bluuebunny/crossref_metadata_2025 huggingface dataset:
Perceptual and articulatory noise in vowel imitation
Authors: Vallabha Gautam K. , Tuller Betty
Published: October 2013

Speakers exhibit systematic errors when asked to imitate self-produced vowels [B. Tuller and G. Vallabha, J. Acoust. Soc. Am. 107, 2855 (2000)]. These results were replicated (3 speakers, 45 systematically distributed self-produced vowel targets, 10 imitations per target) and compared to a simulation of the effects of articulatory and perceptual noise. Articulatory noise was modeled using the Haskins Articulatory Synthesizer. The tongue body center (TBC) was interpolated between the configurations for /i/, /a/, and /u/ and 90 of the intermediate configurations were perturbed by adding independent Gaussian noise (sd=1 mm) to the x and y coordinates of the TBC. Perceptual noise was modeled by (1) transforming each of the 90 intermediate vowels to F1/F2 bark (‘‘perceptual’’) space and (2) adding independent Gaussian noise (sd=vowel discrimination limen=0.28 barks) to each bark coordinate. The models did not match the directions of the imitation errors made by the speakers. However, the distribution of the error magnitudes, relative to the mean imitation, was approximately Rayleigh (0.28) across both the models and the speakers. These results indicate that while speech perception and/or production may have intrinsic Gaussian noise, this noise is not responsible for the systematicity of the imitation errors. [Work supported by NIMH.]
Phonemes at the cocktail party: Insights on the processing of competing speech signals
Authors: Bashford James A. , Warren Richard M. , Lenz Peter W.
Published: October 2013

A recorded word repeated over and over is heard to undergo a series of illusory changes (verbal transformations) to other syllables and words in the listeners lexicon. When cocktail party listening conditions are simulated by adding a second image of the repeating word through dichotic presentation with an interaural delay preventing fusion, the resulting two distinct lateralized images of the word undergo independent transformations [Lenz et al., J. Acoust. Soc. Am. 107, 2857 (2000)]. The present study found that adding a second image of the same word increased the extent of phonetic distortions for each image, whereas presenting a different competing word reduced the transformation rate by one-half when the competitor differed by one phoneme (e.g., ‘‘dark’’ paired with contralateral ‘‘bark’’) and by three-fourths when it differed in all phonemes (e.g., ‘‘dark’’ versus contralateral ‘‘light’’ or ‘‘might’’). These interactions were independent of semantic relations and occurred with competitors from both same and different speakers. In contrast, transformations were unaffected by a nonspeech competitor (a repeating contralateral door slam). These and additional results indicate that shared resources at an abstract level of phonological analysis provide the basis for the independent syllabic/lexical organizations derived for competing speech signals. [Work supported by NIH.]
Resolution enhancement algorithm utilizing multiple reflections for image reconstruction in the conduit
Authors: Teramoto Kenbu , Tsuruta Kohsuke
Published: October 2013

In the acoustical endscopy, due to the physical limitations, the transducer array is composed of a small number of elements and each interspacing is larger than the acoustical wave length that is called a sparse array system. In such case, the volumetric image reconstruction process becomes ill-posed caused by the interference of the reverberation in the tube. Consequently, the reconstructed image is interfered by widely spread strong spurious lobes. This paper proposes a novel volumetric image reconstruction algorithm for a 3D sparse aperture acoustical holography, which visualizes the forward-looking image in the conduit tube. The proposed algorithm utilizes prior knowledge about both field, object, and observed signal as (1) the support of the object is limited within the conduit, and (2) the multiple reflected wave field scattered by the object is statistically independent of the reverberation in the tube. The proposed sparse array acoustical imaging system utilizes, therefore, the multiple reflected wavefronts in order to enhance the resolution. By experiments, it is proven that the proposed algorithm has the following improvements: (1) Targets can be identified when multiple reflections exist in the tubes. (2) Artifacts caused by the spurious lobes can be reduced.
Two-dimensional synthetic focusing using large apertures—Deconvolution of the spatial impulse response
Authors: Lingvall Fredrik , Stepinski Tadeusz
Published: October 2013

The synthetic aperture focusing technique (SAFT) has been used successfully in both medical and nondestructive evaluation (NDE) applications of ultrasound. Transducer size is an important issue when implementing SAFT algorithms. The classical SAFT method is based on the assumption that a point source is used for emitting ultrasonic waves, which means in practice that the transducer aperture is so small compared to the wavelength that it emits spherical waves. If the transducer aperture is large its spatial impulse response (SIR) significantly differs from a Dirac pulse and nonspherical waves will be generated. Therefore, to successfully perform synthetic focusing using large apertures, the transducer’s SIR must be taken into account. A new method for compensating transducer SIR in synthetic focusing is presented in the paper. The method employs a stochastic, time-domain Wiener filtering technique for two-dimensional synthetic focusing. Results of the experiments performed in water using a linear phased array are presented to demonstrate the performance of the proposed method. The use of an array enabled altering the aperture without changing the electrical characteristics of the whole system. The results obtained using the proposed technique for finite apertures are compared to those obtained with a classical SAFT algorithm.
Seabed layer discrimination using normal incidence backscatter returns in the 1–10 kHz frequency band
Authors: Heald Gary J. , Hines Paul C.
Published: October 2013

