bluuebunny/crossref_metadata_2025 huggingface dataset:
Landmine detection from acoustic/seismic signals using ordered weight-averaging operators
Authors: Gader Paul , Grandhi Roopnath , Hocaoglu Ali , Keller James , Wang Tsaipei
Published: September 2013

Acoustic/seismic systems demonstrate significant potential for reliable detection of landmines, particularly when coupled with ground-penetrating radar (GPR). Acoustic/seismic systems provide complementary information to GPR. One difficulty in developing detection algorithms for acoustic/seismic systems is that the frequency bands at which information occurs vary according to factors such as mine depth and mechanical properties of the soil, which are not known during mine detection. Ordered weighted-averaging (OWA) systems offer the potential for robust detection of landmines under different conditions because they naturally provide the capability to process the best collection of frequency bands in a fashion that can be optimized. In this paper, a processing framework is presented that combines OWA operators for feature analysis with decision making. This framework can be optimized using acoustic/seismic signals as well as acoustic/seismic signals combined with GPR. In the latter case, the optimization can help to provide the appropriate mechanism for combining the complementary information from both sensors. Results are presented using real data from both mines and other buried objects collected at an outdoor test site.
False alarms associated with the acoustic-to-seismic detection of buried land mines
Authors: Fokin Vladimir N. , Fokina Margarita S. , Sabatier James M.
Published: September 2013

Important problems in landmine detection include false alarms and clutter [high values of the acoustic-to-seismic transfer function (TF) in some frequency bands] that mimic the physics of a buried landmine. Many of these high values of the TF are due to the natural variability of the ground. In this paper both the space-frequency variability of the TF(f,x,y) and the connection with ground variability are discussed. The viscoelastic, layered model of the ground qualitatively explains high values of the TF both in certain spatial regions of the scan site and for some frequencies. This model and the measured spatial dependence of the TF(f,x,y) were used to model the space-frequency distribution of the acoustic-to-seismic transfer function. Comparison between the calculated and measured transfer functions in the space-frequency volume show that this model can satisfactorily explain variability of the acoustic-to-seismic transfer function. To investigate the spatial variability of ground parameters, the measured TF for a few sites was analyzed. This analysis reveals that different frequency modulation scales exist in the acoustic-to-seismic TF. These different frequency modulation scales are due to spatial dependencies of ground parameters and ground layering. [Work supported by ONR Grant N00014-02-1-0878.]
The oscillation of vapor bubbles
Authors: Prosperetti Andrea , Yin Zhizhong
Published: September 2013

Bob Apfel had so many interests that it is impossible—however fitting and desirable—to pay homage to his work as a whole. Some of his early studies were devoted to bubble nucleation at high superheats. In the first part of this paper a recent application of this phenomenon is described. Once a vapor bubble is generated, its subsequent oscillations (free and forced) present analogies and differences with those of a gas bubble: the second part of the paper focuses on this topic. [Work supported by NSF and NASA.]
Quantitative theoretical explanation of Apfel’s experimental phase diagrams for sonoluminescing bubbles
Authors: Lohse Detlef , Toegel Ruediger
Published: September 2013

Robert Apfel had an enormous impact on the research on single bubble sonoluminescence, the light emission of a single sound driven bubble [for a review, see Brenner et al., Rev. Mod. Phys. 74, 425 (2002)]. In 1996, at the ASA Meeting in Hawaii, he posed a challenge to the theoreticians in the field: Make experimentally testable predictions on single bubble sonoluminescence. Apfel collected the predictions and gave a wonderful review talk on the state of the field. Later, he several times came back to that list, comparing the predictions with latest experimental results. Our own predictions those days referred to the phase diagrams of single bubble sonoluminescence. Later Apfel himself, together with Ketterling, measured those phase diagrams experimentally [J. A. Ketterling and R. E. Apfel, Phys. Rev. Lett. 81, 4991 (1998)]. Though qualitatively our 1996 predictions turned out to be correct, a full quantitative model could only be developed recently [R. Toegel and D. Lohse, J. Chem. Phys. 118, 1863–1875 (2003)]. In the presentation we will compare the model predictions with Apfel’s data.
Reflector geometry and pressure at the focus of a shock wave lithotripter
Authors: Iloreta Jonathan , Iyengar Prahallad , Szeri Andrew
Published: September 2013

It has been experimentally shown [Zhong, J. Acoust. Soc. Am. 113, 586–597 (2003)] that refinement of the geometry of the HM-3 lithotripter reflector suppressed the expansion of cavitation bubbles without compromising stone comminution. This effect has been attributed to a change in the rarefaction tail of the pressure wave near the second focus of the original reflector (F2). Following this idea, a numerical model of the reflection and steepening of a pressure wave from an axisymmetric lithotripter has been developed. The model is based on the Euler equations coupled with the Tait equation of state. Preliminary results of the pressure fields produced by numerous reflector shapes are presented. The results show the changes in the rarefaction tail of the pressure wave near F2 for different geometries, thus hinting at the possibility of optimization of the reflector shape for an ideal waveform. [Work supported by NSF.]
Reduction of variance in statistical estimates of spectra for use in correction of ultrasonic aberration
Authors: Astheimer Jeffrey P. , Pilkington Wayne C. , Waag Robert C.
Published: September 2013

