bluuebunny/crossref_metadata_2025 huggingface dataset:
The role of spectral factors in the occurrence of the gap transfer illusion
Authors: Kuroda Tsuyoshi , Nakajima Yoshitaka , Tsunashima Shimpei , Yasutake Tatsuro , Mitsuba Shiro
Published: October 2013

When a long ascending frequency glide with a gap at the temporal middle and a short continuous descending glide cross at their central position, observers report that the gap belongs to the short glide in spite of the physical position of the gap. This phenomenon is called the gap transfer illusion. We employed multiple-component crossing patterns and systematically varied the spectral slopes or formants of glides. In patterns that contained glides consisting of three harmonic components, the gap transfer illusion occurred when the crossing glides had equal spectral slopes, while the illusion was inhibited when the crossing glides had different spectral slopes. In patterns that contained 25 harmonic components representing formants of Japanese vowels, the gap transfer illusion took place only when the crossing glides had the same formants. Although Nakajima et al. (2000) explained the gap transfer illusion in terms of the proximity of onsets and terminations, the present results show that it is also important to consider the role of spectral factors. [Work supported by JSPS and 21st Century COE, Kyushu University.]
The command-response model for F0 contours and its application to phonetics and phonology of tone
Authors: Fujisaki Hiroya
Published: October 2013

The command-response model for the F0 contours, first proposed for nontone languages such an Japanese and English, has been extended by the author and his co-workers to cover F0 contour of tone languages and has been proved to be applicable to several dialects of Chinese including Mandarin, Cantonese, and Shanghainese, as well as other tone languages including Thai and Vietnamese. Namely, it allows one to approximate F0 contours of these languages/dialects with a very high accuracy using positive and negative tone commands. This paper first presents some results of analysis of F0 contours of these languages/dialects to demonstrate that the model can capture the acoustic-phonetic characteristics of their tones quite well. It then shows that coarse qualitative distinctions in the pattern of tone commands serve to distinguish tones within a language/dialect, as well as to distinguish tone systems of different languages/dialects. Thus the model proves to be a useful tool, not only in the studies of phonetic features of tones, but also in the studies of phonological structure of tone systems.
The acoustic correlates of stress in compounds
Authors: Adams Tuuli M.
Published: October 2013

The acoustic correlates of linguistic stress are traditionally considered to be intensity, pitch, and segment duration. The current study investigates the correlates of stress in English compounds such as ‘‘blackboard’’ by comparing their interaction with different intonational patterns. This study includes an experiment in which participants pronounced compounds in controlled prosodic and intonational environments and various changes in intensity (dB), pitch (Hz), and duration were recorded for each vowel. Compounds in English are generally said to have a definitive stress pattern that distinguishes them from phrasal sequences; for example, ‘‘greenhouse’’ (for plants) would be stressed only on the first syllable, and its counterpart ‘‘green house’’ (house that is green) has a more even stress pattern. However, until now there has been no empirical study that successfully identifies the acoustic cues that distinguish compounds from phrasal structures. Results of this experiment show that intensity, pitch, and duration participate in a form of cue trading to signify stress in different contexts. This experiment provides new evidence that the acoustic correlates of stress interact with different intonational environments in a consistent way that distinguishes compounds from phrases.
Second occurrence focus and the acoustics of prominence
Authors: Howell Jonathan
Published: October 2013

Partee (1991) challenged the significance of the observation that certain adverbs (e.g., only) reliably associate with phonologically prominent words to truth-conditional effect, noting second occurrence (i.e., repeated or given) focus (SOF) appears to lack a phonological realization. Rooth (1996), Bartels (2004), Beaver et al. (2004), Jaeger (2004), and Fry and Ishihara (2005) argued that, while not intonationally prominent, an SOF word can be marked by increased duration and/or increased rms intensity. An acoustic study of verb-noun homophone pairs is reported. Three sophisticated speakers uttered five repetitions of the targets, embedded in discourses, in first occurrence (FOF), SOF, and unfocused (NF) contexts, both pre- and postnuclearly. Syntagmatic comparison shows a durational effect for focus and ordering in the FOF/NF case, yet only an effect of ordering in SOF/NF; similarly, an effect of spectral balance (H1-H3) was found for FOF/NF but none for SOF/NF. Results suggest, contrary to previous assumptions, that simple durational difference is not a straightforward cue of prominence. Sluijter and van Heuven (1996) and Campbell and Beckman (1997) have shown spectral balance calculated by various means is a cue to focus; its absence in these data raises doubts that prominence is reliably produced (let alone perceived) on SOF words.
Sound education in urban environment
Authors: Tsuchida Yoshio
Published: October 2013

