bluuebunny/crossref_metadata_2025 huggingface dataset:
Auditory and Visual Sensations: Yoichi Ando’s theory of architectural acoustics.
Authors: Cariani Peter A
Published: October 2010

Yoichi Ando is a well-known architectural acoustician who employed genetic algorithms, acoustics, and psychophysical models of listener percepts and preferences to optimize the design of the Kirishima International Concert Hall in Japan. Ando’s recent book [Auditory and Visual Sensations, guest editor P. Cariani (Springer, New York, 2009)] summarizes decades of psychophysical experiments and neurophysiological observations (ABR, SVR, EEG, MEG). This paper will outline Ando’s psychophysics-based approach to architectural acoustics and his correlation-based theory of hearing and vision, along with supporting psychophysical and neurophysiological experimental observations. Ando proposes a temporally coded, correlation-based model of neuronal signal processing in which features of an internal autocorrelation representation subserve “temporal sensations” (pitch, timbre, loudness, duration), while features of an internal interaural cross correlation representation subserve “spatial sensations” (sound location, size, diffuseness related to envelopment). Together these two representations account for the basic auditory qualities most relevant for listening to music and speech in indoor performance spaces. Remarkably, Ando and colleagues have found many analogs of auditory percepts and preferences in vision. These include perception of the missing fundamental of flickering light as well as preferences for flickering lights, oscillatory movements, and texture regularity. Ando’s theory suggests possible common temporal processing mechanisms for hearing and vision.
Prediction of reverberant speech intelligibility against multiple noise interferers in rooms: Binaural useful-to-detrimental ratios.
Authors: Lavandier Mathieu , Culling John F. , Jelfs Sam
Published: October 2010

Monaural measures of temporal smearing of speech can accurately predict intelligibility in rooms in quiet, but in noise they neglect the listener’s ability to separate target speech from interfering sound sources using the binaural system. Lavandier and Culling (2010) proposed a binaural model which combines effects of better-ear listening and binaural unmasking to accurately predict the effects of spatial cues on segregation in a reverberant environment. A computationally efficient version of this model is presented, which has been validated in realistic conditions, including head-shadow, multiple interferers, and real-room acoustics. It can generate intelligibility maps from room designs, forming the basis of powerful architectural tools and providing potential guides to treatment strategies for the hearing impaired. Following Bradley et al. (1999), this model is extended to also account for the smearing effect of reverberation on target speech by computing the binaural useful-to-detrimental ratio. The useful component contains the early part of the target binaural room impulse response (BRIR), whereas the detrimental part contains its late part and the BRIRs of the interferers.
Deconstructing the spatial-temporal variation of soundscapes in natural areas.
Authors: Lam Kin-Che , Brown A. Lex , Marafa L. M.
Published: October 2010

The spatial and temporal variation of the acoustic environment across a natural area is complex. Attempting to measure and describe the spatial and temporal variation of the soundscape is resource intensive and time consuming. This paper uses a small coastal area and settlement in the countryside of Hong Kong to examine how the a-priori identification of landscape units can provide efficiencies in the design of sampling schemes for the measurement of soundscapes. The sources of natural sounds that contribute to the soundscape, and the spatial-temporal-intensity patterns of sound from these sources, are shaped, to a large extent, by the nature of each landscape unit and its surrounding units. We suggest that knowledge of the landscape units, and their attendant sound sources, can help decomposition of the complex soundscape pattern of a natural area into its contributing components, which makes the task of designing a spatial and temporal sampling program for their measurement more tractable. Particular landscape units exhibit distinctive, and partially predictable diurnal and seasonal patterns of variation. We demonstrate the utility of a landscape unit approach in assessing visitor perception of the soundscape in the area and intrusion of noise into the natural soundscape.
Locating a small change in a multiple scattering environment (LOCADIF): Application to monitoring concrete.
Authors: Planes Thomas , Rosseto Vincent , Larose Eric , Margerin Ludovic
Published: October 2010

Most conventional imaging techniques are operated in single scattering media where the Born approximation is found relevant. In the multiple scattering regime where this approximation does not hold, the wave forgets its initial direction after a few scattering events and we cannot have access to the precise trajectory of the wave. All waves are mixed together in phase, amplitude, wavevector, etc., which defines the diffuse regime. This paper presents an imaging technique to locate a weak perturbation in a multiple scattering environment. We derive a formula to predict the spatio-temporal decorrelation of diffuse coda waves induced by an extra-scatterer. Locating this new defect is then formulated as an inverse problem which is solved by a maximum likelihood approach. Using elastic waves in the 100–350 kHz frequency band, we show that the position of a millimetric hole drilled in a heterogeneous concrete sample can be retrieved with a precision of a few centimeters. We illustrate the robustness of the method and define an optimal time of propagation that depends on the geometry of the acquisition and the diffusion constant of the medium. Applications to monitoring civil engineering structure are discussed.
Time coherence of acoustic arrivals during the PhilSea09 experiment.
Authors: Dzieciuch Matthew A. , Worcester Peter F. , Colosi John A.
Published: October 2010