Parameters of ultrasonic aberration can be obtained from power spectra of scattering when individual scattering measurements from which the spectra are estimated have a common aberration and the same nominal geometry. However, the scattering volumes are then confined to a small spatial region and use of finitely many overlapping volumes that result in a nonzero variance is necessary for the measurements. Assuming the scattering is from a spatially uncorrelated medium, the variance of the spectral estimates is expressed as the product of the variance for a single measurement and a reduction factor that depends on the amount of overlap between each volume pair. This factor describes the rate of convergence and the accuracy of the estimates as a function of the number and the overlap of the scattering volumes. Assuming further that the individual volumes are localized by a Gaussian window and that the centers of the volumes are located on orbits of an icosahedral rotation group, the variance is minimized by adjusting the weight and radius of each orbit. Numerical evaluations using orbits formed by icosahedral vertices, face centers, and edge midpoints show that a significant reduction of variance can be achieved from volumes in a confined region.
Effect of compressor design on auditory function and a psychoacoustic Turing test
Authors: Edwards Brent W.
Published: September 2013

Hearing aids incorporate multiband compression to compensate for the loudness recruitment that results from sensorineural hearing impairment. No consensus exists in the hearing-aid industry on the best compressor design or on what the design criteria should be. Differences exist in compressor time constants, number of bands, filter shapes, and fitting formula. Such design differences can result in different aided auditory ability as measured by psychoacoustic tests. This research investigated the effect of different compressor designs on fundamental psychoacoustic ability. Multiband compressors with different time constants and different analysis/synthesis filterbanks designs were simulated. Aided performance by the hearing impaired was calculated under each configuration for several psychoacoustic measures, including forward masking, loudness summation, and simultaneous off-frequency masking. Differences in aided performance in these tasks were found: some compressor designs produced aided psychoacoustic ability that resembled normal performance, while other compressor designs did not significantly alter performance from unaided impairment. A psychoacoustic Turing test is proposed that incorporates these aided performance measures as part of a hearing-aid design, a validation strategy, and a comparative technique for evaluating different hearing-aid designs.
Compression, cochlear implants, and psychophysical laws
Authors: Zeng Fan-Gang
Published: September 2013

Cochlear compression contributes significantly to sharp frequency tuning and wide dynamic range in audition. The physiological mechanism underlying the compression has been traced to the outer hair cell function. Electric stimulation of the auditory nerve in cochlear implants bypasses this compression function, serving as a research tool to delineate the peripheral and central contributions to auditory functions. In this talk, I will compare psychophysical performance between acoustic and electric hearing in intensity, frequency, and time processing, and pay particular attention to the data that demonstrate the role of cochlear compression. Examples include both the cochlear-implant listeners’ extremely narrow dynamic range and poor pitch discrimination and their exquisite sensitivity to changes in amplitude and phase. A unified view on the complementary contributions of cochlear compression and central expansion will be developed to account for Webers’ law and Stevens power law.
The compression curves of different models of cochlear nonlinearities
Authors: Harte James M. , Elliott Stephen J. , Rice Henry
Published: September 2013

A widely used method for representing the nonlinear response of the basilar membrane (BM) is a graph of log amplitude of the BM motion against the log amplitude of the sinusoidal driving pressure, known as the input–output level curve. At low sound pressure levels (less than approximately 30 dB) the level of the BM response rises with sound pressure level at a slope of about 1 dB/dB, indicating a linear response. Above this region the slope of the level curve decreases, typically to about 1/2 to 1/3 dB/dB, indicating a compressive nonlinearity. Various models for cochlear nonlinearity will be presented, particularly contrasting the characteristics and behavior of an instantaneously acting nonlinear function and those of level-dependent systems. Both of these models have been used historically to model the nonlinearities present in the cochlea. Comparisons will be made between the properties of these models, highlighting potential methods of distinguishing between them. Similar nonlinear responses to those of the BM are observed in otoacoustic emissions. Experimental results will be reported which attempt to distinguish between the instantaneously acting and level-dependent models of this nonlinearity, which suggest that the latter is more likely.
The effect of amplitude compression, time constants, and number of channels on speech intelligibility in noise
Authors: Houben Rolph , Smoorenburg Guido F.
Published: September 2013