There are various viewpoints for education and there are various levels for a sound. We have some cases to teach a ‘‘sound.’’ A certain person teaches a sound as music, another person teaches it as physics. But education is not teaching. It is more important to feel and enjoy. Then, one’s sensitivity for a sound will be nourished. Experimental listening is called Sound Education based on soundscape. The case of a workshop on Sound Education is addressed in this paper. Sound Education finally aims at improvement of our environment. Many activities of Sound Education were devised by R. Murray Schafer [A Sound Education, 100 Exercises in Listening and Soundmaking, Arcana Editions]. Some activities were devised by Joseph Bharat Cornell as Nature Game [Sharing Nature With Children (Sharing Nature Series), Dawn Publications.] Sound Education has several aims. It differs according to the group that sponsors it. Here are the workshops executed by me. One aim for environmental preservation, and another one aim for community design.
Effect of burst-modulated sinusoidal dither waveforms on the effectiveness of dither forces to eliminate friction-induced oscillations
Authors: Badertscher Jeff W. , Cunefare Kenneth A. , Ferri Al A. , Michaux Michael M.
Published: October 2013

This paper examines the effectiveness of high-frequency dither-cancellation techniques using burst-modulated signals. The classic single-degree-of-freedom (SDOF), mass on moving belt, model is analyzed using the method of averaging and numerical integration of system dynamics. Two different friction laws are used to model the contact point, both exhibiting a negative friction coefficient-velocity relationship. Recent results [M. Michaux and A. Ferri, Proceedings of IDETC05 Paper no. DETC2005-84491 (2005)] found that sinusoidal dither forces could stabilize or destabilize such a system, depending on the system and frictional characteristics as well as the dither signal amplitude and frequency. This paper extends this previous analysis to burst-modulated sinusoidal dither signals. Earlier experimental research has shown burst-modulated dither signals to be an effective method of suppressing automotive brake squeal. It is found that, for a given amplitude and frequency, burst signals are also capable of stabilizing the system. Individual effects of each burst parameter on system stability are established.
Comparative study of perception of Japanese dialectal accent in children
Authors: Shirose Ayako , Kakehi Kazuhiko
Published: October 2013

This paper reports the acquisition course of perception of Japanese dialectal accent. We carried out an experiment of accent perception for native speakers of Japanese dialects, namely, Tokyo dialect (Standard Japanese) and Kagoshima dialect. Then we compared results between the two dialects. Experimental stimuli were real words with several types of synthesized accent patterns. Results revealed that the Kagoshima children and adults accepted both the native Kagoshima accent and the Tokyo accent, whereas the Tokyo children and adults accepted only the native Tokyo accent. These results suggest that the Kagoshima speakers had insensitivity to the perception of accent. In our previous studies, the Kagoshima children showed different characteristics of accent production from the Tokyo and Kyoto children: the Tokyo and Kyoto children acquired their native accent correctly, whereas the Kagoshima children failed to produce it. [Shirose et al., J. Cognit. Sci. 2(1), 33–44 (2001)]. The insensitivity of accent perception of both Kagoshima adults and children might be one factor contributing to results of accent production. These intriguing parallels of accent perception and production will be discussed inclusively.
Cognitive influences in infant speech perception
Authors: Conboy Barbara T. , Sommerville Jessica , Randles Josie , Deskines Caryn , Kuhl Patricia K.
Published: October 2013

Speech perception during the first year reflects increasing attunement to native language phonetic features, but the mechanisms underlying this development are not well understood. Reductions in non-native phonetic discrimination have been linked to improvement in native phonetic discrimination and later vocabulary growth (Kuhl et al., 2005), and performance on nonlinguistic tasks (Lalonde and Werker, 1995). The present study examined links between native and non-native voice onset time discrimination, receptive vocabulary (MacArthur-Bates CDI), and cognitive control abilities at 11 months. Infants (n=18) completed a double-target conditioned head turn task and two nonlinguistic tasks requiring attentional control and resistance to irrelevant cues (means-end and detour-reaching object retrieval). Infants with CDI scores above the median showed higher native discrimination scores, t(16)=2.15 0.05, but no group differences for the nonnative contrast. Infants with scores above the median on either cognitive task showed worse discrimination of the nonnative contrast (means-end, t(15)=2.27p0.04; detour-reaching, t(15)=3.49, p.01), but no group differences for the native contrast. These results suggest that cognitive control plays a role in infants’ ability to ignore acoustic cues that are irrelevant to their native languages phonemic categories. [Work supported by NICHD (Grant HD37954) and a UW NSF Science of Learning Center Grant (LIFE).]
Distinguishing voices: Are infants’ abilities affected by age or language experience?
Authors: Sundara Megha , Kuhl Patricia K.
Published: October 2013