During the spring of 2009, 275-Hz signals were transmitted over a distance of nearly 200 km from an axial source to a vertical line array with 30 elements. The source transmitted a 100-Hz bandwidth signal every 5 min over a period of several days. This time interval is fast enough to capture the slightly slower changes in the ocean acoustic sound channel which are due to internal waves. The internal waves cause a loss of coherence over time, frequency, and space. The unique sampling of this experiment allows us to examine all three domains at once. The time coherence has not been captured in this detail in any previous experiment. Loss of coherence in these domains cause the received signal to have variations in time of arrival, phase, and intensity. These signal variations are measurable over both geometric and non-geometric arrivals.
Matched field localization and environmental inversion from deep-water vertical line array measurements.
Authors: Heaney Kevin D. , Campbell Richard C. , Baggeroer Arthur B. , D’Spain Gerald L. , Worcester Peter , Dzieciuch Matthew A.
Published: October 2010

Distributed vertical line array (DVLA) measurements of a controlled towed source were made as part of the Office of Naval Research Philippine Sea 2009 experiment (PhilSea09). The vertical line array consisted of two 1000 m sections, one spanning the sound channel axis (∼1000 m) and the other spanning the surface conjugate depth (∼5000 m). The source transmitted a combination of narrowband tones and LFM signals ranging between 20 and 450 Hz was towed at ranges from 3 to over 100 km from the DVLA. In this paper, matched field localization methods, developed in the 1990s are applied to the pair of DVLA receivers. A comparison of the detection and localization skill of each array is made as a function of source range and source receiver. Deep-water propagation for shallow sources is dominated by convergence zone propagation and, therefore, the axial and deep receivers will be illuminated by the source at different times. Narrowband and broadband geo-acoustic inversion results will be presented. Seafloor roughness is shown to be a significant feature of the bottom bounce paths and a beam-based matched field processing technique is applied to mitigate for the spreading of bottom interacting ray-paths.
Acoustic-archeological analysis of the Peruvian sound vessels.
Authors: Llimpe Q. Celso E. , Moreno R. Jorge N. , Nuñez B. Roxana
Published: October 2010

The sound vessels have been discovered in several cultures from countries such as Ecuador, Colombia, Mexico, and Peru. This research studied 15 double pots acoustic vessels of the Chimu (1100–1462 d.C.), Lambayeque (700–1100 d.C.), Viru (200 a.C.–200 d.C.), and Chancay cultures (900 d.C.–1100 d. C.). The study consists of two parts: the archaeological survey and acoustic analysis. The archaeological study focuses on the iconographic analysis of the vessels to explain the possible meanings and uses of this instrument. Sound analysis describes its operation using a simple electro-mechanical acoustic model approximation to understand the sound behavior of the vessels.
Hot topics in animal bioacoustics.
Authors: Mellinger David K.
Published: October 2010

Here we examine advances in studying the myriad ways that animals use sound, and we use sound to study them. Right whales are sometimes injured or killed in ship collisions, but real-time passive acoustic tracking systems now allow ships to avoid areas with right whales. Ships also affect right whales by interfering with their acoustic communication; new modeling techniques help us understand the potential impact of this noise. Elephant communication can also be disrupted by noise, even noise in the ground, as they communicate using land-borne vibration heard via their feet. Stranded whales and dolphins often cannot hear vibrations at all, as a large fraction are functionally deaf—perhaps explaining their stranding. Tadpoles, on the other hand, have been discovered to hear sound, as they apparently orient and perhaps navigate with it. Humpback whales use sounds that may be a form of echolocation—the first such finding in a baleen whale. A male topi antelope will use sound to his advantage: when a female in his harem looks likely to stray, he makes sounds warning her that a predator is near—deceptively scaring her back into his group. (Comparisons to specific humans will be avoided.)
Attenuation performance of double hearing protection devices.
Authors: Gallagher Hilary L. , McKinley Richard L. , Bjorn Valerie S.
Published: October 2010

High performance jet aircraft are capable of producing 130–150 dB of noise. Large populations work long hours in these high level noise environments and are at risk of hearing loss and related disabilities. To protect aircrew and ground crew members from the nearly constant assault of hazardous noise, an array of hearing protection devices are issued. The American National Standard Institute (ANSI) Method for Measuring Real-Ear Attenuation at Threshold (REAT) of Hearing Protectors S12.6 was conducted on single hearing protection and double hearing protection (DHP). The term DHP can be slightly deceiving. DHP means that two devices are worn in combination, not the sum of the attenuation from each device worn singly. Passive attenuation data were compiled using the ANSI Methods of Estimating Effective A-Weighted Sound Pressure Levels When Hearing Protectors are Worn ANSI/ASA S12.68-2007. The EPAs noise reduction rating, the octave band method, the noise level reduction statistics for use with A-weighting, and the noise level reduction statistic, graphical for inserts (earplugs) and helmets worn singly and in combination will be presented and discussed.
Earmuff comfort.
Authors: Gerges Samir N. Y.
Published: October 2010