The influence of several compression parameters on speech intelligibility in speech-shaped noise was systematically investigated. Experimental conditions include all combinations of compression ratio (CR=1/2, 2/2, 2/3, 3/3 for low/high frequencies), attack and release times (Ta/Tr=4/4, 4/40, 40/40, 4/400, and 40/400 ms), and number of channels (1, 2, and 6). Twenty subjects with moderate sensorineural hearing loss took part in the experiment. The best average speech reception threshold occurred with two-channel compression at a compression ratio of 2/3 and Ta/Tr=40/40 ms. It was 0.7 dB better than linear amplification. The standard deviation between subjects was 1.7 dB. With six-channel compression the best results (0.4 dB better than linear amplification) were obtained at larger time constants (Ta/Tr=40/400 ms) and CR=2/2 and 2/3. The best result with single-channel compression was equal to linear amplification and occurred for the longest time constant (Ta/Tr=40/400 ms) and CR=2/2. In fluctuating noise the best speech reception threshold was found with single-channel compression at CR=2/2 and Ta/Tr=4/400 ms. It was 0.9 dB better than linear amplification; the standard deviation was 2.3 dB. The results suggest that finding the optimal condition for an individual is tedious.
The effects of complementary and alternative medicine on the speech of patients with depression
Authors: Fraas Michael , Solloway Michele
Published: September 2013

It is well documented that patients suffering from depression exhibit articulatory timing deficits and speech that is monotonous and lacking pitch variation. Traditional remediation of depression has left many patients with adverse side effects and ineffective outcomes. Recent studies indicate that many Americans are seeking complementary and alternative forms of medicine to supplement traditional therapy approaches. The current investigation wishes to determine the efficacy of complementary and alternative medicine (CAM) on the remediation of speech deficits associated with depression. Subjects with depression and normal controls will participate in an 8-week treatment session using polarity therapy, a form of CAM. Subjects will be recorded producing a series of spontaneous and narrative speech samples. Acoustic analysis of mean fundamental frequency (F0), variation in F0 (standard deviation of F0), average rate of F0 change, and pause and utterance durations will be conducted. Differences pre- and post-CAM therapy between subjects with depression and normal controls will be discussed.
Within-speaker speech intelligibility in dysarthria: Variation across a reading passage
Authors: Yunusova Yana , Weismer Gary , Westbury John , Rusche Nicole
Published: September 2013

Understanding factors underlying intelligibility deficits in dysarthria is important for clinical and theoretical reasons. Correlation/regression analyses between intelligibility measures and various speech production measures (e.g., acoustic or phonetic) are often reported in the literature. However, the analyses rarely control for the effect of a third variable (severity of speech disorder, in this case) likely to be correlated with the primary correlated variables. The current report controls for this effect by using a within-speaker analysis approach. Factors that were hypothesized to underlie the intelligibility variations in multiple breath groups within a connected discourse included structural elements (e.g., number of total words) as well as acoustic measures (e.g., F2 variation). Results showed that speech intelligibility in dysarthric speakers with two forms of neurological disease (Parkinson and ALS) does, in fact, vary across breath groups extracted from a connected discourse, and that these variations are related in some cases to a per breath estimate of F2 variation. [Work supported by NIDCD Award No. R01 DC03723.]
Effect of unilateral versus bilateral electrostimulation in subthalamic nucleus on speech in Parkinsons disease
Authors: Wang Emily , Verhagen Metman Leo , Bakay Roy , Arzbaecher Jean , Bernard Bryan
Published: September 2013

Previously, it was found that 16 right-handed patients with idiopathic Parkinsons disease who underwent unilateral implantation of deep brain stimulator in subthalamic nucleus (STN) showed significant improvement in their nonspeech motor functions. Eight of the 16 patients had stimulator in the left STN and eight in the right STN. In contrast, their speech function showed very mild improvement that was limited to the respiratory/phonotory subsystems. Further, there seemed a trend that the patients with right STN stimulation did better than those with left STN stimulation. It was speculated that the difference might be due to a micro lesion caused by the surgical procedure to the corticobulbar fibers run in the left internal capsule. This paper reports speech changes associated with bilateral DBS in STN in four of the 16 subjects who elected to have deep brain stimulator implanted in STN on the opposite side of the brain at a later time. Results show negative changes in speech after bilateral DBS in STN. The changes were not limited to the micro lesion effect due to the surgery itself, but also related to the active stimulation on the dominant hemisphere for speech processing. [Work supported by NIH.]
Grouping Miller–Nicely by linear vector space rotations
Authors: Budhlakoti Suvrat , Allen Jont B. , Larsen Erik
Published: September 2013

Human speech recognition has been studied using response to CV speech stimuli. Miller and Nicely (1955) studied such data in the form of confusion matrices to obtain insight into the psychological structure of the phone in noise. Here, the confusion matrices are modeled as phone coordinates in a high dimensional perceptual vector space. The model generalizes to an eigenvalue decomposition (EVD) [Allen (2004)]. This is followed by agglomerative hierarchical clustering of the transformed data, and an automated process is used to identify the main clusters. The resulting EVD clustering is very similar to other Miller–Nicely groupings, based on both production and MDS derived features, but is more model based. It was found that there is a gradual and highly consistent change in the clustering of sounds, independent of cluster size and configuration. By examining the change in similarity between various speech sounds, it is hoped that perceptual features may be uniquely identified.
Unsupervised learning of broad phonetic classes with a statistical mixture model
Authors: Lin Ying
Published: September 2013