It has been established that the first and second seabed returns from a monostatic echo sounder can be used as a seabed classification tool. These systems operate on the principle that the first echo is purely monostatic, whereas the second echo can be treated as a bistatic return with a virtual source located at a height of 2–3 times the water depth. Typically these systems operate in the range of 100 kHz so that bottom roughness is the dominant scattering mechanism. In the present work we apply this procedure to lower frequencies. This should provide additional discrimination information since contributions from the seabed volume become significant. Additionally, it provides information about the nature of seabed layers that may be present. However, this comes at the cost of further complicating the model. In this paper we modify the high frequency model to be applicable at lower frequencies and examine the effect of sub-seabed layering.
Acoustic scattering from stratified marine sediments
Authors: Ivakin Anatoliy N.
Published: October 2013

There are various mechanisms of seabed scattering. Normally, they are attributed to two main types of bottom irregularities: roughness of the sediment interfaces and volume fluctuations of the acoustical parameters. However, there is ambiguity in the description of these irregularities. It is known, e.g., that roughness can be exactly expressed in terms of volume perturbations near the flat interface. On the other hand, there is a class of volume perturbations within continuously stratified media, which can be described in terms of the roughness of isosurfaces of acoustical parameters. Here, all these cases are considered using a unified approach to volume and roughness scattering [A. Ivakin, J. Acoust. Soc. Am. 103 (1998)]. This approach unifies also the cases of continuous and discrete stratification (layering). Rough interfaces between layers are replaced by continuous transition layers having rough isosurfaces of the acoustical parameters. First-order perturbations of these parameters are proportional to their vertical gradients and to displacement (roughness) of the isosurfaces within the transition layers. The model gives the volume scattering strength as a function of the wave thickness of the transition layers. In the particular case of low frequencies, where the transition layers are thin in comparison with the wavelength, the model provides an exact transition to existing results for scattering from rough interfaces and, thus, naturally links volume and roughness mechanisms of seabed scattering at low and high frequencies within a unified approach.
Validity of backscatter strength models at high (110–170 kHz) frequencies
Authors: Page Sarah J. , Brothers Richard J. , Thomson Patricia A. G.
Published: October 2013

Sonar performance models use several low-level acoustic models to predict the effect of the environment on reverberation and noise. Many of these acoustic models, particularly those predicting seabed scattering strength, were developed at lower frequencies than those currently used by both the scientific and mine-hunting communities. This paper examines the popular Applied Physics Laboratory of the University of Washington (APL-UW) acoustic backscatter strength model and tests its validity in the extended frequency band of 110–170 kHz. Acoustic and environmental data (gathered as part of a joint program with NATO SACLANTCEN) from several sites are collated and compared in this paper. This provides compelling evidence regarding the validity of the APL-UW backscatter model at these higher frequencies.
Acoustic field of hydroacoustic antenna arranged near a rough surface
Authors: Dat Tran. Huy
Published: October 2013

The characteristics of an averaged field of a hydroacoustic antenna arranged near a penetrable ocean surface with large-scale random roughness is formulated. The model of an antenna is described as an activity cylindrical surface, restricted absolutely rigid toroid. The given model rather convincingly mirrors actual design features of modern hydroacoustical antennas. The averaged field of a hydroacoustic antenna is analyzed depending on a degree of roughness of boundary, sizes of antennas, field pattern on activity surfaces, way of the view of space and bottom penetration. It showed that, the rough surface could result into differences in distribution of averaged values of a far field of hydroacoustic antennas up to 20% in comparison with a case of free environments, the value bottom penetration, after which one of the distribution values of the mean characteristics of a near field of the antenna practically coincides appropriate values for a free environment, is established. By the given calculation and discussion, it is followed that, the ocean rough surface could be considered as the natural factor resulting in decreasing of acoustic shadow. Nevertheless, bistatic calculations of coherent field suggest useful strategies for improving in design a new model of the hydroacoustic antenna.
Speech levels in open plan offices
Authors: Warnock Alf , Chu Wing T.
Published: October 2013

A key variable for predicting speech privacy in open offices is the level of speech used. Since the environment influences vocal effort, speech levels needed to be measured in typical open office situations to more accurately calculate speech privacy in such offices. This paper describes measurements of voice levels used in face-to-face conversation in nine open offices in the Ottawa/Hull Area. Speech from 118 subjects was recorded and analyzed. Average voice levels in each office ranged from 48 to 55 dBA. The overall mean was 51 dBA with a standard deviation of 3.4 dBA. Levels from the current study were significantly lower than those in ASTM standard E1130, but the overall average and spectrum agree well with Pearsons’ results for casual conversation (K. S. Pearsons, R. L. Bennett, and S. Fidel, ‘‘Speech levels in various noise environments,’’ EPA-6001-77-025, U. S. Environmental Protection Agency, May 1977). The paper describes the procedures and the results obtained. This work was done as part of a project for Public Works and Government Services Canada.
Acoustic concerns related to multi cultural societies
Authors: Gade Anders Chr.
Published: October 2013