We know surprisingly little about infants’ ability to encode voice information when listening to speech. Research with infants has focused on their ability to recognize their mother or fathers’ voice and distinguish it from an unfamiliar voice. However, communication consists of interactions with unfamiliar speakers as well. Adults are well able to extract information about the speakers gender, size, age, dialect, etc., even from short samples of an unfamiliar persons voice. In experiment 1, we tested 4- and 12-month-old English-learning infants on their ability to distinguish an unfamiliar male and female voice when listening to Hindi monosyllabic words. Results indicate that the 12-, but not the 4-month-olds can distinguish an unfamiliar male and female voice when listening to Hindi words. In experiment 2, we tested 4- and 12-month-old English-learning infants on their ability to distinguish an unfamiliar male and female voice when listening to English monosyllabic words. Now, both 12- and 4-month-olds are able to distinguish the voices. Together, these results indicate that infants’ ability to distinguish voices is affected by age as well as their language experience. We are now testing adults to determine if the effects of language experience on the ability to distinguish voices persists even after development.
Theory and method in automatic evaluation of spoken language proficiency
Authors: Bernstein Jared
Published: October 2013

Several kinds of theory (linguistic, psycholinguistic, and sociolinguistic) have been used to justify different methods of language proficiency assessment or to explain patterns of performance observed in proficiency tests. After reviewing some traditional methods of testing spoken language performance (e.g., structured interviews, listening comprehension, word identification in noise, and elicited imitation), the assumptions underlying these methods are compared. When the practicality, reliability, and validity of assessment methods are evaluated, differences in the associated theories become clearer. A low perplexity testing method is presented that scores a set of context-free speech-interactive tasks using speech recognition and analysis techniques to measure aspects of the linguistic and phonetic content of spoken responses. Data from several experiments comparing interview testing with low-perplexity testing on populations of second language speakers (of English, Spanish, and Dutch) suggest that nearly all the information that expert judges extract about levels of communicative competence can be accounted for in terms of automaticity of spoken language processing. Furthermore, measurement techniques that are feasible in low-perplexity testing provide much greater precision in the spoken language proficiency reporting scale than was previously feasible, opening up a range of new applications.
Precise position measurement of objects using a coded acoustic signal in an indoor environment
Authors: Nishihara Kuramitsu , Yamaguchi Tadashi , Hachiya Hiroyuki
Published: October 2013

Acoustic sensing in the air might serve as an effective technique for acquiring information about an object, such as its distance, shape, and surface configuration. Nevertheless, it is difficult to identify acoustic waves clearly because many mirror reflections or environmental noises exist in the air. In addition, acoustic measurement requires a longer time than measurements using lasers or electromagnetic waves because sound propagates much more slowly than light. For this study, we attempted highly precise position measurement using an M-sequence signal in an indoor environment. The degrees of M-sequence were 7–14; the transmitting frequency was 25 kHz. A signal reflected from various angles was received by moving the receiving unit, which contained three microphones. Measurement results with the M-sequence signal showed that the signal-to-noise ratio (S/N) improved 20–40 dB compared with a burst wave. Furthermore, the distance resolution improved when the M-sequence degree was changed. The distance and the angle of multiple objects installed indoors or in walls were detectable as a result of construction of a two-dimensional plane based on received signals.
Spatio-temporal invariants of the time-reversal operator
Authors: Robert Jean-Luc , Fink Mathias
Published: October 2013

The DORT method gives the spatial invariants of the monochromatic time-reversal operator, at a given temporal frequency. It allows the monochromatic Green functions of point scatterers to be extracted from a data set. However, in acoustics, broadband signals are usually used and thus, it is often interesting to have the invariants in the time domain, or to know the evolution of the monochromatic Green function with frequency. It is not always easy, especially in complex media, because the sequence of the DORT eigenvectors (the invariants) changes as a function of the frequency. Thus, an invariant corresponding to a given scatterer may be the first at low frequencies, but the second at higher frequencies. Therefore, it is difficult to reconstruct a spatio-temporal Green function from monochromatic invariants. We introduce here the spatio-temporal time-reversal operator as a linear operator described by a fourth-order tensor, and deduce a method to obtain its spatio-temporal invariants. The method reduces to a more straightforward matrix decomposition, and thus can be easily implemented. Properties of the invariants are described, and applications to focusing in complex media are presented.
Semiautomatic inverse filtering of vowels with differing degrees of nasality
Authors: Tooher Michelle , Gobl Christer
Published: October 2013

Most approaches to inverse filtering are either manual or automatic. Manual methods, while providing high-quality fine-grained analysis, require user expertise and manpower to analyze even small amounts of data. Automatic methods allow large-scale analysis, but lack the precision required to capture non-neutral voice dynamics as present in, for example, expressive speech. This paper reports on development of a semiautomatic inverse-filtering toolkit, focusing on an adaptive nasal/non-nasal vowel inverse filter. For vowels with differing degrees of nasality (from not nasalized to heavily nasalized), each frame is analyzed using machine learning techniques to assess the degree of nasality. Based on this assessment, the inverse filter is adapted according to acoustic theory, to best analyze each frame of the data, employing either all-pole or pole-zero analysis where necessary. User intervention is possible, in both specifying analysis frames and fine-tuning of pole and zero number and location. Training data for machine learning are selected on the basis of auditory and acoustic analysis. This semiautomatic approach allows analysis of large-scale corpora while retaining the robustness and fine-grained analysis of manual methods. Results from analysing both non-nasalized and nasalized vowels from a corpus of Irish are presented. [This work is funded by the IRCSET Embark Initiative.]
Ambiguity analysis of matched-field source localization under spatially correlated noise field
Authors: Xu Wen
Published: October 2013