Unsupervised learning of broad phonetic classes by infants was simulated using a statistical mixture model. A mixture model assumes that data are generated by a certain number of different sources—in this case, broad phonetic classes. With the phonetic labels removed, hand-transcribed segments from the TIMIT database were used in model-based clustering to obtain data-driven classes. Simple hidden Markov models were chosen to be the components of the mixture, with mel-cepstral coefficients as the front end. The mixture model was trained using an expectation-maximization-like algorithm. The EM-like algorithm was initialized by a K-means procedure and then applied to estimate the parameters of the mixture model after iteratively partitioning the clusters. The results of running this algorithm on the TIMIT segments suggested that the partitions may be interpreted as gradient acoustic features, and that to some degree the resulting clusters correspond to knowledge-based phonetic classes. Although such correspondences are rather rough, a careful examination of the clusters showed that the class membership of some sounds is highly dependent on their phonetic contexts. Thus, the clusters may reflect the preliminary phonological categories formed during language learning in early childhood.
Noise robust speech recognition with support vector learning algorithms
Authors: Namarvar Hassan H. , Berger Theodore W.
Published: September 2013

We propose a new noise robust speech recognition system using time-frequency domain analysis and radial basis function (RBF) support vector machines (SVM). Here, we ignore the effects of correlative and nonstationary noise and only focus on continuous additive Gaussian white noise. We then develop an isolated digit/command recognizer and compare its performance to two other systems, in which the SVM classifier has been replaced by multilayer perceptron (MLP) and RBF neural networks. All systems are trained under the low signal-to-noise ratio (SNR) condition. We obtained the best correct classification rate of 83% and 52% for digit recognition on the TI-46 corpus for the SVM and MLP systems, respectively under the SNR=0 (dB), while we could not train the RBF network for the same dataset. The newly developed speech recognition system seems to be noise robust for medium size speech recognition problems under continuous, stationary background noise. However, it is still required to test the system under realistic noisy environment to observe whether the system keeps its adaptability and robustness under such conditions. [Work supported in part by grants from DARPA CBS, NASA, and ONR.]
A novel dynamic acoustical model for speaker verification
Authors: Li Gongjun , Espy-Wilson Carol
Published: September 2013

In speaker verification, the conventional acoustical models (hidden Markov model and vector quantization) are not able to capture a speaker’s dynamic characteristics. In this paper we describe a novel dynamic acoustical model. The training data are viewed as a concatenation of many speech-pattern samples, and the pattern matching involves a comparison of the pattern samples and the test speech. To reduce the amount of computation, a tree is generated to index the entrance to pattern samples using an expectation and maximization (EM) approach, and leaves in the tree are employed to quantize the feature vectors in the training data. The obtained leaf-number sequences are exploited in pattern matching as a temporal model. We use a DTW scheme and a GMM scheme to match the training data and the test speech. Experimental results on NIST’98 speaker recognition evaluation data show that the accuracy of speaker verification on 3- and 10-s test speech is raised from 71.1% and 75.2% for a baseline GMM-based system to 80.0% and 82.1% for the dynamic acoustical model, respectively. Furthermore, some pattern samples in the training data are correctly tracked by the test speech.
Automatic detection of the features [high] and [low] in a landmark-based model of speech perception
Authors: Slifka Janet
Published: September 2013

This research is part of a landmark-based approach to modeling speech perception in which sound segments are assumed to be represented as bundles of binary distinctive features. In this model, probability estimates for feature values are derived from measurements of the acoustics in the vicinity of landmarks. The goal of the current project is to automatically detect the features [high] and [low] for vowel segments based on measurements from average spectra. A long-term and a short-term average spectrum are computed using all vowel regions in the utterance and are used to estimate speaker-specific parameters such as average F0 and average F3 (an indicator of vocal tract length). These parameters are used to estimate F1 using a peak-picking process on the average spectrum at each vowel-landmark. Preliminary results are derived from read connected speech for 738 vowels from 80 utterances (two male speakers, two female speakers). Speaker-independent logistic regression analysis using only average F0 and F1 determines the feature [high] with 73% accuracy and the feature [low] with 84% accuracy. Proposals are made for methods to use additional spectral detail to create a more robust estimate for vowels which show significant formant movement. [Work supported by NIH Grant No. DC02978.]
Spectral variability at the transition between successive phonemes
Authors: Dusan Sorin
Published: September 2013