Hearing protectors must be efficient both in terms of noise attenuation and in comfortability of wearing. The purpose of this paper is to review publications related to earmuff comfort, most of which are based on the measurement of the total headband force and subjective evaluation using questionnaires. Most of the published results show a weak correlation between total headband force and subjective evaluation. This paper presents new quantitative indices based on comfort parameters, mainly measurements of the contact pressure distribution between the earmuff cushions or headphones and the circumaural flesh of the human head. The comfort parameters were investigated and equations were developed to calculate comfort indices. The calculated indices are correlated with subjective evaluations. Measurement results show good correlation with subjective evaluation.
A free-field method to calibrate bone conduction devices.
Authors: Pollard Kimberly A. , Tran Phuong K. , Letowski Tomasz R.
Published: October 2010

Bone conduction can be valuable for communication applications and can transmit speech reliably whether or not hearing protection is worn. Audiometric standards for bone conduction vibrators focus primarily on a single model of bone vibrator placed at certain locations on the skull. However, a variety of other models and skull locations may be used for communication, so more general calibration methods are needed. Here we developed a method to calibrate bone vibrators, analogous to free-field methods of air conduction earphone calibration. Participants were seated in an anechoic chamber with a concealed loudspeaker and were fitted with a bone vibrator headset. Participants adjusted bone vibrator voltage output to perceptually match the loudness of third-octave noise bands played alternately from the bone vibrator and from the loudspeaker. The judgments were made by six participants, each using two different vibrator models and two different skull locations. Data were analyzed and compared to the data obtained using an artificial mastoid. Discussion will focus on how device types, skull locations, and individual variability affect response curves and comparison to the artificial mastoid and on the feasibility of using the described method as a general method for calibrating and assessing vibrator sensitivity. [Work supported by ORAU.]
Microscale diffusers for absorption of ultrasound.
Authors: Lee Jae-Wan , Shim Jong-Myeong , Lee Hunki , Kang Shinill , Ohm Won-Suk
Published: October 2010

Although diffusers are used primarily for enhanced scattering, they can be designed to absorb sound as part of an anechoic surface. In this study, microscale diffusers are investigated for their antireflection properties in the ultrasonic frequency regime (above 1 MHz). Computations indicate that periodic surface structures on the order of a few to a dozen micrometers can lead to a dramatic reduction in pressure reflection coefficient at frequencies commonly encountered in medical ultrasonics. Possible applications include the design of a thermally resistant absorbing target for a radiation force balance in high-intensity focused ultrasound power measurement.
Speaker identification: The case for speech vector analysis.
Authors: Hollien Harry , Harnsberger James D.
Published: October 2010

The problem of identifying speakers from voice analysis is a serious one. Many analytical procedures have been proposed; most have been based on signal analysis algorithms. Yet it is clear that human perceivers can make accurate identifications (even under difficult circumstances), and extensive research supports this position. How do they accomplish this task? They do so by acoustic and temporal assessment of the talker’s speech/voice signal. Several analysis procedures have attempted to mimic human perception for this purpose, and a brief review of the relevant data will be presented. It will be followed by a presentation of the results of a four-vector experiment in which each vector integrates 3–5 speech parameters. Identification of 28 male voices resulted from three replications in a field of 10 foil voices. It was found that identification scores for the voice, vowel, and fundamental frequency vectors were high, and that for the temporal vector was modest. Moreover, of the summation scores across all vectors and replications, only one failed to identify the target speaker. While these results were based on high-quality audio recordings, it can still be argued that robust speaker identification is possible from a limited set of speech and voice characteristics.
Automatic speaker recognition: Advances toward informative systems.
Authors: Garcia-Romero Daniel , Espy-Wilson Carol
Published: October 2010

Joint factor analysis (JFA) has become the state-of-the-art in automatic speaker recognition systems. In this paradigm, the information contained in a variable-length speech recording is summarized as a fixed-length supervector by means of a soft partition of the acoustic space through a Gaussian mixture model. Moreover, an explicit mechanism to account for the speaker and undesired (inter-session) variability in terms of a small set of factors results in very accurate answers to the question of whether two speech samples are uttered by the same speaker or not. However, no apparent answer to the question of what exactly is that makes two particular voices similar or different is obtained by this approach. To address this issue of interpretability, we propose the modification of the standard JFA approach in two ways. First, by explicitly incorporating phonetic information in the construction of the supervectors so that subsets of its entries get associated with specific phonetic contexts. Second, by modifying the estimation of the speaker and inter-session factors so that phonetically contextualized factors are obtained. A study of the recognition accuracy as well as the interpretability of the results of the proposed approach will be performed on the NIST 2008 speaker recognition evaluation.
Measuring the effects of adaptive multirate (AMR) codecs on formant tracker performance.
Authors: Enzinger Ewald
Published: October 2010

