2, 8, 16). Findings revealed that while speech-intelligibility performance improved markedly with everyday speech compared to /CVC/ words for both groups, normal-hearing listeners identified the target words significantly better than impaired listeners did. Speech-intelligibility performance was similar across number of channels for normal-hearing listeners, but decreased significantly with a fast-fast amplification scheme. For impaired listeners, performance declined for channels greater than 2 and with the inclusion of the fast-fast amplification scheme. We discuss the implication of these findings relative to clinical application and hearing aid design.
Relationship between subjective and objective evaluation of noise-reduced speech with various widths of temporal windows
Authors: Mizumachi Mitsunori
Published: May 2013

It is necessary to enhance adverse speech signals for building useful speech interfaces. Speech enhancement is essential under noisy and reverberant acoustic environments. Therefore, quality assessment of the enhanced speech signals should be also an important issue in noise reduction and dereverberation. Subjective evaluation is given by carrying out listening tests, and objective evaluation is provided by speech distortion measures. However, there is the discrepancy between subjective and objective evaluation of speech distortion. The author has investigated the relationship between subjective and objective evaluation of noise-reduced speech signals. The objective speech distortion was calculated in each short-term frame, of which length was fixed, and the statistical characteristics of the short-term speech distortion were investigated using higher-order statistics such as skewness and kurtosis. The preliminary result suggested that skewness of the short-term speech distortion could give an explanation for the discrepancy between subjective and objective evaluation. Further investigation of the relationship between subjective and objective evaluation of noise-reduced speech signals is carried out with a variety of temporal window widths. [Work supported by NEDO, Japan.]
Performance estimation of speech recognition based on Perceptual Evaluation of Speech Quality and acoustic parameters under noisy and reverberant environments with Corpus and Environment for Noisy Speech RECognition 4
Authors: Fukumori Takahiro , Nakayama Masato , Nishiura Takanobu , Yamashita Yoichi
Published: May 2013

CENSREC-4 evaluation framework has been distributed for evaluating distant-talking speech under various noisy and reverberant environments. It however has not been evaluated how variable noisy and reverberant features in this contains. We thus try to evaluate CENSREC-4 with our designed noisy and reverberant criteria based on PESQ and acoustic parameters. We specifically focus on criteria to represent the difficulty of noisy and reverberant speech recognition, and also confirm why it is difficult to easily evaluate the recognition accuracy in a part of CENSREC-4 corpus with our proposed noisy and reverberant criteria. We first designed the noisy and reverberant criteria using the relationship among the D value, the PESQ, and the ASR performance. We then tried to estimate the recognition accuracy in various noisy and reverberant environments with CENSREC-4. We carried out evaluation experiments to confirm the difficulty to easily evaluate the recognition accuracy in a part of CENSREC-4 courpus0. As a result of evaluation experiments, we confirmed that it was difficult to estimate the accuracy of noisy and reverberant speech recognition in heavy noisy and reverberant environment with CENSREC-4. We therefore confirmed that CENSREC-4 contained very challenging and variable noisy and reverberant data.
On instantaneous vocal tract length estimation from formant frequencies
Authors: Lammert Adam , Narayanan Shrikanth
Published: May 2013

The length of the vocal tract and its relationship with formant frequencies is examined at fine temporal scales with the goal of providing accurate estimates of vocal tract length from acoustics on a spectrum-by-spectrum basis despite unknown articulatory information. Accurate vocal tract length estimation is motivated by applications to speaker normalization and biometrics. Analyses presented are both theoretical and empirical. Various theoretical models are used to predict the behavior of vocal tract resonances in the presence of different vocal tract lengths and constrictions. Real-time MRI with synchronized audio is also utilized for detailed measurements of vocal tract length and formant frequencies during running speech, facilitating the examination of short-time changes in vocal tract length and corresponding changes in formant frequencies, both within and across speakers. Previously proposed methods for estimating vocal tract length are placed within a coherent framework, and their effectiveness is evaluated and compared. A data-driven method for VTL estimation emerges as a natural extension of this framework, which is then developed and shown to empirically outperform previous methods on both synthetic and real speech data. A theoretical justification for the effectiveness of this new method is also explained. [Work supported by NIH.]
Evaluation of system configuration to check the suitability for the sound field rendering using the inverse approach
Authors: Ih Jeong-Guon , Cho Wan-Ho , Hong Seung-Wan
Published: May 2013