Immigration has increased cultural diversity in western societies. The process of integrating immigrants into their host countries can be smoothed if acousticians learn to recognize (1) the acoustic traditions of immigrant cultures and (2) the specific acoustic needs of the new society members. Two related projects are discussed. The ‘‘Cahrisma’’ project (Conservation of Acoustical Heritage by the Revival and Identification of the Sinan’s Mosque Acoustics) is sponsored by the European Commission and carried out in cooperation among researchers in Turkey, Malta, Italy, France, Switzerland, and Denmark. Its purpose is to combine visual and acoustical concerns in the identification, conservation, and restoration of architectural heritage. It focuses on the famous Turkish mosques of the Osmannic architect Sinan. Some of the acoustic features of these large domed buildings and of muslim liturgy will be presented. Another project was conducted by students at the Danish Technical University, who found that speech intelligibility for non-native listeners is much more sensitive to poor speech conditions than for native listeners. Also discussed are the unique features of culturally based urban soundscapes. It is suggested that these soundscapes can provide comfort to recent immigrants by increasing their sense of being ‘‘at home.’’
Songlike vocalizations and infrasound from the Sumatran rhinoceros
Authors: von Muggenthaler Elizabeth , Reinhart Paul , Lympany Brad
Published: October 2013

Within the last ten years the Sumatran rhino (Dicermoceros sumatrensis) population has dropped 50%, and only 200–300 individuals are left in the world. The oldest living species of rhino in evolutionary terms, Sumatran rhinos are solitary, although males and females are seen together during courtship. Their native habitat is dense tropical forest and mountain moss forest. They are the smallest living rhino, standing 0.09–1.5 m tall, and are covered in course, reddish-brown hair. Three Sumatran rhinos, housed at the Cincinnati Zoo, were recorded from 1–3 m. Two Statham Radio microphones, and two Sony TCD-D8 DAT recorders recorded from 9 Hz to 22 KHz. Analysis, including FFTs, spectrographs, and filtering, were performed using National Instrument’s Polynesia. The rhinos proved to be extremely vocal, producing signals almost constantly. Distinct calls, including several types of ‘‘eeps,’’ 70 Hz–4 kHz (57–92 dB); ‘‘whales,’’ 100 Hz–3.2 kHz (87 dB); and ‘‘whistle-blows,’’ 17 Hz–8 kHz (100 dB) were discovered. The ‘‘whistle-blow’’ has high dB infrasound that would be advantageous for use in the rhino’s forest habitat. Some Sumatran rhino vocalizations sound similar to and resemble (under analysis) some humpback whale signals.
Ultrasound contrast agents to enhance and monitor therapy
Authors: Ferrara Katherine , Chomas James , May Donovan , Allen John , Pollard Rachel , Wisner Erik
Published: October 2013

New techniques for the assessment of tissue perfusion and the local delivery of drugs have been developed using ultrasound contrast agents. A perfusion estimator, based on the destruction and wash-in of ultrasound contrast agents, has been developed and applied to tumor models. This technique requires the transmission of a train of pulses with varying amplitude, frequency, and phase within the train. High-amplitude and low-frequency pulses are applied to destroy the agent, and low-amplitude higher-frequency pulses are used to monitor the refresh of contrast agents into the sample volume. The signal processing techniques applied following data acquisition utilize the subharmonic component of the received signal. Mechanistic studies of contrast agent destruction, a Rayleigh–Plesset based signal model, and experimental data acquired from a rat tumor model will be included in our presentation. In addition, studies of the local delivery of drugs using oil-based drug delivery vehicles have been conducted. Results of these studies demonstrate that the probability of agent fragmentation and the size of the resulting fragment can be controlled using ultrasonic parameters. [We acknowledge the support of NIH CA 76062.]
Equilibrium loading of cells with molecules by ultrasound: Effects of molecular size, acoustic energy, and cell-to-bubble ratio
Authors: Prausnitz Mark , Guzman Hector , Schlicher Robyn
Published: October 2013

Ultrasound has been shown to disrupt cell membranes and thereby load cells with molecules. In this study, we quantified levels of cellular uptake of calcein (0.6 kDa), bovine serum albumin (66 kDa), and two dextrans (42 and 464 kDa) into DU145 prostate cancer cells using 500 kHz ultrasound in the presence of Optison contrast agent. The results demonstrated that levels of uptake (i) were similar for all four molecules at the same ultrasound conditions and (ii) for some cells reached thermodynamic equilibrium with the extracellular solution. Over a broad range of acoustic conditions, these levels of uptake, as well as loss of cell viability, were shown to correlate with acoustic energy exposure. In a separate set of experiments, increasing cell concentration was shown to increase both viability and uptake, and increasing Optison bubble concentration decreased viability with no significant effect on uptake. As a parameter that can unify these two sets of data, the cell-to-bubble ratio was found to correlate well with both cell viability and the number of cells with uptake normalized relative to the number of Optison bubbles provided, which can be viewed as a measure of bubble ‘‘efficiency.’’
Sonodynamic treatment with cavitation-promoting agent activated by second-harmonic superimposed progressive waves
Authors: Umemura Shin-ichiro , Kawabata Ken-ichi , Sugita Nami , Sasaki Kazuaki
Published: October 2013