Several approaches to forensic speaker comparison rely on formant center-frequency measurements as features due to their rather straightforward interpretation as resonance frequencies of the cavities of the human vocal tract. Formant tracking algorithms, mostly based on linear predictive coding, are commonly used for automatic extraction. Telephone conversations constitute a substantial amount of forensic material, which increasingly involves wireless communication channels instead of landline transmission. The effects and limitations introduced by the adaptive multirate (AMR) set of codecs that is used for speech transmission in GSM and UMTS networks are therefore of special interest in forensic settings. To evaluate the extent of the effects that are caused solely by the codecs, speech recordings were en- and de-coded with the different bitrate levels provided by the AMR narrowband codec. The formant frequencies of vowel segments were extracted using different trackers and settings. The preliminary results suggest partial shifts in frequency depending on codec level and individual speakers, but no consistent trend emerges.
Parametric representations of the formant trajectories of Spanish vocalic sequences for likelihood-ratio-based forensic voice comparison.
Authors: San Segundo Fernández Eugenia
Published: October 2010

Non-contemporaneous speech samples from 30 Spanish male speakers were compared within the forensic-likelihood-ratio framework. The acoustic parameters studied were the formant trajectories of a series of vocalic sequences (pronounced as diphthongs /we/ /je/, and in hiatus /ia/ /ai/) in order to analyze their suitability for forensic voice comparison. Following Morrison [J. Acoust. Soc. Am. 119, 2387–2397 (2009)], parametric curves (polynomials and discrete cosine transforms) were fitted to these formant trajectories. The estimated coefficient values from the parametric curves were used as input to a multivariate-kernel-density formula for calculating likelihood ratios expressing the probability of obtaining the observed differences between two speech samples under two opposing hypothesis: that the samples were produced by the same speaker and that the samples were produced by different speakers. Cross-validated likelihood-ratio results from systems based on different parametric curves were calibrated and evaluated using the log-likelihood-ratio-cost function (Cllr). The cross-validated likelihood ratios from the best-performing system for each vocalic sequence were fused using logistic regression.
Acoustic analysis and visualization using real time three-dimensional parametric modeling.
Authors: Rife Dave
Published: October 2010

As we move further toward the 3-D world in building design, specifically for non-standard shaped performance spaces, it is becoming more and more useful to easily transition between 3-D models and simple acoustic calculations. Using RHINOCEROS, a widely used and flexible 3-D modeling program, and GRASSHOPPER, a parametric modeling plugin, tools have been developed to streamline simple acoustic analysis and visualization directly from an architectural model. These real time parametric tools help facilitate a clearer understanding of room-specific acoustic phenomenon to the designer, and have proven to be particularly useful in efficiently communicating acoustic principles to non-acousticians. Examples are shown, including visualizing reflection sequences, time delays, distance attenuation, and reverberation time calculations, which are all parametrically driven and updated in real time within the 3-D modeling software.
A simplified two-dimensional model of acoustic diffusion.
Authors: Sequeira Martín , Cortínez Víctor
Published: October 2010

The acoustic control is a topic of growing interest in workspaces. For the purpose of designing acoustic solutions, it is necessary to make correct predictions of sound levels in order to advise the most appropriate acoustic treatments. There are many methodologies to predict the noise levels in industrial rooms, from the simplest empirical models based on diffuse sound fields assumption or experimental data to a more robust method such as the ray tracing technique. A new concept for the acoustic enclosure prediction was recently proposed. [Picaut (1997).] This model may be considered as an extension of the classical theory of reverberation and has been applied to a lot of situations with good agreement and low calculation time. The objective of this work is to provide an equivalent 2-D diffusion model achieving similar prediction results in terms of sound pressure level distributions with a lower calculation time.
Modeling and shaping specular and diffuse reflection sound fields in enclosures.
Authors: Michalis Krista , Bliss Donald , Franzoni Linda
Published: October 2010

Steady-state sound fields in enclosures, with specular and diffuse reflection boundaries, are modeled with an energy-intensity boundary element method using uncorrelated broadband directional sources. The specular reflection case is solved using an iterative relaxation method extended from a 3-D diffuse reflection solution. When chosen properly, only a few directivity harmonics are required. Starting with a diffuse calculation, higher harmonicsare estimated by post-processing, and the original diffuse influence matrix is refined accordingly and converges in a few relaxation steps. The method is higher-order accurate, computationally efficient, and similar to the diffuse case in that it includes strict enforcement of energy conservation. The simulations give accurate results compared to exact modal solutions. Sample calculations are presented to show that the mean-square pressure spatial variation can be tailored using a combination of diffuse and specular surfaces for the design of acoustic spaces.
Optimizing parameters for ultrasound-induced disruption of the blood-brain barrier.
Authors: O’Reilly Meaghan A. , Hynynen Kullervo
Published: October 2010