Matched-field processing (MFP) concerns estimation of source location by exploiting full-wave modeling of acoustic waveguide propagation. The ambiguity output in MFP relies on the correlation between the signal field at the true source position and the signal field at each scanning source position. This correlation often shows a multimodal structure due to the nonlinear parameter dependence: in addition to a mainlobe around the true source position, there are unpredictable prominent high sidelobes elsewhere. For a well-conditioned problem, in the absence of noise the peak output is guaranteed to occur at the true source position. In the presence of noise, we will probably have some peak outputs around the sidelobes, introducing a large localization error. Therefore, output ambiguity structure is a very important factor in the development of any matched-field algorithm operated in low-SNR scenarios. To analyze the ambiguity behavior, a quantitative approach for error analysis has previously been developed in the context of the maximum likelihood estimate (MLE) with spatially white noise. In this talk, generalization to including spatially correlated noise field introduced by discrete interferences and/or surface distribution sources is discussed.
Modeling the effect of multipath on high-resolution synthetic aperture sonar image statistics
Authors: Johnson Shawn F. , Lyons Anthony P. , Abraham Douglas A. , Williams Kevin
Published: October 2013

Recent analysis of Synthetic Aperture Sonar data collected during the Office of Naval Research sponsored SAX04 experiment in the Gulf of Mexico during October 2004 has shown that the processed image has varying amplitude statistics as a function of range. Since synthetic-aperture sonar maintains a constant range-independent resolution cell size, unlike traditional sonar systems, one would expect that the statistics of the image would be based solely on the seafloor properties. The K-distribution shape parameter was estimated as a function of range and increases at discrete ranges corresponding to the onset of multi-path propagation (low shape parameter values represent a heavy tailed distribution). At these points of increase in shape parameter, scattered returns arising from additional propagation paths arrive with the result that two or more parts of the seafloor are contributing to the same sonar resolution cell, which leads to a larger K-distribution shape parameter than would be expected when only the direct path contributes. Model results using the sonar system geometry and interface scattering theory will be used to illustrate this effect. [Work supported by ONR Codes 321 and 333. First author supported by a National Defense Science and Engineering Graduate Fellowship.]
Laboratory study of high-frequency scattering from water-saturated granular sediments
Authors: Ivakin Anatoliy , Sessarego Jean-Pierre , Guillermin Regine
Published: October 2013

Sound backscattering and reflection from water-saturated granular sediments at frequencies from 200 kHz to 7 MHz were studied in controlled laboratory conditions. Two kinds of well-sorted sandy sediments, fine and coarse sands, and two kinds of glass beads with corresponding sizes were chosen for the study. The two types of sand had narrow log-normal size distributions of particles with the mean diameters 0.25 and 1.5 mm for fine and coarse sand, respectively. The sediments were degassed and their surface was flattened carefully. In these conditions, the grain-scattering mechanism can be considered as a dominating factor controlling incoherent component of the field scattered from the sediment. Frequency dependencies for the backscattering strength at various grazing angles and the reflection coefficient at normal incidence were measured. The effects related to the sediment grain size are analyzed and their possible applications to remote sensing of marine sediments are discussed. [Work supported by ONR and CNRS.]
Sound absorption characteristics of a honeycomb-backed microperforated panel (MPP) absorber
Authors: Sakagami Kimihiro , Nakajima Kosuke , Morimoto Masayuki , Yairi Motoki , Minemura Atsuo
Published: October 2013

Microperforated panels (MPPs) are typically made of a thin metal or plastic panel and are often unsuitable for an interior finish because thin limp panels do not have enough strength. In particular, an interior finish of room walls requires appropriate strength. In order to solve this problem, a honeycomb structure is attached behind MPPs to stiffen the construction. Thus, it is possible to stiffen an MPP without increasing its thickness, which is important to keep MPPs at their best absorption performance. Furthermore, a honeycomb can increase MPPs’ absorption coefficient in a similar way as a porous layer backed by a honeycomb. In this study, an experiment was performed to gain insight into the acoustical effect of a honeycomb structure behind MPPs and a simple theoretical model to interpret the experimental effects is presented. The experimental results show that the honeycomb affects the absorption characteristics of MPPs: the absorption peak increases and shifts to lower frequencies. This effect becomes more significant as the thickness of the honeycomb increases. The results from the theoretical model show the same tendency. This is attributed to the fact that the honeycomb makes a similar condition to local reaction in the back cavity.
Quantitative evaluation of characteristics of an auditorium shape in horizontal cross section
Authors: Hoshi Kazuma , Hanyu Toshiki , Sekiguchi Katsuaki
Published: October 2013