In an experimental study of identification of truncated Japanese syllables it was stated that a short speech interval (approximately 10 ms) that includes the position of maximum spectral transition between a consonant and a vowel carries the most important information for the perception of the consonant and the syllable [S. Furui, J. Acoust. Soc. Am. 80, 1016–1025 (1986)]. A reduced spectral variability at the transition position could partially explain the increase of information at this position. The current study investigates whether there is a decrease in spectral variability at the transition position between successive phonemes compared with the spectral variability at the phoneme centers. The training part of the TIMIT acoustic-phonetic database, containing sentences in English from 462 American speakers, is used to build 2471 diphone models, based on the 61 symbols used in the database for phonetic transcription. The variability of the mel-frequency cepstral coefficients (MFCCs) is evaluated for various diphone models at the phoneme centers and at the phoneme transition position. Preliminary results suggest that spectral variability is not significantly lower at the phoneme transition positions than that at the phoneme centers in these diphone models.
Automatic speech recognizer based on the Spanish spoken in Valdivia, Chile
Authors: Sanchez Maria L. , Poblete Victor H. , Sommerhoff Jorge
Published: September 2013

The performance of an automatic speech recognizer is affected by training process (dependent on or independent of the speaker) and the size of the vocabulary. The language used in this study was the Spanish spoken in the city of Valdivia, Chile. A representative sample of 14 students and six professionals all natives of Valdivia (ten women and ten men) were used to complete the study. The sample ranged in age between 20 and 30 years old. Two systems were programmed based on the classical principles: digitalizing, end point detection, linear prediction coding, cepstral coefficients, dynamic time warping, and a final decision stage with a previous step of training: (i) one dependent speaker (15 words: five colors and ten numbers), (ii) one independent speaker (30 words: ten verbs, ten nouns, and ten adjectives). A simple didactical application, with options to choose colors, numbers and drawings of the verbs, nouns and adjectives, was designed to be used with a personal computer. In both programs, the tests carried out showed a tendency towards errors in short words with monosyllables like ‘‘flor,’’ and ‘‘sol.’’ The best results were obtained in words with three syllables like ‘‘disparar’’ and ‘‘mojado.’’ [Work supported by Proyecto DID UACh N S-200278.]
Measures of voiced frication for automatic classification
Authors: Jackson Philip J. B. , Jesus Luis M. T. , Shadle Christine H. , Pincas Jonathan
Published: September 2013

As an approach to understanding the characteristics of the acoustic sources in voiced fricatives, it seems apt to draw on knowledge of vowels and voiceless fricatives, which have been relatively well studied. However, the presence of both phonation and frication in these mixed-source sounds offers the possibility of mutual interaction effects, with variations across place of articulation. This paper examines the acoustic and articulatory consequences of these interactions and explores automatic techniques for finding parametric and statistical descriptions of these phenomena. A reliable and consistent set of such acoustic cues could be used for phonetic classification or speech recognition. Following work on devoicing of European Portuguese voiced fricatives [Jesus and Shadle, in Mamede et al. (eds.) (Springer-Verlag, Berlin, 2003), pp. 1–8]. and the modulating effect of voicing on frication [Jackson and Shadle, J. Acoust. Soc. Am. 108, 1421–1434 (2000)], the present study focuses on three types of information: (i) sequences and durations of acoustic events in VC transitions, (ii) temporal, spectral and modulation measures from the periodic and aperiodic components of the acoustic signal, and (iii) voicing activity derived from simultaneous EGG data. Analysis of interactions observed in British/American English and European Portuguese speech corpora will be compared, and the principal findings discussed.
Communication acoustics
Authors: Blauert Jens
Published: September 2013

Those aspects of acoustics which concern the relations of acoustics to the information and communication technologies are now frequently called ‘‘communication acoustics.’’ After a short review of the history of this field, relevant results from recent research at the Institute of Communication Acoustics at the Ruhr-University of Bochum, Germany, will be reported. This work can be seen in light of the research areas of computational auditory scene analysjs (CASA) and auditory virtual environments (AVE)—both dealing with the parametric representation of auditory scenes. Recent application opportunities and future trends will be discussed. It will be argued that modern communication–acoustical systems—which are often only embedded components in more complex communication systems—require more and more built-in explicit knowledge. Among other things, the development of such components and systems calls for data and knowledge from the cognitive sciences. Today’s programs for education in communication engineering and communication acoustics often do not take sufficient account of this trend.
Acoustical standards in the Society: Historical perspective
Authors: Embleton Tony F. W. , Schomer Paul D. , Blaeser Susan B.
Published: September 2013