Microbubble-mediated ultrasound-induced blood-brain barrier disruption (BBBD) is a promising and increasingly investigated technique for the targeted delivery of therapeutics in the brain. Currently in the pre-clinical stages of research, there is a need to establish treatment parameters which will produce consistent and safe BBBD. The BBB was disrupted in rats at 1.18 MHz using modified pulses which have been shown to eliminate acoustic standing waves in the skull cavity. 10-ms bursts consisting of single excitation cycles separated by a set interval were repeated at 1 Hz for 2 min. The transducer used (10-cm aperture, FN=0.8) rang for approximately 3 μs following a single cycle at 1.18 MHz. The interval between cycles was varied using values of 6, 60, 300, 600 μs, and a single cycle every second. Enhancement levels measured via contrast-enhanced T1W MRI decreased as the time interval was increased. This did not appear proportional to the time-averaged acoustic power, and a single excitation cycle every second successfully caused BBBD. Enhancement levels following 5-min sonications with bolus injection of microbubbles versus those with slow infusion were also compared. Enhancement levels were higher following bolus; however, sonications with infusion produced more consistent enhancement levels.
Combined ultrasound and computed tomography for image-guided acoustic therapy.
Authors: Caskey Charles F. , Hlawitschka Mario , Qin Shengping , Ferrara Katherine W.
Published: October 2010

Fusion of computed tomography (CT) and ultrasound (US) imaging is currently used for guiding interventional procedures, including biopsy, local drug delivery, and radiofrequency ablation. While US facilitates real-time visualization of blood flow and tissue structure, CT provides a 3-D view of an extended region of interest, can differentiate tissue types, and operate in a fluoroscopic mode for catheter placement. Our laboratory is working to extend this fusion to include US therapy and imaging co-registered with CT. Here, we present methods for semi-automated registration between tracked US and CT with the objective of imaging and guiding acoustic therapy in small animals. A calibration phantom and fiducial markers were used to generate accurate transformation between the real-time US and pre-acquired CT image stack. The mean error in the transducer calibration was 1.1 ± 0.76 mm with a maximum error of 3.1 mm. A target registration error of 1.7 ± 0.9 mm was observed in US and CT images of a dual-modality phantom. We assessed in vivo feasibility by imaging MET-1 hind limb tumors in a mouse model using both CT and US contrast agents to enhance vascular structures in both modalities. The ability to plan and guide US-induced hyperthermia is currently being tested. [Work supported by NIHR01CA103838.]
Vibro-acoustics of organ pipes: Coincidence effect between acoustical and mechanical modes.
Authors: Gautier Francois
Published: October 2010

How does the wall vibration of a wind instrument affect its sound? This question has engendered a long-lasting debate among scientists, musicians, and instrument makers. In fact, the wall vibrations are easily felt or measured on most wind instruments. However, their influence on the produced sound is more difficult to bring to light, because the fluid-structure couplings involved are weak. This leads to the conclusion that the wall vibration effect does not induce any significant or audible contribution except when particular conditions are gathered. The conditions correspond to coincidence relationships between acoustical and structural modes of the instrument. A century ago, Miller published an experimental study about vibrations of organ pipes. [D. Miller, Science 29, 161–171 (1909)]. He used a double walled organ pipe, for which the space between the two walls could be filled with water while the pipe was sounded, presumably damping the wall vibrations. Miller then observed, without explaining it, that the filling led to the unusual behavior of the pipe, clearly audible. Some heights of the water jacket produced pitch changes or unstable tones. A copy of this historical experiment has been made recently and is now interpreted using the coincidence concept.
Train noise: Peruvian case study to legislate its emission.
Authors: Gushiquen Elena , Montano Walter
Published: October 2010

This work presents the results of train noise EIS in order to write down legislation for allowed maximum limits for Peruvian gas, electric, and diesel convoy.
Toward an engineering of noise design in rituals.
Authors: Elizondo-Garza Fernando J.
Published: October 2010

The design of rites is not something new since shamans, tribal heads, and leaders from very ancient times knew clearly that the conjunction of physical–chemical–biological stimuli produces very useful states of euphoria to reinforce the sense of unity and thus to obtain the favors of multitudes. These rites, or of tribal, religious, sport, or festive character, in many cases, use the noise like one of their central elements. In this paper a structure and tools to the noise design in rituals are proposed.
Resonant ultrasound spectroscopy for small or arbitrarily shaped samples and thin films.
Authors: Maynard J. D.
Published: October 2010