Sonodynamic therapy has been proposed based on the finding that certain chemicals such as hematoporphyrin are activated by acoustic cavitation and thereby induce a significant antitumor effect. It has also been found that sonodynamically active cavitation can be induced at a relatively low ultrasonic intensity through superimposing the second harmonic onto the fundamental. An in vivo as well as an in vitro cavitation threshold was an order of magnitude lowered by second-harmonic superimposition especially when combined with a cavitation-promoting sonochemically active agent such as rose bengal. Amphiphilic derivatives of rose bengal were synthesized so as to obtain tumor accumulating agents of this kind for sonodynamic tumor treatment. The tissue distribution of the derivatives in mice was investigated and those having a normal alkylic chain with 10–15 carbon atoms were proved to accumulate well in tumor tissues. With an administration of one of the derivatives, experimental murine tumors were treated with second-harmonic superimposed progressive waves at 0.5 and 1 MHz. The tumor growth was significantly suppressed by sonodynamic treatment in this combination while ultrasound alone showed only a slight effect. [Work supported in part by the National Research and Development Program for Medical and Welfare Apparatus, Japan.]
Inflatable lightweight low-frequency loudspeaker
Authors: Corsaro Robert
Published: October 2013

In an aerospace application we required a low-mass sound generator with good performance at low audio frequencies (i.e., 30–300 Hz). One approach which has been successful in meeting these needs makes use of an inflatable tube. The actuation component is a ‘‘Thunder’’ actuator (Face Intl. Corp.), selected for its high displacement and fast response. Conventionally, when coupled to loudspeaker diaphragms these actuators have failed to produce the combination of force and displacement needed for low-frequency sound generation. In the present device the actuator is coupled instead to a flat, compliant diaphragm supported on the edges by an inflatable tube. Sound pressure levels measured in the farfield (25 cm) using only 200 V peak drive (one-third or its rating) we found nominal levels of 746 dB over the band from 38 to 330 Hz. In the uninteresting region above 330 Hz the output is somewhat higher, but remains well behaved to at least 2 kHz. [Work supported by the NRL/ONR Smart Blanket program, which is exploring the use of active controlled surface covers for reducing sound levels in payload fairing regions.]
Directivity of the diffracted field around a half-plane
Authors: You Jeong Ho , Menounou Penelope , Wright Wayne M.
Published: October 2013

The directivity of the diffracted field around a half-plane barrier is investigated. Experiments were carried out in air using a spark source generating N-shaped pulses, a capacitor microphone, and a rigid aluminum plate with: (i) straight top edge (reference case), (ii) jagged top edge, and (iii) straight edge covered with sound absorptive material. Measurements were taken around the plate by varying the angular position of the microphone while keeping its radial distance from the barrier edge constant. The following were observed: (i) Straight edge: The diffracted pulse changes polarity as it passes across the two shadow boundaries and the continuation of the barrier in the half-space above it. (ii) Jagged edge: Although jaggedness in the top edge of the barrier drastically changes the shape of the diffracted pulse, the directivity pattern remains roughly the same. (iii) Sound absorptive edge: Although the shape of the diffracted pulse remains the same as in the reference case, the directivity patterns change considerably and changes are more pronounced at the source side. The experimental data and observations were in good agreement with theoretical predictions from the Directive Line Source Model [Menounou et al., J. Acoust. Soc. Am. 107, 103–111 (2000)]. [Work supported by Texas ATP.]
Numerical prediction of noise generated from low-pressure axial-flow fans
Authors: Song Li , Dongbao Wang
Published: October 2013

Low-pressure axial-flow fans are widely used in ventilation, air conditioning, and cooling, but it is very difficult to predict their noise. In 1993 Lee et al. predicted the noise level of low-pressure axial fans according to a wake-flow shedding model based on the mechanism that the noise is induced by a lift force pulse caused by wake-flow vortex shedding [J. Sound Vib. 164, 327–336 (1993)]. The main approximation that results in considerable errors is that the parameters of a flat-plate boundary layer at the trailing edge are adopted instead of that of a blade boundary layer. In this paper the parameters of the boundary layer, including the distribution of nominal thickness and momentum thickness of boundary layer along the blade height, are obtained by numerical simulation of the 3-D turbulent flow field in axial-flow fans. It is shown from the predicted results for two model axial fans that the noise level agrees with measured data quite well. The errors are 2–4 dB for A-weighted sound pressure level (SPL) and 3–4 dB for linear SPL. Furthermore the effects of parameters of the blade boundary layer on noise level and spectrum are also discussed in more detail in this paper.
Source models for turbulence generated sound in ducts
Authors: Abom Mats
Published: October 2013

