bluuebunny/crossref_metadata_2025 huggingface dataset:
Effects of vocal training on voluntary responses to pitch-shifted voice auditory feedback
Authors: Patel Sona , Nishimura Cristina , Larson Charles
Published: May 2013

Control of voice fundamental frequency (F0) relies on the interaction between various forms of sensory feedback and neural motor control mechanisms. Several studies have shown that unexpected changes in pitch in the auditory feedback lead to reflexive compensatory vocal responses. We examined voluntary vocal responses to unpredictable perturbations in pitch auditory feedback. Ten subjects were trained over a five-day period to change their voice F0 in the opposite direction to the pitch-shifted feedback (±100 cents, 1000 ms) and 10 in the same direction as the feedback. Results showed that responses that followed the stimulus direction had significantly shorter latencies (200 ms) than opposing responses (324 ms). The reduced latency of the following responses suggests a switch from a feedback to a feedforward control strategy. The feedback strategy requires monitoring feedback and correcting for errors between the feedback signal and the intended vocal goal. The feedforward strategy relies less on auditory feedback and more on an internal model of the desired vocal pitch goal. Furthermore, feedback systems generally are slower than feedforward strategies, which would explain the shorter latencies of the responses that followed the stimulus direction. Results of this study will be discussed in terms of the differing strategies that may be used in various vocal contexts.
Investigating developmental changes in phonological representation using the imitation paradigm
Authors: Nielsen Kuniko
Published: May 2013

This study investigates the developmental changes in phonological representation, by examining the word- and phoneme-level specificity of phonetic imitation by children. Prevailing linguistic theories assume three levels of phonological representations: word, phoneme, and feature. Previous research suggests that phonological representations develop throughout childhood, and that phonological awareness develops from larger to smaller units [e.g., Edwards et al. (2004), Treiman and Zukowski (1996)]. It has been shown that adult speakers implicitly imitate the phonetic properties of recently heard speech [e.g., Goldinger (1998)], and recently, Nielsen (2011) showed the sub-phonemic generalizability and word- and phoneme-level specificity of imitation, indicating that three levels of phonological representations simultaneously contribute to the observed patterns of phonetic imitation. In order to test whether young children manifest similar patterns of imitation and specificity, an experiment with a modified imitation paradigm with a picture-naming task was conducted, in which participants' VOT was compared before and after they were exposed to target speech with artificially increased VOT. Our preliminary results reveal that two groups of children (5 year-olds and 8 year-olds) show greater imitation than adults, while word-and phoneme-level specificity was greater for 8 year-olds than 5 year-olds. These results provide support for the continued development of phonological representations.
Acoustic analysis of perceived accentedness in Mandarin speakers’ second language production of Japanese
Authors: Wei Peipei , Idemaru Kaori
Published: May 2013

Many second language (L2) learners, particularly adult learners, retain foreign accent on their L2 production. What acoustic sources give rise to the perception of foreign accent? This study examines beginning and intermediate Chinese learners’ production of Japanese in terms of segmental and suprasegmental features and investigates the relationship between acoustic characteristics of the L2 production and accentedness ratings provided by native Japanese listeners. Results of acoustic examination indicated that learners’ production varied considerably from that of native speakers in terms of durational features of stops, spectral features of some vowels, pitch (F0) peak alignment, and F0 contour. Multiple regression analysis identified the second formant of /u/, F0 peak alignment and contour as the strong predictors of perceived accent, accounting for nearly 90% of variance. These findings confirmed Flege’s Speech Learning Model hypothesis—L2 sounds that are similar to L1 sounds, while subphonemically distinct, seem to pose greater difficulty for acquisition than dissimilar sounds. Moreover, longer classroom experience was found to show limited effects in reducing perceived accent, with slightly greater effects on segmental than suprasegmental variables.
Acoustic and articulatory information as joint factors coexisting in the context sequence model of speech production
Authors: Duran Daniel , Bruni Jagoda , Dogil Grzegorz
Published: May 2013