When new materials are developed, among the first properties which should be measured are the elastic constants, and resonant ultrasound spectroscopy (RUS) has been found to be an appropriate method for accomplishing such measurements. However, novel materials to which RUS might be applied are often available only as small samples, only a few hundred micrometers in size, or as thin films deposited on a substrate, with film thicknesses of only a few hundred nanometers. On some occasions samples are fragile or chemically reactive so that they cannot be polished into the shapes required by conventional RUS; for such cases a finite element method is required. In this talk the development of RUS for small or arbitrarily shaped samples and thin films will be presented. [Research supported by NSF DMR 0804105.]
Electronics for resonant ultrasound.
Authors: Migliori Albert
Published: October 2010

An essential advantage of resonant ultrasound spectroscopy (RUS) is that there is no need to bond the transducers to the specimen. Needed to preserve the free-surface boundary conditions so important to RUS, the lack of bonding also makes signals weak. Exacerbating the signal-level problem is a requirement of RUS to measure over a broad frequency range, precluding the use of gain-enhancing resonant transducers. So with broadband transducers and weak contact, the demands on RUS electronics, from preamps to computers, are substantial. The evolution of electronics for RUS has paralleled the availability of high-performance analog-to-digital converters, FET-based electronic components, and digital signal processing. We describe here all the essential features for state-of-the-art RUS signal acquisition. [Work at Los Alamos National Laboratory (LANL) was supported by NSF-DMR-0654118, DOE, the State of Florida. NSF-DMR 0602859 (FSU), and LANL LDRD-DR20070013. LANL is operated by LANS LLC.]
Source segregation in noisy environments by children with normal hearing and bilateral cochlear implants.
Authors: Litovsky Ruth , Misurelli Sara , Godar Shelly
Published: October 2010

Spatial release from making (SRM) refers to the improvement in speech intelligibility measured when target and maskers are spatially separated, as opposed to when they are co-located. SRM is affected by numerous factors that are of interest in this study. Asymmetrical placement of maskers to one side of the head leads to SRM as large as 12 dB in normal-hearing (NH) adults, due to a combination of monaural head-shadow and binaural interaction effects. In addition, SRM is affected by the content of the masker and its similarity to the talker. To better study these effects in children, we recently measured SRM when maskers were either symmetrically or asymmetrically distributed in the horizontal plane to the right and left. In addition, the sex of the talker and thus its similarity to the masker were varied. In the NH groups, children between the ages of 4–6 and 7–9 years and adults were tested. In addition, children who are deaf and use bilateral cochlear implants, also 4–6 and 7–9 years of age, participated in many of the conditions. Results will be discussed in the context of developmentally emerging and lacking abilities to utilize binaural cues for spatial unmasking.
Low-frequency normal mode propagation in shallow water and its application to the acoustic inverse problem.
Authors: Lin Ying-Tsong , Lynch James F.
Published: October 2010

Acoustic normal mode theory is useful to describe low-frequency sound propagation in shallow water. In this talk, normal-mode propagation effects will be reviewed, as well as acoustic inverse approaches based on modal propagation theory. The normal modes of low frequency sound in shallow water are geometrically dispersive due to the presence of surface and bottom boundaries. One can exploit this modal dispersion to solve some acoustic inverse problems, such as passive acoustic localization and geoacoustic inversion. The approaches of matched field processing and backpropagation are commonly used. In addition, the modal wavenumber spectrum contains information on seabed properties that can be used to infer the geoacoustic parameters. Underwater sound propagation is strongly affected by water-column fluctuations, and internal waves are one of the significant effects in shallow water environment. An overview of internal wave effects on normal mode propagation will be presented here, along with their consequence on acoustic inversions.
Acoustic design of wall, ceiling, and plenum for college buildings.
Authors: Roy Kenneth P. , Good Kenneth W.
Published: October 2010

What can you do when an owner wants to design and build flexible learning spaces that can be reconfigured as the needs of the school teaching program changes? Well, we have done a research program on this issue and will share both laboratory and field testing data. Demountable wall systems, raised floors, deep ceiling plenums, stuff you stuff into the plenum, and high CAC ceiling tile. Does it make a difference if the teaching is live voice, recorded voice program, or how about music? Of course, it does. So let us talk sound transmission through the ceiling plenum and what to do about it.
Plenums in apartment buildings.
Authors: Beristain Sergio
Published: October 2010

A construction company decided to build an apartment 10-story building with very little investment. There, dozens of small flats were built, and it was expected that solid floors were to be installed over most of the living dining and sleeping areas (with only a few carpeting in some bedroom areas); flats were delivered with a concrete flooring of 5 cm below final level, so owners could decide the type of floor. In order to reduce the high sound levels from furniture movement and women shoe steps in the above apartment, which were considered the biggest noise problems, the space between the concrete roof surface and the false ceiling was constructed as a plenum, as there was no other way to reduce those sounds. Some sound results and comments are presented.
Predicting vibrational isolation efficiency using mobilities.
Authors: D’Antonio Peter , Vanstraelen Michael
Published: October 2010