Turbulence or flow generated sound is a problem in many duct and pipe systems. This type of noise will add to the primary noise from fluid machines (e.g., fans) connected to the system and often sets the limit for the sound attenuation possible. For low Mach numbers the flow generated noise will primarily be associated with unsteady flow separation occurring, e.g., at constrictions and bends, resulting in a dipole type of source. In this paper a procedure for deriving source models for turbulence generated sound in the low-frequency plane wave range is described. The source models are formulated as active acoustic 1- and 2-ports [H. Boden and M. Abom, Acta Acustica 3, 549–560 (1995)], which makes them suitable for implementation in software for analysis of duct networks. Regarding the active part or the source strength, it can be obtained using a semi-empirical model for flow separation noise [P. A. Nelson and C. L. Morfey, J. Sound. Vib. 79, 263–289 (1981)]. The passive part can be well approximated in the low-frequency limit by a quasi-stationary model derived from the conservation of mass and momentum in the flow. Examples of tests and validation of the proposed model are also presented.
The effect of active noise control actuator placement on the total sound power radiated by modally vibrating cylinders and plates
Authors: Cooper Jennifer L. , Tichy Jiri
Published: October 2013

Many practical radiators can be modeled as vibrating bars or plates. The radiation caused by their modal vibration is studied replacing them by point sources at each antinodal location. Active control of their radiated sound power is investigated using point secondary sources. The maximum obtainable attenuation is examined as a function of the distance from primary to secondary sources and the separation of secondary sources. For a given distance from the primary sources, the obtained attenuation is independent of secondary source separation for a range of wavelengths. For larger secondary source separations, the control can be simplified, allowing each secondary source to be controlled by only one microphone, as opposed to the usual method of controlling all secondary sources by all the microphones.
Acoustic agglomeration experiments and modeling
Authors: Meegan G. Douglas , Peterson Chris E. , Kumon Ronald E.
Published: October 2013

Acoustic agglomeration of aerosols has been known since at least 1931 when it was observed that particles tend to ‘‘stick’’ together in the presence of an intense acoustic field, thereby forming larger particles. Since the 1970s acoustic agglomeration has been an active area of research because of its potential applications for the control of particulate emissions resulting from fossil fuel combustion. In this talk, experimental methods used in the past to study acoustic agglomeration are described, along with improvements we have introduced. A simple standing wave tube is one of the devices used in our laboratory to measure the dependence of acoustic agglomeration on sound pressure level, frequency, and exposure time. Results are presented and compared with theoretical predictions based on existing models. The potential application to diesel engine emissions and the filtration of hazardous particulate will be discussed.
A computational study of acoustic streaming in a duct
Authors: Erskine William R. , Selamet Ahmet
Published: October 2013

The present work is a continuing investigation of Rayleigh streaming, which is a nonlinear phenomenon characterized by acoustically driven flows attenuated by viscous effects due to the presence of walls, resulting in nonzero average streaming velocities and circulation zones within a duct. A 2-D long and narrow duct is considered with one end closed and the other end subject to a sinusoidal acoustic velocity at quarter-wavelength frequencies and varying amplitudes, leading to a standing wave within the duct. A computational fluid dynamics code based on the Pressure Implicit with Split Operators (PISO) algorithm and finite volume method is used to solve the unsteady, compressible Navier–Stokes equations with turbulence modeling. Since the formation of streaming depends on the velocity profile within the Stokes layer, the grid resolution is increased near the walls. The effect of inlet conditions is studied in terms of circulation patterns, including the number of zones per wavelength, penetration depth into the fluid, and mean velocity profiles across the boundary layer. Some of the computational predictions are also compared with the classical analytical treatments.
Performance of a thermoacoustic prime mover without a stack
Authors: Turneaure Stefan J. , Wakeland Ray Scott , Keolian Robert M.
Published: October 2013

A standing-wave thermoacoustic prime mover which does not contain a stack has been constructed and tested. The device consists of two heat exchangers and a resonator body. The cold heat exchanger may be moved relative to the hot heat exchanger while the resonator is sealed and thermoacoustically generated sound is present. The device has a cross-sectional area of about 500 cm2 and operates near 600 Hz with ambient air as the working gas. It will be explained how a no-stack thermoacoustic device may in principle have a greater efficiency than a stack-based device. In contrast to the well understood stack-based standing wave thermoacoustic engines, it is expected that the efficiency for a no-stack device will be nearly proportional to the acoustic pressure amplitude. Preliminary measurements on the device have achieved acoustic pressure amplitudes of 2.5% of the mean pressure. An interesting result is the presence of spontaneous thermoacoustic oscillations even when the gap between the cold and hot heat exchangers is several times larger than the gas displacement, 2x1. [Work supported by ONR.]
Investigation of stack materials for miniature thermoacoustic engines
Authors: Abdel-Rahman Ehab , Kwon Young , Symko Orest G.
Published: October 2013