The room shape of an auditorium is often classified as a ‘‘shoe-box type,’’ ‘‘fan-shape type,’’ ‘‘round type,’’ ‘‘surrounding (vineyard) type,’’ or other shape. The surface diffusivity index (SDI), which is an evaluation index of surface irregularities, has been proposed by Haan and Fricke, but the SDI is based on visual surveys of photographs or drawings [Appl. Acoust. 51(1), 53–69 (1997)]. Clarification of the relationship between an auditorium’s shape and its acoustic characteristics would be simplified if the auditorium shape were understood objectively rather than subjectively. Therefore, in this study, we tried to objectively obtain characteristics of a room shape in horizontal cross section using an elliptic Fourier descriptor, which can detect the periodicity of a two-dimensional contour. Room shapes were expressed as spatial frequency characteristics calculated from their Fourier descriptors. Based on them, we can understand the amplitude of irregularities and their scale. In addition, by using principal component analysis, it is possible to understand, quantitatively, the main differences of room shapes in horizontal cross section.
Introduction of auditory filter into evaluation of characteristics of small enclosures
Authors: Shimada Saori , Suzuki Hisaharu , Omoto Akira
Published: October 2013

Physical measures, which are useful for evaluating small enclosures such as studios and car cabins, are examined. Especially, auditory filtering using Zwicker’s loudness model is introduced for calculating the measures. Most current measures used in evaluating the large sound field such as concert halls are based on measured or calculated impulse responses. Instead, the music signals of some genres are used in this study. Using the auditory model, the amount of information can be reduced, even when complicated signals are used as stimuli: the signals are convertible to information in 25-bark bands. Correspondence of such physical quantities and subjective judgments are considered from the viewpoint of sound-field similarity. Subjects are asked to judge the similarity of two successive sounds whose frequency characteristics are modified artificially using the equalizer. The analysis results obtained using the auditory filter were compared with the subjective evaluation. Results indicate that analysis using the auditory filter can yield fairly good correspondence with the subjective evaluation, especially in a middle frequency band. Regarding the low-frequency band, the degree of correspondence depends on the musical genre. Expanding the effective frequency bands is an important theme of our ongoing research.
Time-sharing acoustic stream analysis to recognize multiple objects in space by echolocating bats
Authors: Riquimaroux Hiroshi
Published: October 2013

Echolocating bats need to catch prey while they have to avoid obstacles and escape from their predators at the same time. Therefore, they have to perceive precise information about multiple objects from their surrounding space while they are flying. However, there are strong restrictions in the system; the neural system is too slow to perform real-time processing, and the bat has only one sound source emission and only two ears to receive echoes. The purpose of the present study was to investigate how echolocating bats perceive important information about different objects in space during their flight by analyzing acoustic characteristics of emitted pulses and returning echoes. Pulses and echoes of CF-FM and FM bats were recorded by a wireless telemetry microphone system (Telemike) on their head. In order to reconstruct their flight trajectory, a dual high speech video camera system was used. Obtained data showed multiple streams for temporal change in echo delay, Doppler-shift compensation, and echo intensity compensation. The present findings indicate that echolocating bats may adopt a time-sharing analysis, which is practically fast enough to recognize multiple objects in space during flight. [Work partly supported by the Innovative Cluster Creation Project promoted by MEXT and by a grant to RCAST at Doshisha University from MEXT.]
Echolocating bats control the directionality of sonar signals to sort out the auditory scene
Authors: Surlykke Annemarie , Moss Cynthia F.
Published: October 2013

A primary function of hearing is the localization and identification of sound sources in the auditory scene. Results from humans and other animals have shown that spectral and temporal features, as well as source direction, are important parameters used for organizing acoustic sources. Echolocating bats control the features of vocalizations to shape information carried by echoes from auditory objects. Our research on big brown bats, Eptesicus fuscus, suggests the importance of vocal control for scene analysis: When a bat performs a dual task, requiring both obstacle avoidance and prey capture, it solves the task sequentially, adapting the temporal pattern of its cries to produce sonar strobe groups, strobing first the obstacle and then the prey. Directional control over the sonar beam pattern confirms this sequential processing: The bat alternates fixation of the beam axis on the obstacle and prey. The importance of directionality is corroborated by studies in another bat, Trachops cirrhosus, where behavior and morphology optimize sonar directionality for different tasks. When hanging from a perch T. cirrhosus emits sounds for general orientation with limited demands for directionality. When hunting, prey localization necessitates high directionality associated with sound emission through the nostrils surrounded by the moveable nose leaf.
Comparisons of song perception in male and female songbirds utilizing cardiac responses
Authors: Ikebuchi Maki , Okanoya Kazuo
Published: October 2013