To solve complex vibrational problems, we have developed a predictive program using mobility, which is the inverse of the impedance. The analogy with electrical circuits, in which mobility corresponds to resistance, is used to solve mass-spring-damper systems. To determine the vibration isolation between two points, one must first determine the mass and complex stiffness mobilities and use Kirchoff’s rules to define the equivalent mobility in each node. The force is then determined everywhere in the circuit, starting from the excitation force, and used to calculate the speed potential, using the mobility definition. A derivation of the isolation efficiency will be presented. Examples of a 1 degree of freedom system, in which you have a motor isolated from a rigid floor, and a 2 degree of freedom system, in which you have a motor isolated from a resilient floor, will also be given.
An analytical study on active control and mechanisms of sound transmission through double panel-cavity system.
Authors: Jin Guoyong , Liu Zhigang , Chen Yuehua , Yang Tiejun
Published: October 2010

Combing the subsystem modal coupling theory and the impedance-mobility approach, a general analytical model is developed for the analysis and active control of structural acoustic transmission through the double panel-cavity structure into the free field, in which control methods including incident panel piezoelectric (PZT) actuator, acoustic actuator in cavity, and radiating panel PZT actuator are considered. The matrix formulae for the modal response of the coupled structural acoustic system and optimal secondary source strengths are obtained. On the basis, active control of sound transmission is studied analytically and numerically in detail. Simulations and analyzes are conducted to compare the control effects between different control strategies and examine the dominant control mechanisms involved in each control case, and the effect of system parameters on control effect is also examined. The results demonstrate that the incident panel actuator can obtain sound attenuation by modal suppression or rearrangement of three subsystems so many complex control mechanisms are involved and it is effective for incident panel modes, cavity modes, and radiating panel modes. Using the secondary acoustic source in the cavity is a better control strategy compared to the other control strategies due to the dominance of the acoustic mode (0,0,0) in the modal couplings and energy transmission in low frequency range.
Tools and methods for acoustic data storage and retrieval.
Authors: Pavan Gianni
Published: October 2010

Technologies available today allow to record a large amount of acoustic data that require to be stored and catalogued properly to facilitate retrieval, analysis, and comparison. The multisensor/multichannel recording system developed at CIBRA includes tools to georeference recordings and to collect context data (e.g., navigation data when used in research cruises). Specific file naming rules allow an easy retrieval of recordings and an easy connection to GIS maps. The analysis of large amounts of files (e.g., weeks of continuous 24/24-h recordings or months of scheduled recordings made by unattended equipment) is further facilitated by an automatic classification software able to discriminate simple acoustic categories and to produce a summary report about the contents of the recordings.
A review and inventory of cabled and radio-linked hydrophones for passive acoustic monitoring of marine mammals.
Authors: Norris Thomas , Oswald Julie N. , Sousa-Lima Renata
Published: October 2010

Fixed cabled hydrophone (FCH) and radio-linked hydrophone (RLH) systems are permanently, or semi-permanently, installed acoustic monitoring systems that are located on or moored to the seafloor. These systems have the capability to passively monitor bio-acoustic signals from marine mammals and, therefore, have great potential for monitoring and mitigation of potential impacts caused by anthropogenic activities. As part of a Joint Industries Programme sponsored effort, we reviewed past and present FCH and RLH systems with respect to their capabilities, advantages/disadvantages, and effectiveness for monitoring marine mammals in relation to oil and gas exploration/production activities. Based on this review, we provide examples and applications of these technologies. FCHs are typically powered by an external source and send data continuously to a receiving station that is usually located on shore. RLHs are moored or fixed to the seafloor, and transmit acoustic signals via radio-waves to a receiving station on shore. Both these systems allow acoustic data to be remotely monitored and processed in (or near) real-time. Hybrid systems can offer a good compromise between cost and capability by providing near real-time data transmission/processing with greater flexibility in deployment possibilities, but are usually limited in longevity and bandwidth of monitoring.
Dynamical equations for deformation of bubble wall and neighboring tissue interface under acoustical excitation.
Authors: Ilinskii Yurii A. , Hay Todd A. , Zabolotskaya Evgenia A. , Hamilton Mark F.
Published: October 2010