Sound field control by the inverse approach based on the acoustical holography is useful to render an arbitrary target sound field within a selected control zone if the target condition is given in a detailed format. This method needs information on the various factors constituting the total system: source array configuration, relative position of source and control region, assigning method of target field condition, conditioning method, etc. Because these factors heavily affect the accuracy of the generated sound field, a proper definition of the problem including all factors related to the system configuration is important. In this work, we have studied on the condition of major factors of various configurations to generate the target sound field efficiently with high accuracy. Because the difference between target and generated sound field strongly depends on the noise and the information error existing in the actual situation, the expected accuracy should be calculated in relation to the characteristics of system transfer matrix. To this end, variances of uncorrelated noise, condition number, and linear independency of the transfer matrix are evaluated to check the suitability of transfer matrix for accurately rendering the sound field in both free-field and enclosed space.
Sound-field reconstruction performance of a mixed-order Ambisonics microphone array
Authors: Marschall Márton , Chang Jiho
Published: May 2013

Recently, there has been increasing interest in using spherical microphone arrays for spatial audio recordings. Accurate recordings are important for a range of applications, from virtual sound environments for hearing research through to the evaluation of communication devices, such as hearing instruments and mobile phones. Previously, a mixed-order Ambisonics (MOA) approach was proposed to improve the horizontal spatial resolution of spherical arrays. This was achieved by increasing the number of microphones near the horizontal plane while keeping the total number of transducers fixed. The approach is motivated by the fact that human spatial hearing is most acute in the horizontal plane. This study presents simulations of the performance of an MOA rigid-sphere microphone array, and its robustness to variations in microphone characteristics. Specifications of a commercially available microphone were used to simulate self-noise, sensitivity, and phase response variations between the microphones. To quantify the reconstruction error and the “sweet area” as a function of source elevation, the reconstructed sound field based on a simulated array measurement was compared to the reference sound field for both horizontal and elevated sources. It is expected that the MOA approach results in a larger sweet area for mid to high frequencies for horizontal sources.
Design and implementation of a personal audio system in a car cabin
Authors: Cheer Jordan , Elliott Stephen J.
Published: May 2013

The generation of personal listening zones in a car cabin would allow the different occupants to listen to different audio programs without the use of headphones. This would allow, for example, the driver to listen to a navigation system while the rear passengers watched a film. Personal audio systems have previously been implemented in mobile devices and monitors, for example, however, the investigation of the effects of an enclosure on the generation of personal listening zones has been limited. This paper presents an investigation of the effects of a car cabin sized enclosure on the generation of independent listening zones in the front and rear seats. The standard car audio loudspeaker array is used to produce independent listening zones at low frequencies, while a second array of small loudspeakers positioned at the four headrest positions is used to provide control over the rest of the audio bandwidth. The performance of the arrays is first simulated in a car cabin sized rectangular enclosure to understand the physical limits on the achievable control. The proposed arrays are then implemented in a real car to validate the simulation results and the results of a real-time implementation are presented.
Seafloor measurements using synthetic aperture sonar
Authors: Lyons Anthony , Brown Daniel , Olson Derek , Johnson Shawn
Published: May 2013

The past decade has seen considerable growth in the use of synthetic aperture sonar (SAS) imaging systems in both the civilian and military domains. Although SAS systems are almost always uncalibrated, they can still yield information about the seafloor given an understanding of the mechanisms affecting the statistical properties of the images produced by these systems. This talk will describe our recent efforts to link SAS image statistics to seafloor properties through the use of seafloor scattering models. Sample results from several SAS systems encompassing frequencies ranging from 6 to 300 kHz will be shown.
Seabed roughness parameters for the Malta Plateau from joint backscatter and reflection inversion
Authors: Steininger Gavin , Holland Charles W. , Dosso Stan E. , Dettmer Jan
Published: May 2013