In most songbirds, males sing to defend territories and to attract females, and females evaluate the quality of males based on songs. This should lead to differences in song perception between the sexes, but it has been difficult to devise an assay that can evaluate song perception without introducing motivational bias. For example, the copulation solicitation assay is useful for evaluating female perception of songs, but this assay cannot be utilized for males. We found that birds react to a sound of interest by increasing the heart rate, and this response was used to compare song perception in Bengalese and zebra finches. We used conspecific and hetero-specific songs as stimuli and these were presented in a planned order. In Bengalese finches, only females responded to the changes in conspecific song repertoire. In zebra finches, both males and females reacted to such changes. Bengalese finches are a domesticated species and males do not establish breeding territories. Conversely, male zebra finches establish a small territory when breeding. In both species, songs are an important trait by which females select males. These ecological differences concur with the species and sex differences in the cardiac response. [Work supported by JSPS and JST.]
Underwater noise and zones of masking with respect to hopper dredging and manatees in the St. Johns River in Jacksonville, FL
Authors: Gerstein Edmund R. , Blue Joseph E. , Pinto Gerard F. , Barr Seth
Published: October 2013

Underwater noise radiating from dredging can effectively obscure or mask biological and other important sounds. This study recorded underwater acoustic characteristics of hopper dredging in the St. Johns River, Jacksonville, FL, to evaluate noise impacts in the waterway with respect to the endangered West Indian manatee. Of particular interest was the extent and range that dredging noise may mask the sounds of approaching commercial and recreational vessels. Vertical hydrophone arrays and a multi-channel PC-based recording system were used to measure dredging noise at various distances. Ambient noise surveys, active propagation of calibrated sources, and controlled boat noise measurements were conducted along the waterway. These data were integrated with behavioral hearing data to estimate zones of masking surrounding dredging. Three discernable noise sources that masked boat noise were (1) cavitation from dredge propellers, (2) draghead vacuuming, and (3) noise from submerged slurry pipelines. Sustained high ambient noise levels from dredging can significantly increase the risk of manatee-boat collisions by masking the sounds of approaching vessels over large radii (up to 2.5 miles). Mitigations suggested include ship quieting, reducing propeller cavitation, insulating or elevating slurry pipelines, and minimizing transects to pump out stations. [Work funded by the City of Jacksonville Waterways Commission.]
Ultrasound stimulated vibroacoustography and vibrometry for measuring tissue properties
Authors: Greenleaf James F. , Shigao Chen
Published: October 2013

Ultrasound stimulated vibroacoustography has been used to image tissues using the acoustic response of tissues to localized harmonic radiation pressure. The method provides high resolution and high dynamic range images of tissues. The parameter being imaged is a complex combination of scattering, attenuation, and nonlinearity. However, specific use of harmonic or pulsed radiation pressure and subsequent measurement of the tissue response can be used to measure fundamental material properties of tissue. The measurement of shear wave dispersion can be used to estimate elastic shear moduli of tissue. Fundamental model free properties such as elastic storage and loss constants can also be measured. Ultrasound radiation pressure is used to induce free propagating shear waves. The measurable properties of the shear waves are sensitive to only the material properties of the tissue under certain circumstances. Shear moduli in tissue are measured with high accuracy and precision given appropriate models of wave propagation within the geometry of the tissue. Careful use of shear wave propagation and subsequent measurements can provide fundamental measurements of tissue mechanical properties if models are accurate.
Evaluation of liver fibrosis using the acoustic characteristic and statistical signal processing of echo
Authors: Yamaguchi Tadashi , Hachiya Hiroyuki
Published: October 2013

We evaluate the relationship between changes in acoustic properties accompanying change of a tissue and echo information using human liver specimens in vitro. The rf echo signals of a 20-mm-thick human liver specimen (200 frames) were acquired continuously using a probe using 2.0-MHz transmission frequency and 4.0-MHz receiving frequency. The signals were processed using statistical signal processing to extract abnormal echo information. After ultrasonic scanning, 50×50-mm2 sections were cut from the liver specimens for microscopy and acoustic microscope measurement. The liver fiber structure was extracted from 20 4-μm-thick Masson’s trichrome stained microsections. The speed of sound and the attenuation of 2-mm-thick cut specimens were measured using 25-MHz ultrasound. Results of comparing 3-D structures of abnormal information extracted from echo signals and the 3D fiber structures extracted from microsections showed that both structures were clearly in agreement. Furthermore, it was confirmed that the speed of sound of fibers measured using an acoustic microscope was faster than that for normal liver tissue.
The importance of cells and arrangement of cells to ultrasound backscatter
Authors: Oelze Michael
Published: October 2013