Recent photographs of ultrasound contrast agent microbubbles excited by a high-intensity acoustical pulse in ex vivo blood vessels reveal that bubbles often translate toward the center of the vessel while forming jets in the direction of translation. Vessels have also been observed to first distend but then settle in a contracted state after bubble motion subsides. A model is developed for a translating aspherical bubble at an arbitrary position in an incompressible viscous fluid confined within a channel formed by parallel elastic layers with low shear moduli. Translation of the bubble and deformation of its surface are taken into account by including spherical harmonics for the dipole, quadrupole, and octupole modes in the boundary condition on the bubble wall. Lagrange’s equations yield a system of four coupled second-order ordinary differential equations for the volume, position, and shape of the bubble. Numerical integration of the dynamical equations predicts behavior consistent with experimental observations, i.e., an initially spherical bubble located near one side of the channel and excited acoustically translates toward the center of the channel and initiates a jet in the direction of translation. Predictions of channel contraction are also consistent with observations. [Work supported by NIH DK070618.]
Acoustic scene analysis using microphone arrays.
Authors: Terez Dmitry
Published: October 2010

A spherical microphone array is built to perform acoustic scene analysis in a realistic reverberant environment with several speakers talking at the same time. Speech sound sources and their significant reflections can be localized and grouped together using a newly developed algorithm. The algorithm uses pitch cue to achieve speech sound source segregation and has significantly better time resolution compared to other known methods. The time-delays of arrivals are also precisely computed and match geometric calculations. Attempts are made to mimic some well-known auditory effects in humans such as the Haas effect as an essential first step toward usable computational auditory scene analysis. The proposed methods can significantly improve signal-to-noise ratio of speech input for robust speech recognition by combining sounds from speech sources of interest with their reflections and rejecting all other sources. Real-time algorithm implementation on commodity hardware is enabled by using a graphics processor.
Image enhancement of the temperature field by the inverse estimation using the acoustic pyrometry.
Authors: Kim Tae-Kyoon , Ih Jeong-Guon
Published: October 2010

In the acoustic pyrometry for the measurement of temperature distribution of a cross section in a furnace, the temperature can be reconstructed from the measured time-of-flights between sensor-actuator array systems. The calculated transfer matrix is composed of the approximated slowness function and its coefficient vector. In this study, the least-squares method is used to inversely reconstruct the temperature. If the transfer matrix is highly singular, reconstruction result will be deteriorated in the presence of the measurement error. To reduce the effect of the singularity, the redundancy of the sensor data can be removed and/or the regularization of the restored field can be employed. As a simulation example, a thermal field within a rectangular cross section of a 1-D infinite duct with asymmetrical temperature distribution was considered. Condition number of the transfer matrix revealed a highly singular state of the system, regardless of the number of sensors. Sensor/actuator positions were selected by using the effective independence method. A modified Tikhonov technique was adopted for the regularization. Although the resultant precision depended on the number of sensors, the reconstruction error was drastically reduced, from 214% to 14% in the case of using 36 sensors, in particular by applying the regularization. [Work supported by BK21.]
Fabrication and characterization of a sapphire based fiber optic microphone for harsh environments.
Authors: Griffin Benjamin A. , Mills David A. , Schmitz Tony , Sheplak Mark
Published: October 2010

This paper describes the fabrication and characterization of a sapphire based fiber optic microphone designed for the harsh environment of gas turbine engines. The performance requirements are driven by turbine inlet temperatures that have risen in excess of 1500 °C. The harsh environment makes conventional instrumentation unsuitable for time-accurate, continuous, direct measurements. The use of commercially available sapphire substrates and optical fibers allows for performance in extremely high temperature environments due to the high melting point and matched coefficient of thermal expansion. The sensor is based on a fiber optic lever transduction mechanism with a remote photodiode optical readout allowing for isolation of the electronics from the harsh environment. The microphone’s diaphragm consists of a thin sapphire substrate with a sputtered titanium/platinum reflective surface. The back cavity is formed by macro-machining of a 1-mm-thick sapphire substrate. The diaphragm and back cavity are joined using high temperature, alumina based epoxy. The paper focuses on the fabrication and packaging of the sensor with characterization results given using silica based optical fibers at room temperature. The future work is aimed at the implementation of sapphire optical fibers and high temperature characterization. [Work supported by the Air Force.]
Interactive exploration of three-dimensional deflections on vibrating structures using linked multiview displays.
Authors: Villarreal José Luis , Torres Jesús Alejandro , Ramírez Ramón
Published: October 2010

A software was developed to explore and compare structural deflections under different conditions. Modal analyzes of the same structure with slight variations were previously calculated through finite element method (FEM). Four modal analyzes of a guitar top plate (at different stages of its construction) were compared as an example using the developed visualization software. The resulting FEM meshes were imported to the developed application due to the incapability of the FEM software for real time visualization of the results of different modal analysis data. In the main screen of the developed software were shown several mode shapes. The user picks any mode shapes to compare, and their 3-D controllable versions appear in full or in divided screen versions (depending on the number of modes picked). The developed software allowed the models to be freely turned, zoomed, and moved using any standard laptop. The whole software characteristics brings a powerful technique to researchers and teachers, exploring and explaining some details in 3-D structural deflections, often limited and showing only static 2-D images.