In a thermoacoustic engine one of the most critical components is the stack that interacts thermally with a sound field. Its effective surface area determines directly the cooling power and efficiency while the thermal properties determine the losses. Moreover, as the engines are miniaturized, stack performance on a small scale becomes a limiting factor, especially when a stack length of 1000 μ or less has to maintain temperature differences of around 30°C. We have investigated as stack material cotton wool, glass wool, ceramics, silica aerogel, carbon fibers, and carbon powder by measuring flow resistance at high acoustic frequencies, 4–20 kHz, and thermal transport properties. The mechanical properties and ease of machining to small dimensions, of the order of 1 μ, have been studied. In selecting the ideal material for very small devices, a compromise needs to be made between thermoacoustic and mechanical properties. [Work supported by ONR.]
The ‘‘opposite direction shift’’ phenomenon in sound localization using two sound sources
Authors: Fujii Hiroyuki , Kakehi Kazuhiko
Published: October 2013

It has been known that in a situation where tone and noise are presented at once, the perceived direction of the tone is shifted to the opposite direction of the noise. However, it is still not very clear which cues for sound localization are responsible for this ‘‘opposite direction shift’’ phenomenon. In this study, three factors of two sounds were investigated: temporal configuration, frequency overlapping, or level difference of two sounds. The experiment was based on a subjective sound localization task of the designated sound source among two sound sources. The target sound was a 1 kHz or a 500 Hz tone, and the distractor sound was pink or white noise or, notched noise. A target sound was presented for the cases where the distractor sound was at 25° left or 25° right of the loudspeaker array. The perceived direction of the target sound was compared. The ‘‘opposite direction shift’’ phenomenon was observed for the cases where the distractor was longer in duration or had a larger level than the target, or is overlapping with the target frequency. These results suggest that the important perceptual factor of the ‘‘opposite direction shift’’ phenomenon is related to the distractor’s ‘‘saliency.’’
Detectability of phase change and its computational modeling
Authors: Akagi Masato , Nishizawa Masato
Published: October 2013

The ability to detect a phase change was measured in paired-stimuli comparison-experiments using monotonic phase (MPH) and phase-shifted (SPH) complex tones. Fundamental frequencies (F0) were 62.5 to 250 Hz in 1/4-octave steps. The number of harmonics was set to cover frequencies, 300–9300 Hz. All of the harmonics except one were started in sine phase. The remaining component was shifted pi (SPH) or not shifted (MPH). Subjects indicated whether paired stimuli were ‘‘the same or not’’ and reported whether phase-shifted harmonics appeared to ‘‘pop-out.’’ A computational model based on AIM (Patterson, 1995) was constructed and the experimental results were interpreted from autocorrelograms (ACGs) calculated by the model. The findings are as follows. When F0 was below 150 Hz and the number of harmonics in ERB at the phase-shifted harmonic exceeded 2, a phase change was detected and ‘‘pop-out’’ phenomena occurred. When F0 was over 200 Hz, however, MPH and SPH were not discriminated and there was no ‘‘pop out.’’ Discriminable paired stimuli have different ACGs and this reflects the detection ability of phase changes. Fine peaks in the ACG of SPH were found to correspond to the periods of the phase-shifted harmonics. [Work supported by CREST, JST.]
Backward interference of easily identified tones by simple and complex maskers
Authors: Pastore Richard E. , Berens Melody S.
Published: October 2013

There are several different types of interference in the perception of brief target tones due to the presence of subsequent stimulus in close temporal proximity. Backward masking is interference in the detection of the target tone, with the masker usually either a tone or noise. Backward recognition is interference of a tone masker in the discrimination of target tones typically separated by no more than a few jnd’s. Components of natural temporal sequences are usually more complex than tones and more structured than noise, with major distinctions contrasted by more than a few jnd’s. For example, brief initial position stop consonants are followed by complex vowel formants. The current study examines the ability to differentiate initial tones that, in isolation, are easily detected and distinguished (a musical fifth). In addition to noise, subsequent masking stimuli consisted of all possible combinations of three tones that are easily differentiated from each other (a musical third) and from the target tones. There is little backward interference from noise, small to moderate interference from individual tones, and significant interference from specific combinations of tone maskers. Thus backward interference is not limited to near threshold detection or discrimination.
Pitch interval perception by musically trained listeners
Authors: Reetz Henning
Published: October 2013

This research investigates the reaction times to identify musical intervals by subjects with some musical training. All intervals within an octave, upward and downward, consisting of all combinations sine tones, complex tones with a fundamental, and complex tones without a fundamental were presented to a group of 18 subjects. Subjects had to decide which signal pairs form a specific musical interval. For some subjects the identification of certain intervals (e.g., a third major upward) was only 20 ms longer than the identification of two tones with identical pitch but the overall reaction time doubled when the same interval was presented downward. Some subjects showed a great dependency on the presence and absence of the fundamental whereas other subjects were not influenced by this parameter. Furthermore, the sort of musical instrument played by the subjects seem to influence the speed of perception of certain intervals. In total, reaction times indicate that the perception of the missing fundamental is idiosyncratic and depends on the musical education of the subjects and is not necessarily a part of normal auditory processing.
Computer simulations of hearing aid acoustical system performance
Authors: Szwoch Grzegorz , Kostek Bozena
Published: October 2013