The Acoustical Society of America has been active in national standards since 1932 when it asked the American Standards Association to initiate a project to standardize acoustical measurements and terminology. The new committee was assigned to the Society and designated Z24. By 1942, Z24 had expanded to include vibration. By 1957 the activity had grown so much that Z24 was split into three committees: S1 Acoustics, S2 Mechanical Vibration and Shock, and S3 Bioacoustics. (S12 Noise was added later, in 1981). Until 1969, these committees were administered by the USA Standards Institute, the predecessor of the American National standards Institute (ANSI), and new standards were approved by the Executive Council of the Society. In 1971, the Society assumed the responsibility for developing the US position on international standards in acoustics being generated by TC108 of the International Organization for Standardization. During the early 1970s ANSI became the body to approve all national standards, and the Soceity became a standards developer. ASA established a Committee on Standards (ASACOS) in 1978 to guide the work of the S-Committees and the Standards Secretariat. In the past few years ASACOS has increasingly improved its links with the Society’s other technical activities.
Role of standards in ASA: Animal bioacoustics
Authors: Bowles Ann E.
Published: September 2013

Animal bioacoustics (AB) is a participant in ASACOS committee S3, Bioacoustics, and has one working group S3/WG90. However, standards that could be written for animals cut across committee boundaries, from development of terminology (S1), to specification of audiometric methods (S2), to hearing conservation programs (S12). From a biologist’s perspective, there will be a fundamental difference between developing AB standards and those that have been published during the 75-year history of ASA—instead of focusing on one species, Homo sapiens, AB standards will be applied to many, even if data are available for only a few. Given the intensive research dedicated to establishing the existing standards for H. sapiens, the task ahead appears daunting. How should standards for thousands of species be specified? How should the standards process include a measure critical for wild animals, population sustainability? Writers of AB standards must apply a powerful conceptual tool, the comparative approach, and must design standards to incorporate new data quickly. Funding sources must recognize the need for these approaches. If they do, the standards developed will be marketable; it is also likely that important new perspectives on human bioacoustics will emerge.
Acoustical standards in engineering acoustics
Authors: Burkhard Mahlon D.
Published: September 2013

The Engineering Acoustics Technical Committee is concerned with the evolution and improvement of acoustical techniques and apparatus, and with the promotion of new applications of acoustics. As cited in the Membership Directory and Handbook (2002), the interest areas include transducers and arrays; underwater acoustic systems; acoustical instrumentation and monitoring; applied sonics, promotion of useful effects, information gathering and transmission; audio engineering; acoustic holography and acoustic imaging; acoustic signal processing (equipment and techniques); and ultrasound and infrasound. Evident connections between engineering and standards are needs for calibration, consistent terminology, uniform presentation of data, reference levels, or design targets for product development. Thus for the acoustical engineer standards are both a tool for practices, for communication, and for comparison of his efforts with those of others. Development of many standards depends on knowledge of the way products are put together for the market place and acoustical engineers provide important input to the development of standards. Acoustical engineers and members of the Engineering Acoustics arm of the Society both benefit from and contribute to the Acoustical Standards of the Acoustical Society.
Analysis of masking effects on speech intelligibility with respect to moving sound stimulus
Authors: Chen Chiung Yao
Published: September 2013

The purpose of this study is to compare the disturbed degree of speech by an immovable noise source and an apparent moving one (AMN). In the study of the sound localization, we found that source-directional sensitivity (SDS) well associates with the magnitude of interaural cross correlation (IACC). Ando et al. [Y. Ando, S. H. Kang, and H. Nagamatsu, J. Acoust. Soc. Jpn. (E) 8, 183–190 (1987)] reported that potential correlation between left and right inferior colliculus at auditory path in the brain is in harmony with the correlation function of amplitude input into two ear-canal entrances. We assume that the degree of disturbance under the apparent moving noisy source is probably different from that being installed in front of us within a constant distance in a free field (no reflection). Then, we found there is a different influence on speech intelligibility between a moving and a fixed source generated by 1/3-octave narrow-band noise with the center frequency 2 kHz. However, the reasons for the moving speed and the masking effects on speech intelligibility were uncertain.
Multi-dimensional analysis of subjective acoustical ratings and acoustical measures in existing concert halls
Authors: Okano Toshiyuki
Published: September 2013

Correlations between subjective acoustical ratings and hall-averaged values of acoustical measures are studied among existing worldwide major concert halls. It was shown that the classified acoustical ratings by Beranek [Concert and Opera Halls, How They Sound (ASA, 1996)] are discriminated correctly by combining binaural quality index (BQI) with some other acoustical measures. BQI is determined by the arithmetic average of inter-aural cross correlation coefficient in three octave bands of 500, 1000, and 2000 Hz, subtracted from unity, calculated from the early 80-ms part of binaural impulse response. Considering that the upper limit value of BQI not to cause disturbing image shift is approximately 0.85 at individual seat [Okano, J. Acoust. Soc. Am. 2219–2230 (2000)], the values of 0.6 or higher in hall averaged value of BQI, 0.85 or smaller in individual seat value of BQI, and approximately 5 dB or higher in strength factor at middle frequencies are proposed as design objectives to attain a high acoustical quality. It should be provided that other acoustical measures are also optimized. These target values will be very effective in studying room shape of halls, using scale models or computer models.
Microperforated stretched ceilings basics and applications
Authors: Nocke Christian , Hilge Catja
Published: September 2013