This simulation study presents the integration of an articulatory factor into the Context Sequence Model (CSM) [Wade et al. (2010)] of speech production using Polish sonorant data measured with the electromagnetic articulograph technology (EMA) [Mücke et al. (2010)]. Based on exemplar-theoretic assumptions [Pierrehumbert (2001)], the CSM models the speech production-perception loop operating on a sequential, detail-rich memory of previously processed speech utterance exemplars. Selection of an item for production is based on context matching, comparing the context of the currently produced utterance with the contexts of stored candidate items in memory. As demonstrated by Wade et al. (2010), the underlying exemplar weighing for speech production is based on about 0.5 s of preceding acoustic context and following linguistic match of the exemplars. We extended the CSM by incorporating articulatory information in parallel to the acoustic representation of the speech exemplars. Our study demonstrates that memorized raw articulatory information—movement habits of the speaker—can also be utilized during speech production. Successful incorporation of this factor shows that not only acoustic but also articulatory information can be made directly available in a speech production model.
Articulatory compensation to second formant perturbations
Authors: Neufeld Chris , van Lieshout Pascal , Purcell David
Published: May 2013

There is a fast-growing literature examining speakers’ response to real-time alterations of auditory feedback. The majority of these studies examine the response of the subject in acoustic terms. Since many subjects fail to (acoustically) compensate for the perturbation, the current experiment examines whether there are systematic articulatory responses to formant perturbation in the absence of compensation at the level of acoustics. Articulatory data are collected using a 3D electro-magnetic-articulograph. F2 is gradually shifted up or down and preliminary results from three English-speaking subjects showed that two subjects show no response in their acoustics or articulation. However, the remaining speaker who did not show compensation at the level of acoustics displayed a systematic response in some articulatory variables. The acoustic effects of his response were masked because the other articulators behaved in a more variable way, making the second formant vary randomly from trial to trial. Based on these results we expect to see a spectrum of response patterns from a larger population of speakers, from total non-compensation in both acoustics and articulation, partial compensation in articulation, and global articulatory compensation, which induces the appropriate compensation at the level of acoustic output.
Acoustic vowel space and speech rate in Mandarin-speaking children with cerebral palsy
Authors: Hsu Wei Chen , Chen Li-mei , Cheng Fan-Yin
Published: May 2013

This study examines the variability in speech production in four Mandarin-speaking children: two with cerebral palsy (CP) and two typically developing (TD) from 4 to 5 years of age. Recordings collected from the picture-naming task and spontaneous interaction with adults was analyzed. Acoustic vowel space and speech rate in their production were investigated. Study findings indicated the following: (1) Due to defect in speech motor control, children with CP have a smaller overall vowel space than TD children; (2) In CP group, there are more variability of formant values of individual vowels and the vowel space of individual vowels thus overlap more; (3) There is a trend of decrease of vowel formant values in both TD and CP; (4) Children with CP tend to spend more time in speech production because of their impaired speech-motor control, in terms of syllable per minute and intelligible syllable per minute; (5) Slower speech rate seems to increase speech intelligibility in CP. However, this needs to be verified in further studies. Extended longitudinal observation can provide more complete profile of individual differences in the development of vowels and speech rate to verify these preliminary findings. The variability features in the production of children with CP provide important references in speech therapy.
Investigating perceptual attributes associated with reproduction of early reflections via virtually elevated loudspeaker
Authors: Kim Sungyoung
Published: May 2013

Reproducing early reflections related to “height” information through elevated loudspeakers delivers enhanced presence of auditory images and integrates with a three-dimensional visual content homogeneously. Nonetheless, it is practically difficult for consumers to place loudspeakers required for the height-channel reproduction in a listening room. To overcome this limitation, many academic or commercial institutions propose various methods that render vertical sound images and reproduce them with smaller number of loudspeakers that are typically located in the horizontal plane. The rendered image then could deliver vertically extended impression of a sound field, which is likely related to listeners’ perception of enhanced presence. To better understand this relationship, this paper investigated idiosyncratic difference between one surround sound field and another with early reflections that is virtually elevated. The elicitation result revealed that listeners used four salient attributes—ASW, LEV, Powerfulness, and Clarity—to describe the difference. The subsequent result showed that perceived magnitudes of those percepts were accounted for by a physical parameter, correlation coefficient between the elevated signal and the loudspeaker signal that is to feed to the closest loudspeaker in the horizontal plane.
Approximate convolution using partitioned truncated singular value decomposition filtering for binaural rendering
Authors: Atkins Joshua , Strauss Adam , Zhang Chen
Published: May 2013