The early work of Professor Floyd Dunn identified the importance of collagen structures and globular protein to ultrasound backscatter [Fields and Dunn, J. Acoust. Soc. Am. 54, 809–812 (1973)]. Recent advances in tissue characterization through ultrasound backscatter have again focused on determination of the underlying microstructures responsible for scattering. The focus of new models for ultrasound backscattering has been on the contribution of cells. Specifically, new models have sought to address the ultrasound backscatter from single cells and from different arrangement of cells. The importance of cells and their arrangements in tissues to classification of disease, i.e., cancer, through ultrasound was also noted by Professor Dunn [Saijo et al., Ultrasound Med. Biol. 17, 709–714 (1991)]. In the present work, the construction of a new cell model of scattering was made from measurements of acoustic properties of cells. Different arrangements of cells, i.e., uniform distribution of cells and cells arranged in groups, were examined in simulation using the new cell model. Finally, the new cell model and simulations were compared to ultrasound backscatter measurements from different kinds of cancer. The arrangement of cells within a cancerous region was observed to be significant for diagnostically differentiating between cancers using ultrasound backscatter.
Reduction of cavitation threshold by second-harmonic superimposition and its potential application to sonodynamic therapy
Authors: Umemura Shin-ichiro , Kawabata Ken-ichi , Sasaki Kazuaki , Sugita Nami
Published: October 2013

Acoustic cavitation is known to induce bioeffects under certain conditions. Cavitation threshold, the acoustic strength needed to induce such cavitation, in various biological systems was intensively studied by Dr. Dunn’s group. Their results have significantly contributed to the safety concerns of diagnostic and therapeutic uses of ultrasound. Acoustic cavitation is also known to be the primary cause of sonochemical reactions. Chemical compounds such as hematoporphyrin were found to be activated by acoustic cavitation and induce in vitro and in vivo antitumor effects thereby. In order to make use of this effect in therapeutic practice, cavitation must be generated reproducibly without depending on standing waves at acoustic power inducing no significant heating. It was also found that the cavitation threshold can be significantly reduced by superimposing the second harmonic onto the fundamental in insonation. The threshold was reduced by more than an order of magnitude, especially in the presence of a certain compound such as Rose Bengal (RB). An amphiphilic derivatives of RB, selectively accumulating in tumor tissues, was synthesized and used in combination with focused ultrasound with second-harmonic superimposition to treat experimental murine tumors, resulting in significant reduction of their sizes. A hypothesis on the mechanism behind this will also be discussed.
Exposure estimation and dosimetry for ultrasound thermal therapies
Authors: ter Haar Gail
Published: October 2013

The success of ultrasound thermal therapies relies on achieving a predetermined temperature rise in a specified tissue volume. If these treatments are to be used in clinical application, it is important to be able to estimate the required exposure parameters and to undertake appropriate dosimetry. Floyd Dunn and his co-workers were pioneers in this field for high-intensity focused ultrasound (HIFU). In early papers they addressed the problem of ultrasonic dosage and dose rate. This remains a topic of active discussion. A major part of these investigations relies on accurate temperature measurement in ultrasonic fields in tissue. The viscous heating artifact that occurs when thermocouples are used was first recognized by the Illinois group in the 1950s. Effective ways of estimating this component of the measured temperature rise are still being sought. Investigation of the dependence of HIFU lesion size on in situ ultrasound intensity, exposure time, and frequency will be described, and the importance of acoustic cavitation and boiling for the definition of an ultrasound dose parameter will be discussed.
Numerical analysis of temperature in tissue radiated by focused ultrasound
Authors: Endoh Nobuyuki , Tsuchiya Takenobu
Published: October 2013

Higher power ultrasonic equipment is recently extending not only into therapy but also into diagnosis because newer diagnostic systems such as harmonic imaging require higher ultrasound than conventional imaging systems. It is very important to assure the safety in the use of diagnostic equipment, especially for prevention of burns caused by radiation from ultrasonic imaging systems. In this two-dimensional FDTD-HCE study, temperature increases in tissue were simulated at the focal point of a phased array transducer like a common B-mode imagine system. A center frequency of radiated pulse is assumed to be 2.5 MHz and ISTPA=0.72 W/cm2. Results show that the temperature in tissue at the focal point is proportional to the pulse repetition frequency (PRF) when the PRF is changed from 100 to 400 kHz. It is also shown that common equipment is safe because the maximum temperature rise in tissue is only 0.0004°C at the focal point of the transducer even though the PRF is 400 kHz. [Work supported by High-Tech Research Center Project from the Ministry of ECSST.]
Digital hearing aids: A brief history
Authors: Levitt Harry
Published: October 2013

The journey leading to the development of digital hearing aids began at Bell Laboratories in the mid 1960s when a digital computer was used to simulate an experimental high gain telephone with frequency shaping for people with hearing loss. The simulation was off-line at about 100 times real time. About 20 years later, a digital master hearing aid operating in real time was developed using a high speed array processor. The development of wearable digital hearing aids followed soon after as high speed digital signal processing (DSP) chips were developed. The first wearable digital hearing aid was a body worn instrument and was not a commercial success. At this stage, digitally controlled analog hearing aids could be made small enough to be worn on the ear and these instruments were widely used by the end of the decade. During the 1990s, low-power DSP chips were developed that were small enough to allow for all-digital ear-worn hearing aids to be developed. The challenge today is to tap the tremendous potential of digital signal processing techniques within the power and size constraints of a practical, wearable instrument.
A new speech enhancement method for two-input two-output hearing aids
Authors: Li Junfeng , Sakamoto Shuichi , Suzuki Yo-iti , Hongo Satoshi
Published: October 2013