This paper presents seabed interface-scattering and geoacoustic parameters estimated on the Malta Plateau, Mediterranean Sea, by joint Bayesian inversion of monostatic backscatter and spherical-wave reflection-coefficient data. The data are modeled assuming a stack of homogeneous fluid sediment layers overlying an elastic basement. The scattering model also assumes a randomly rough water-sediment interface with a von Karman roughness power spectrum. Scattering and reflection data are inverted simultaneously using a population of interacting Markov-chains to sample roughness and geoacoustic parameters as well as residual error parameters. Trans-dimensional sampling is applied to treat the unknown number of sediment layers and unknown autoregressive order of the errors (to represent residual correlation). Results are considered in terms of marginal posterior probability profiles and distributions, which quantify the effective data information content to resolve scattering/geoacoustic parameters and structure. Results indicate well-defined scattering (roughness) parameters in good agreement with existing measurements, and a multi-layer sediment profile over a high-speed (elastic) basement, consistent with independent knowledge of sand layers over limestone. [Work supported by ONR.]
An initial model-data comparison of reverberation and clutter from a near-shore site in the Gulf of Mexico
Authors: Ellis Dale D. , Preston John R.
Published: May 2013

Reverberation measurements were made in the Gulf of Mexico off Panama City, Florida, USA, in April 2012 in preparation for the main Transmission Reverberation Experiment (TREX) in May 2013. The data were gathered using the triplet section of the ONR Five Octave Research Array (FORA), deployed as a fixed receiver. By steering cardioid beams to the right or left, the array can reduce ambiguity. Beamformed data from the 2012 trial show background noise with high directionality and variability due to nearby shipping. Model predictions of reverberation and target are compared with data using a range-dependent Clutter Model, which uses adiabatic normal modes as the computational engine. The initial predictions use isovelocity water, over a sandy bottom halfspace with Lambert scattering, and bathymetry from the GEBCO08 database. These initial results will be presented, hopefully supplemented by improved predictions with better environmental inputs and additional clutter data obtained during the May 2013 experiment. [Work supported by ONR Code 322 OA.]
Energy evolution in enclosure geometries as exhibited by a finite difference time domain method
Authors: Belanger Zackery
Published: May 2013

Measurements and simulations conducted for the purpose of extracting or constructing impulse responses are inextricably dependent on a limited number of receiver locations. Wave-based simulations offer the opportunity to discard this dependency and assess an entire evolving sound field. In this work, a finite difference time domain method is implemented to simulate an evolving sound field in a range of enclosure geometries with reflective boundaries. The distribution of energy is monitored statistically as mixing and ergodic states are approached, and evidence is presented for the predictability of this evolution based on room geometry alone. A system for importing geometries from the Rhinoceros 3D CAD software, using the Grasshopper parametric environment, is also presented. A form known as the Barnett Billiard Table in mathematics is included in this study, which exhibits exceptional behavior even without the presence of diffusive treatment.
Perceptual evaluation of violin vibrations and audio-tactile interaction
Authors: Altinsoy M. Ercan , Merchel Sebastian , Tilsch Sebastian
Published: May 2013

When playing a violin, the musician communicates with his instrument not only through his ears but also his fingers, chin, shoulder, and eyes. He uses different sensory inputs, which are provided by different sensory channels, such as auditory, tactile, kinesthetic, and visual, to play his musical instrument. The perceived vibrations are useful for the player to feel and to control the instrument. The interaction between sound and vibration plays also a role on the overall instrument perception. In this study, violin vibrations and their interaction with violin sounds were evaluated. Therefore, the vibration amplitudes of the neck and the violin sounds were recorded simultaneously during normal playing. The vibration recordings were analyzed, and then additional stimuli were generated by filtering or modifying frequency components. In the first experimental session, the vibration stimuli, which were presented to the subjects via a mini electrodynamic shaker, were evaluated. In the second experimental session, an investigation with multimodal (auditory-haptic) stimuli was conducted. The results show the importance of vibrations on the overall perception of the instrument and provide information on useful vibration features for the player-instrument interaction.
Recent experiences with vibration of stage and audience floors in concert halls
Authors: Wulfrank Thomas , Lyon-Caen Igor , Jurkiewicz Yann , Brulez Johan , Kahle Eckhard
Published: May 2013