The recent developments in the hearing aid technology enabled a number of improvements in hearing aids. This includes advanced signal processing algorithms, better speech intelligibility, miniaturization etc. One of the existing limitations is, however, the problem with providing patient-related characteristics of the acoustical system of a hearing aid. The aim of this paper is to show that using the physical modeling method it is possible to first build a model of the acoustical system of a hearing aid and then to simulate its performance. The waveguide model of the acoustical system of a hearing aid is proposed. Exemplary results of the computer simulations using such a model are presented and compared with some measurement data of existing hearing aid acoustical systems. The model proved to behave similarly to the real system. Conclusions regarding the application of such a method in the fitting process of a hearing aid are included.
Adaptive dual-microphone directional noise reduction scheme for hearing aids using nearby microphones
Authors: Kompis Martin , Bettler Matthias
Published: October 2013

A new directional dual-microphone noise reduction scheme for hearing aids is presented. The acoustic field is sampled by two nearby microphones, which can be mounted conveniently in a single behind-the-ear or in-the-ear unit, such as a hearing aid or the external part of a cochlear implant system. A combination of time-invariant and adaptive postprocessing of the microphone signals is employed. First, time-invariant signal processing is used to combine the two microphone signals into two intermediate signals, one containing predominately sounds arriving from the front, one predominately from the rear. These signals are postprocessed by an adaptive noise canceller using filter lengths between 1 and 50 ms. Experiments have been performed in several simulated, realistic acoustic environment. In a moderately reverberant room (reverberation time 0.4 s, volume 34 m3) improvements in signal to noise ratio (SNR) of 3.4–8.7 dB were achieved, depending on the length of the adaptive filter and the angle of incidence of the noise signal. A classic time-invariant dual-microphone approach yielded 0.2–6.9 dB lower SNRs in the same acoustic settings. The computational load of the proposed system is well within reach for current hearing aid technology. [Work supported by the Swiss National Science Foundation.]
Implementation of narrow-band noise algorithms according to ANSI S3.6-1996
Authors: Ruiz Mariano , Salvador Ruben , Recuero Manuel
Published: October 2013

In order to generate narrow-band noise according to ANSI S3.6-1996 in an efficient way, it is necessary to employ multirate signal processing techniques. Thanks to the successive conversion of the sampling rate in the stages of the system, an efficient implementation of the required algorithms is obtained. These signals are entirely bandpass signals. The adopted solution is to design equivalent lowpass filters followed by a frequency modulation of the filtered signal with the purpose of moving it to the desired frequency. To obtain a better performance in the resulting filter, multistage structures have been used to achieve the required changes in the sampling rate to implement the lowpass filter by means of a multirate signal processing technique. A LabVIEW application have been implemented for designing these types of filters. This program generates the filter coefficients and offers the possibility of carrying out a simulation and saving these coefficients into a C header file. These filters have been successfully tested, on a new DSP based audiometer that uses TI TMS320C6701, by means of Code Composer Studio and a series of routines that implement each one of the stages that compose the filter.
Real-time switching of piezoelectric shunts for structural vibration control
Authors: Clark William W.
Published: October 2013

The use of piezoelectric actuators to passively add damping to vibrating structures has been investigated for over a decade. Simple resonant shunt circuits consisting of resistors and inductors are connected to the piezoelectric actuators to add damping to one of the structural modes [Hagood and Crawley, J. Guid. Control Dyn. 14]. Modifications to the method have been proposed to enable a multi-modal damping technique, but at the cost of performance to any one mode [e.g., Wu, Proc. of 1998 SPIE Smart Structures and Materials Conference, Vol. 3327, pp. 159–168]. The objective of this talk is to discuss recent research into methods that help eliminate the disadvantages of passive resonant shunt damping. The methods to be discussed involve real-time switching of piezoelectric shunt circuits, and are based on the foundations laid by Larson [Ph.D. Dissertation, Georgia Institute of Technology, 1996], Clark [J. Intell. Mater. Syst. Struct. 11, 263–271], and Richard et al. [Proc. of 1999 SPIE Smart Structures and Materials Conference, Vol. 3672, pp. 104–111]. This talk will begin with the basic concepts of real-time switching, and will lead up to recent research that has demonstrated the use of the technique for multi-modal vibration suppression.
A nonlinear semipassive vibration damping using piezoelectric elements
Authors: Guyomar Daniel J. , Richard Claude L. , Audigier David S.
Published: October 2013

The proposed approach is based on a nonlinear filtering of the piezoelements output voltage and does not require any tuning to ensure damping performances over a large bandwidth. The output voltage is short circuited for a very brief period of time when the voltage reaches a maximum or a minimum. As a consequence of this filtering, the voltage gets distorted, amplified, and shifted in the time domain. It can be split in two signals, one is in phase with the vibration displacement and contributes to the potential energy while the second is in phase with the vibration speed and participates to the energy damping. The technique is semipassive since it requires a very low electrical power to drive the switch sequences. Experimental data show that the switching technique leads to better damping performances than purely passive methods whatever the natural damping of the structural material is. Furthermore, switching the piezovoltage through a small inductor leads to a voltage amplification and consequently to an increase of the damping. Damping capabilities range from 6 dB for an epoxy beam to nearly 20 dB for a steel beam even in the low frequency regime. Experimental results and theoretical interpretations will be presented.
Analytical solutions for characterizing tortuosity and characteristic lengths of porous materials using acoustical measurements: Indirect model
Authors: Olny Xavier , Tran Van Jerome , Panneton Raymond
Published: October 2013