In conventional binaural rendering, a pair of head-related impulse responses (HRIR), measured from source direction to left and right ears, is convolved with a source signal to create the impression of a virtual 3D sound source when played on headphones. It is well known that using HRIRs measured in a real room, which includes a natural reverberant decay, increases the externalization and realism of the simulation. However, the HRIR filter length in even a small room can be many thousands of taps, leading to computational complexity issues in real world implementations. We propose a new method, partitioned truncated singular value decomposition (PTSVD) filtering, for approximating the convolution by partitioning the HRIR filters in time, performing a singular value decomposition on the matrix of filter partitions, and choosing the N singular-vectors corresponding to the N largest singular values to reconstruct the HRIR filters. We will show how this can be implemented in an efficient filter-bank type structure with N tapped delay lines for real-time application. We also show how improvements to the method, such as modeling the direct path HRIR separately can lead to improved rendering at minimal computational load.
The influence of regularization on anechoic performance and robustness of sound zone methods
Authors: Coleman Philip , Jackson Philip , Olik Marek , Olsen Martin , Mo/ller Martin , Pedersen Jan Abildgaard
Published: May 2013

Recent attention to the problem of controlling multiple loudspeakers to create sound zones has been directed toward practical issues arising from system robustness concerns. In this study, the effects of regularization are analyzed for three representative sound zoning methods. Regularization governs the control effort required to drive the loudspeaker array, via a constraint in each optimization cost function. Simulations show that regularization has a significant effect on the sound zone performance, both under ideal anechoic conditions and when systematic errors are introduced between calculation of the source weights and their application to the system. Results are obtained for speed of sound variations and loudspeaker positioning errors with respect to the source weights calculated. Judicious selection of the regularization parameter is shown to be a primary concern for sound zone system designers—the acoustic contrast can be increased by up to 50 dB with proper regularization in the presence of errors. A frequency-dependent minimum regularization parameter is determined based on the conditioning of the matrix inverse. The regularization parameter can be further increased to improve performance depending on the control effort constraints, expected magnitude of errors, and desired sound field properties of the system.
Scales of time and space variability of sound fields reflecting obliquely from underwater slopes
Authors: Duda Timothy , Lin Ying-Tsong , Cornuelle Bruce D.
Published: May 2013

Ocean flows impart a time dependence to interference patterns of sound reflecting at low horizontal angle from sloping bathymetry. This is a higher level of complexity than interference patterns within vertical slices in spherically symmetric environments. The time-space statistics may be co-dependent and non-normal, making conventional methods inapplicable. Here, patterns that occur when 100 to 1000 Hz sound reflects at slopes are simulated with three-dimensional methods. Multiple statistics of sound at virtual arrays are computed over the domain, including incoherent power, beam power, spatial correlation length, and array gain assuming idealized noise conditions. These depend on source and receiver locations with respect to bathymetric features and can be computed for instantaneously sampled ocean conditions to study their evolution within the time frame of the dominant flow, or computed via averaging over a few periods of the dominant flow features (tides, for example). Here, the spatial complexity of patterns found in 100-Hz (and upward) simulations at a slope area off San Diego, CA, in a time varying flow are linked to the imposed seafloor roughness as well as geometry, with mean intensity, scintillation index, and correlation scale used to quantify the effect. Implications for statistics employed in detection and tracking algorithms are discussed.
The effect of random fluctuations in bottom bathymetry on acoustic coherence in shallow water
Authors: DeFerrari Harry A. , Wylie Jennifer
Published: May 2013

The loss of temporal coherence after long range propagation in shallow water is often studied as a consequence of sound speed variability from internal waves. Here, we add the complication of small amplitude and very long wavelength random fluctuations of bottom bathymetry. It is shown that the same range dependent sound speed fluctuations result in markedly different coherence times depending on acoustic wavelength and mode number—a first order effect. A range dependent PE code (MMPE) is used to predict temporal coherence for individual surface reflected-bottom-reflected (SRBR) mode arrivals. Here, a mode coherence calculation is developed and compared for varying RMS bathymetry. Temporal coherence is inferred from mode coherence. We find first order and /or low frequency modes are insensitive to the bottom but when the (sine of the mode angle approaches 1/10 of an acoustic wavelength) the modes structure in amplitude and phase is randomized and the signal decorrelate rapidly in time from just the slightest temporal variations in sound speed. It does not take much; just 1 m in 200 m of range will randomize all but the first mode at mid frequencies (0.5 to 1 kHz). Predictions are in close agreement with SW06 mode coherence measurements.
Improving the indoor sound quality by using cymatic shapes
Authors: Algargoosh Alaa S. , Hossam Eldien Hany , El Wakeel Hala
Published: May 2013