In 2001 a new product based on the theory of microperforated sound absorbers has been introduced. The thin foil of a stretched ceiling has been microperforated. Stretched-membrane ceilings have been a popular product with regard to modern architecture worldwide for more than 25 years. However, so far only optical and other aspects of the product were generally of interest. A brief introduction of the underlying theory of microperforated sound absorbers according to D.-Y. Maa will be given. Four geometrical parameters (diameter of perforation, distance between holes, thickness of panel/foil, and backing air volume) fully determine the sound-absorption coefficient of these sound absorbers. Results of independent qualification measurements of various assemblies will be shown. The agreement between theoretical predictions and measurements from the reverberation chamber is very high. Finally, different applications in a church, a sports hall, and other rooms will be presented. Again, the comparison between theoretical design and final result show a very high degree of agreement.
Lateral efficiency in small auditoriums
Authors: Moller Henrik , Peltonen Timo
Published: September 2013

The paper will examine the measured lateral efficiency parameters measured in small halls, that is, halls with less than 750 seats. Comparisons between measured LF and IACC parameters will be presented as well as a study of the correlation between the lateral efficiency parameters and the width and other geometrical attributes of the halls. The measurements were done as part of a larger survey of the Finnish halls that were built between 1980 and 2000.
Singer’s preferred acoustic condition in performance in an opera house and self-perception of the singer’s voice
Authors: Noson Dennis , Kato Kosuke , Ando Yoichi
Published: September 2013

Solo singers have been shown to over estimate the relative sound pressure level of a delayed, external reproduction of their own voice, singing single syllables, which, in turn, appears to influence the preferred delay of simulated stage reflections [Noson, Ph.D. thesis, Kobe University, 2003]. Bone conduction is thought to be one factor separating singer versus instrumental performer judgments of stage acoustics. Using a parameter derived from the vocal signal autocorrelation function (ACF envelope), the changes in singer preference for delayed reflections is primarily explained by the ACF parameter, rather than internal bone conduction. An auditory model of a singer’s preferred reflection delay is proposed, combining the effects of acoustical environment (reflection amplitude), bone conduction, and performer vocal overestimate, which may be applied to the acoustic design of reflecting elements in both upstage and forestage environments of opera stages. For example, soloists who characteristically underestimate external voice levels (or overestimate their own voice) should be provided shorter distances to reflective panels—irrespective of their singing style. Adjustable elements can be deployed to adapt opera houses intended for bel canto style performances to other styles. Additional examples will also be discussed. a)Now at Kumamoto Univ., Kumamoto, Japan. b)Now at: 1-10-27 Yamano Kami, Kumamoto, Japan.
Acoustical qualification of Teatro Nuovo in Spoleto before refurbishing works
Authors: Cocchi Alessandro , Cesare Consumi Marco , Shimokura Ryota
Published: September 2013

To qualify the acoustical quality of an opera house two different approaches are now available: one is based on responses of qualified listeners (subjective judgments) compared with objective values of selected parameters, the other on comparison tests conducted in suited rooms and on a model of the auditory brain system (preference). In the occasion of the refurbishment of an opera house known for the Two Worlds Festival edited yearly by the Italian Composer G. C. Menotti, a large number of measurements were taken with different techniques, so it is possible to compare the different methods and also the results with some geometrical criterion, based on the most simple rules of musical harmony, now neglected as our attention is attracted to computer simulations, computer aided measurement techniques and similar modern methods. From this work some link between well known acoustical parameters (not known at the time when architects sketched the shape of ancient opera houses) and geometrical criteria (well known at the time when ancient opera houses were built) will be shown.
Proposal of a new plane shape of an opera house—optimized by genetic algorithms
Authors: Hotehama Takuya , Ando Yoichi , Tani Akinori , Kawamura Hiroshi
Published: September 2013

The horseshoe-shaped theater has been the main shape from historical circumstances. However, from acoustical points of view, the rationality of the peculiar plane shape is not yet verified more than historical refinement. In this study, in order to make the theater shape more acoustically excellent, optimization for temporal and spatial factors in the theory of the subjective preference was made using genetic algorithms (GAs) by operating the positions of side walls. Results reconfirm that the plane shape of the optimized theater is a leaf shape, which has been verified to be acoustically rational in a concert hall. And, further possible shapes are also offered.
Influence of low-frequency vibration on changes of biochemical parameters of living rats
Authors: Kasprzak Cezary , Damijan Zbigniew , Panuszka Ryszard
Published: September 2013