Vibration of (wood) surfaces plays a significant role in concert hall acoustics, as confirmed by musicians and music lovers. Many acoustic engineers, on the other hand, tend to have strong reservations against vibrating surfaces, and usually try to minimize surface vibration in order to maximize RT and airborne strength (G) at bass frequencies. This has led to a generally accepted preference for massive and stiff surface constructions in new halls. Problems have been known to occur when this general guideline was also applied to the design of wooden floors, in particular stage floors. Despite some good scientific research in this field, a big gap still remains between the vibro-acoustic behavior of wooden floors and subjective preferences of musicians and audiences. This paper further explores the role of vibrations in concert hall design, and the need for balancing surface reflectivity versus vibration transmission. Recent experiences, including the new Konserthus in Stavanger and the renovation of the Bolshoi Hall of the Moscow Conservatory, will be described as well as vibration measurements carried out on a number of existing stage floors. Some implications for the design of wooden floor constructions will be discussed.
Auditory reaction time measurements and equal-latency curves in the California sea lion (Zalophus californianus) and bottlenose dolphin (Tursiops truncatus)
Authors: Mulsow Jason , Finneran James J.
Published: May 2013

Subjective equal-loudness contours are used to create weighting functions for human noise-mitigation criteria. Comparable direct measurements of subjective loudness with animal subjects are, however, difficult to conduct. Using methods similar to those used in previous mammalian studies, this study estimated subjective loudness through the measurement of response time (RT) in an auditory signal-detection task. Measurements were conducted in a sound-attenuating hut with a California sea lion and under water in a quiet pool with a bottlenose dolphin. Tonal stimuli were presented at supra- and near-threshold sound pressure levels (SPLs) using a method of constants. Median RT increased with decreasing SPL for both species across all tested frequencies. A Piéron function, which models RT as a function of SPL, was fitted to the RT-SPL curves in a nonlinear fashion. Equal-latency curves were based on the Piéron functions at each frequency. Preliminary results for the sea lion suggest that the equal-latency curves are similar to the audiogram at longer median RTs (~300 ms), with increasing deviation from the audiogram at the fastest median RTs (~200 ms). Continued testing with additional subjects will provide further data for designing marine mammal auditory weighting functions. [Funded by U.S. Navy Living Marine Resources Program.]
Prey pursuit strategy of Japanese horseshoe bats, Rhinolophus ferrumequinum Nippon, during target selection task
Authors: Kinoshita Yuki , Ogata Daiki , Aihara Ikkyu , Watanabe Yoshiaki , Riquimaroux Hiroshi , Ohta Tetsuo , Hiryu Shizuko
Published: May 2013

We investigated prey pursuit behavior of Japanese horseshoe bats, while they were tasked to make a choice between two tethered fluttering moths during flight. Echolocation pulses were recorded by a telemetry microphone mounted on the bat, combined with a 17-ch horizontal microphone array to measure pulse directions. Flight paths of the bat and moths were monitored by using two high-speed video cameras. Acoustical measurements of CF echoes from fluttering moths (67 kHz : CF2 frequency) was conducted using an ultrasonic loudspeaker, turning the head direction of the moth to the loudspeaker from 0° to 180° in the horizontal plane. Amount of acoustical glints caused by moth fluttering varied with the sound direction, showing the maximum between 70° and 100°. In the flight experiment, moths chosen by the bat fluttered within or moved across these angles to the bat’s pulse direction, which would cause dynamic changes in frequency and amplitude of acoustical glints during flight. This result suggests that dynamic changes in acoustical glints appear to attract the bats for prey selection. Furthermore, mathematical modeling implied that the bats possibly took the optimum flight path for capturing a target, which the bat selected based on the acoustical cues in the echoes.
Effects of the source location on numerical biosonar beampattern predictions for bat noseleaves
Authors: Gupta Anupam Kumar , Müller Rolf
Published: May 2013

Approximately 300 bat species are known to emit their ultrasonic biosonar pulses through the nostrils. In these animals, ultrasound is produced by the larynx, propagates along the vocal tract, exits through the nostrils, and is finally diffracted by intricately shaped baffle structures known as “noseleaves.” Noseleaf geometry determines diffraction and hence the spatial distribution of the emitted ultrasound. As a consequence, numerical predictions of the noseleaves' acoustic function can be made based on the digital models of noseleaf shape. To limit model size and computational effort associated with numerical beampattern predictions, the vocal tract is often only partially included in these models or left out completely. In order to investigate the effect of source placement within a complete or partial vocal tract attached to a noseleaf shape on the numerical beampattern prediction, the noseleaf of the Great Roundleaf Bat (Hipposideros armiger) was studied as a model. Numerical beampattern estimates were obtained for a single monopole source positioned near the vocal folds or closer to the nostrils. Two monopoles sources placed in each nostril were also investigated. It was found that source positioning could impact the beampattern whenever they broke the symmetry in the near-fields of the two nostrils.