Human beings have the ability to pick up a speech signal in noisy environments, which is known as the cocktail-party effect. This phenomenon is, however, often degraded in persons with impaired hearing. Therefore, a good method to enhance this phenomenon must be very useful to improve speech intelligibility in ambient noise for hearing-impaired persons. In this lecture, multi-input two-output speech enhancement techniques are first summarized. Subsequently, a new two-input two-output speech enhancement method is proposed that is based on the frequency domain binaural model (FDBM) [Nakashima et al., Acoust. Sci. Technol. 24, 172–178 (2003)]. In the proposed method, the interaural differences are first calculated from the noisy observations and employed to determine the sound-source locations. A speech absence probability (SAP) estimator is then developed using knowledge of estimated source locations. It is further integrated into the original FDBM to improve the interference-suppression capability. Effectiveness and superiority of the proposed speech enhancement method with respect to the original FDBM are validated through comprehensive experiments in terms of objective and subjective speech-quality measures. [Work supported by Sendai Intelligent Knowledge Cluster.]
Utilizing the auditory systems processing capabilities in the development of methods to auto-steer signal processing strategies in hearing instruments
Authors: Dittberner Andrew B. , Bondy Jeff , Coughlin Maureen , Whitmer Bill , Dominguez Melissa
Published: October 2013

Hearing instruments have evolved in both the hardware and signal processing domains, providing more computational capabilities and sophisticated sound processing strategies. Static or adaptive methods are employed by these sound processing strategies to achieve a desired output. Due to the adaptive nature of some of these processing strategies and the continual increase in the number of parameters that can interact negatively with each other, there is a need to provide some method of global control over the parameter states. In addition, due to individual needs of the end user, parameter states in a hearing instrument may not satisfy the broad range of acoustic environments that a listener encounters, so identification of particular acoustic environment characteristics is also important. The purpose of this work is to introduce and discuss these problems and some of the current research efforts addressing environmental classification and global control of a hearing instrument’s parameter states. The scope of this discussion will be on strategies employed to meet the needs of the end user in reference to the acoustic environment they identify as having a negative impact on their auditory experience and the desired performance they expect from the hearing instrument in that environment.
A study of signal-to-noise ratio conditions for measuring the intelligibility of hearing aids in noisy live environments
Authors: Arai Yasuhiko , Yoshizumi Yoshiyuki
Published: October 2013

The authors studied a method to measure the intelligibility of directional hearing aids using a dummy head in noisy environments. The method is based on ISO recommendations (ISO 8353-2, ISO 8353-3). However, it is sometimes difficult to meet the quasi-free-field conditions. Therefore, a method for approximate measurement of intelligibility, by which the equivalent results are obtained in noisy live environments, is recommended. To simulate noisy environments, a multitalker (MULT) noise or a simulated multitalker noise was simultaneously and noncorrelatively put out through four loudspeakers in the live testing room. The MULT noise was composed of eight voices (four males and four females). The experiment showed that mixed voices of eight to ten persons were adequate for interference of the test speech words. The 50% SNR specification, meaning signal-to-noise ratio that gives intelligibility of 50% correct, was introduced into the measurement system. Intelligibility was measured under various SNR conditions. It was expressed approximately by the sigmoid function of SNR. The 50% SNR was estimated from the experimental equation. The intelligibility was measured adequately in the range between 0% and 100% by using the 50% SNR specification. The system configuration, measuring conditions, and measured samples are discussed.
Perception-model-based sound quality prediction for hearing aids: Big effect, small effect, and the optimization of noise reduction algorithms
Authors: Kollmeier Birger , Huber Rainer , Rohdenburg Thomas , Hohmann Volker
Published: October 2013

Objective sound quality models can be used to optimize hearing aid algorithms and their respective parameter settings by numerically predicting the subjective sound quality assessed by humans without having to perform quality measurements with humans concurrently. The big effect prediction considers how the (individual) hearing loss affects the aided or unaided perceived sound quality. The small effect prediction aims at subtle differences at a given amplification level of the hearing aid in order to compare processing parameters. This talk will concentrate on benchmarking different types of objective prediction methods both for big effect and small effect conditions using a wide range of signals and processing conditions, respectively. In addition, the performance of PEMO—Q, an auditory-model-based quality prediction scheme developed at Oldenburg University will be discussed, which has been modified to include the effect of hearing impairment and the adaptation to different kinds of test data sets. We will also demonstrate the degree to which PEMO—Q is beneficial for optimizing noise reduction algorithms that are suitable for hearing aids. The objective measure helps to efficiently scan the multidimensional parameter space for an optimum parameter set of the noise reduction algorithm. [Work supported by BMBF and by HoerTech.]