The aim of the research was to investigate how some selected biochemical parameters of living rats depend on exposure of low-frequency vibrations. Experiments were run on 30 Wistar rats randomly segregated into three groups: (I) 20 days old (before puberty), (II) 70th day after; (III) control group. The exposure was repeated seven times, for 3 h, at the same time of day. Vibrations applied during the first tests of the experiment had acceleration 1.22 m/s2 and frequency 20 Hz. At the 135th day the rats’ bones were a subject of morphometric/biochemical examination. The results of biochemical tests proved decrease in LDL and HDL cholesterol levels for exposed rats as well as the Ca contents in blood plasma. There was evident increasing of Ca in blood plasma in exposed rats for frequency of exposition.
Hydroacoustic observations of sub-Indian ocean earthquakes using arrays of the International Monitoring System
Authors: Pulli Jay J. , Upton Zachary M.
Published: September 2013

Until recently, there have been few public hydroacoustic recordings of earthquakes beneath the Indian Ocean. Now, the International Monitoring System, which includes three Indian Ocean station sites, is providing excellent data for study. These stations consist of three-element horizontal hydrophone arrays floated to the middle of the sound channel axis, and can record low-level signals from distant sources. System bandwidth is approximately 2–100 Hz. We have accumulated a waveform database of 150 sub-Indian Ocean earthquakes and surrounding continental events recorded by this system. These data allow us to research T-wave generation and propagation. The recordings include seismic body-wave conversions beneath the arrays, T waves, and long-range reflections. Recordings of main shock/aftershock series have allowed us to compute scaling relationships for T-wave amplitudes versus magnitude and examine T-wave excitation versus source mechanism. When combined with distance corrections, we can scale T waves recorded at a given station to a fixed magnitude and distance. Plotted versus azimuth around the station, these scaled T-wave amplitudes provide insight into hydroacoustic blockage, which is essential for understanding detection capability. Seismic body-wave conversions beneath the arrays can be timed to 1 second or better, and provide important location constraints for subocean events occurring far from continental stations.
Thermal ablation by interstitial ultrasound
Authors: Lafon Cyril , Melodelima David , Chapelon Jean-Yves , Cathignol Dominique
Published: September 2013

Interstitial ultrasound applicators have been proposed for treating deep-seated tumors that are unreachable with extracorporeal high intensity focused ultrasound. The technique consists of bringing the ultrasound source as close as possible to the target in order to minimize the effects of attenuation and phase aberration along the ultrasound pathway. We have designed interstitial applicators with flat ultrasound transducers that operate at a frequency between 3.5 and 20 MHz. Nonfocused transducers require working at high frequencies to obtain coagulation in a short period of time. Flat transducers were chosen because the pressure drop in the near field is only due to absorption, and extended therapeutic depth can be achieved. A rotation of the transducer results in a cylindrical or sector-based volume of necrosis. To avoid these mechanical displacements, a cylindrical array has been manufactured; it allows for electronic rotation of a plane wave. Imaging modalities like endoscopic ultrasound and MRI were tested for guiding our treatments. Clinical trials are now being performed for the thermal ablation of digestive tumors and preliminary results are extremely promising.
Comparison between coupled KZK-BHTE numerical simulations and scanned HIFU exposures in excised bovine liver
Authors: Andrew Marilee A. , Brayman Andrew A. , Kaczkowski Peter J. , Kargl Steven G.
Published: September 2013

The use of moving high intensity focused ultrasound (HIFU) treatment protocols is of interest in achieving efficient formation of large-volume lesions in tissue. However, potentially unwanted thermal effects, such as prefocal heating, should be considered. A KZK acoustic model coupled with the BioHeat Transfer Equation has been extended to simulate multiple, moving scans in tissue. Simulation results are compared with experimental data collected over a range of exposure regimes for linear and concentric circular scans with a 3.5-MHz single-element transducer in ex vivo bovine liver. Of particular interest are investigating prefocal thermal buildup and ablating the central core of a circular pattern through conductive heating, that is without direct HIFU exposure. Qualitative agreement is observed between experimental and simulated data; limits of the predictive capability of the model in cavitation regimes will be discussed. [Support provided by the U.S. Army Medical Research Acquisition Activity through The University of Mississippi under terms of Agreement No. DAMD17-02-2-0014. The opinions expressed herein are those of the author(s) and do not necessarily reflect the views of U.S. Army Medical Research Acquisition Activity or The University of Mississippi.]
Measurement of sound intensity near flue organ pipes
Authors: Scholz Matthias , Abdugalder Essa Ibrahim , Kleiner Mendel , Yokota Munetaka
Published: September 2013

Organ voicers sometime experience that organ pipes, within an organ case, acoustically couple to one another. The coupling can be not only between a pipe and its mirror image in the case wall but also between adjacent pipes tuned to the same frequency. This paper reports on intensity measurements done to see the influence of a nearby pipe on the intensity field from a sounding pipe. The sounding pipe was driven pneumatically, and the sound intensity was measured in a plane along the length axis of the pipe. The secondary pipe was positioned, in parallel at different distances, alongside the sounding pipe. The results indicate only a fairly small influence by the secondary pipe on the intensity map of the sounding pipe. [Work supported by the Chalmers Foundation.]