Under the homogenization hypothesis, wave propagation in porous materials can be described using, for example, macroscopic dynamic density and bulk modulus. These functions depend on statistical geometrical parameters. Classically, five parameters are used: porosity, static air flow resistance, tortuosity, and viscous and thermal characteristic lengths. The three last parameters are generally difficult to measure with existing direct methods, for a wide range of materials. The proposed method is based on the measurement of dynamic density and compressibility, in order to separate viscous and thermal effects. With prior knowledge of airflow resistance and porosity, it is then possible to find analytical solutions for the missing parameters, fitting Johnson–Allard–Champoux model. In the same way, we also tackle the problem of the determination of static ‘‘thermal permeability’’ introduced to improve the description of thermal dissipation effect. Experimental results, obtained with various materials, using a Kundt’s tube, are presented to show the efficiency, and relative simplicity of the method. Moreover, the relevance of using a three parameters model for describing the bulk modulus is discussed.
On talker voice in language identification
Authors: Stockmal Verna , Bond Z. S.
Published: October 2013

Listeners find that discriminating between two languages is relatively easy compared to identifying new samples. Listener responses seem to be influenced by regional speech characteristics and talker voice quality. This study attempted to assess listener ability to match spoken samples of unknown languages when produced by male and female talkers. Samples from Arabic and Latvian, three provided by male talkers and three by female talkers, were assembled in a test using ABX format. Listeners matched the X language with either the A or B language, across same sex and different sex talkers. Overall, listeners performed at above chance levels, 60% correct. Listeners identified the two languages at approximately equal rates. They matched speech samples produced by males (75%) better than speech samples produced by females (53%). Listeners matched languages across speaker gender at 55% correct. These results suggest that speaker sex, as part of voice quality, is encoded with a representation of the unknown language.
The pronunciation of English sentences by Korean children and adults
Authors: Flege J. E. , Birdsong David , Bialystok Ellen , Mack Molly , Sung Hyekyung
Published: October 2013

This study examined English sentences produced by four groups of native Korean subjects (18 each) who differed according to mean age (children=12 years, adults=32 years) and length of residence in North America (means=3 vs 5 years). A delayed repetition technique was used to elicit English sentences at Time 1 and one year later at Time 2. Native English-speaking listeners used a 9-point scale to rate the sentences for overall degree of foreign accent. The ratings obtained for the native Korean (NK) subjects were converted to z-scores using the mean ratings and standard deviations obtained for sentences produced by control groups of Native children and adults. As expected, analyses of the standardized ratings revealed that the NK children produced the sentences with milder foreign accents than the NK adults did at both Time 1 and Time 2. Unexpectedly, the adult–child difference was larger at Time 2 than Time 1 because the NK children’s foreign accents diminished whereas the NK adults’ foreign accents grew significantly stronger from Time 1 to Time 2. Possible explanations for this are age-related differences in motivation, English input, or the strength of influence of Korean phonetic structures on the English sound system.
Discrimination of Thai tones by naive and experienced native English speakers
Authors: Wayland Ratree P. , Guion Susan G.
Published: October 2013

This study investigated the ability to discriminate between the low and the mid tones of Thai by two groups of native English speakers: naive and experienced. The naive group was comprised of eight native American English speakers who had no previous exposure to Thai or any tonal language. The eight native English speakers in the experienced group had been learning and had lived in Thailand for a varying amount of time. Eight native speakers of Thai participated as control subjects. Stimuli were eight minimal pairs. Three variables were manipulated: (1) experience with the target language: naive vs experienced; (2) syllable types: closed vs open; and (3) levels of processing: phonetic vs phonological implemented by using two different ISI (500 ms and 1500 ms). It was found that experience with the target language resulted in improved discrimination of the target language’s tonal contrast. Moreover, all three groups of subjects performed better with closed syllables than open syllables. Contrary to previous research, the effect of ISI was found only for the most difficult contrasts and only for the experienced group. An explanation for the findings and implication for futher research will be offered and discussed.
Towed array history, 1917–2001
Authors: Lemon Stanley G.
Published: October 2013

Towed array research and developments over 84 years is examined from the inventions of Dr. Harvey C. Hayes during World War I to multiple arrays towed by oil exploration ships of today. A time line of the effort over the 80 years is presented with a discussion of the driving technical and military reasons for system evolutions. Military surface ship towed arrays from Hayes electric eel to the AN/SQR-14, SURTASS, ETAS, and APL/Johns Hopkins dual towed arrays are reviewed. Submarine array programs are described from the early 1960s through today. A prediction of future towed array developments is presented.