Acoustic diffusers are important components in enhancing the quality of room acoustics. This paper investigates a new type of 2D diffusers obtained by the Cymatics phenomena. Cymatics is the study of sound and vibration made visible, typically on the surface of a plate, diaphragm, or membrane. Four shapes of the diffusers were designed by the Cymatic shapes and modeled by using a quadratic residue sequence. The polar response of the diffusers was measured using DIRAC software. Polar response results were generally consistent with expectations. This type of diffusers can generate a uniform polar response over the frequency range we are interested in (400–4000 Hz). It is found that this type of acoustic diffusers can be used to maintain the acoustic energy in a room and at the same time can treat unwanted echoes and reflections by scattering sound waves in many directions.
Sound concentration caused by curved surfaces
Authors: Vercammen Martijn
Published: May 2013

In room acoustics, the focusing effect of reflections from concave surfaces is a well-known problem. The occurrence of concave surfaces has tended to increase in modern architecture, due to new techniques in design, materials, and manufacturing. Focusing can cause high sound pressure levels, sound coloration, or an echo. Although the problem is well known, the amount of amplification that occurs in the focusing point and the sound field around the focusing point are not. The pressure in the focusing point can only be calculated using wave-based methods. An engineering method that is based on the Kirchhoff Integral is presented to approximate the reflected sound field in and around the focusing point for a few basic geometries. It will be shown that both the amplification and the area of the focusing is strongly related to wavelength. The focusing caused by surfaces that are curved in two directions (sphere, ellipsoid) is much stronger than that caused by surfaces that are curved in only one direction (cylinders). This method enables designers to evaluate and thereby improve or redesign the geometry. The method is illustrated with a few examples.
Optimization of electroacoustic resonators for semi-active room equalization in the low-frequency range
Authors: Rivet Etienne , Boulandet Romain , Lissek Hervé
Published: May 2013

At low frequencies in listening rooms, standing waves cause large frequency-response variations within the whole space. These unwanted phenomena have a significant impact on the sound quality of an audio system rendering. Unfortunately, state-of-the-art soundproofing solutions cannot efficiently handle such low-frequency sound energy. To alleviate this problem, electroacoustic resonators can be used to damp room modes. This concept is based on the connection of direct-radiator loudspeakers to synthetic electrical loads allowing the passive dissipation of a certain part of the incoming acoustic energy of the sound field. Through judicious control of acoustic impedance, and depending of the placement of the electroacoustic resonators in the room, a significant damping of the dominant natural resonances can be achieved in order to meet the specifications of sound reproduction. This paper presents the design of prototypes of electroacoustic resonators and investigates their optimization and spatial arrangement in the perspective of semi-active room equalization.
Physical and subjective factors of spatial envelopment impression of surround sound reproduction
Authors: Kamekawa Toru , Marui Atsushi
Published: May 2013

How we feel the envelopment of reproduced sound? For the evaluation of spatial impression, “envelopment” is one of the key factors among several attributes in audio reproduction. In the authors' past research, three attributes were elicited from participants regarding surround sound reproduction using triadic elicitation procedure. The three attributes are “brightness,” “temporal separability,” and “spatial homogeneity of envelopment.” In this paper, authors focused on “spatial homogeneity of envelopment” and considered related physical parameters such as ESC (ear signal correlation; defined as the correlation between left and right ears' signal). From comparison between the result of subjective test and the data measured with a dummy-head facing toward different angles, it was found that the correlation between the ESC of different head angles and the amount of differences of ESC from focused sound to diffused sound contribute the sense of “spatial homogeneity of envelopment.”
Prof. Dah-You Maa’s Contribution to Acoustics
Authors: Tian Jing
Published: May 2013

In his life-long career pursuing, Professor Dah-You Maa contributed greatly in many areas of acoustics, not only in acoustical research and development, but also in the promotion of acoustical education, application, and legislation. Professor Maa presented a simplified method for calculation of normal modes for room acoustics, invented micro-perforated panel absorbers and micro-perforation jet mufflers, gave a formula of jet noise power via air pressure, and acoustically designed the first and biggest Congress Hall, built the first set of acoustical laboratories, and established the acoustical standard system in China. He also supervised tens of postgraduate students working in environmental acoustics, building acoustics, speech signal processing, nonlinear acoustics, and active noise control. In this paper, the main contribution of Professor Maa is introduced. Several of his typical research works, such as micro-perforated panel absorbers and micro-perforation jet mufflers, are explained in detail, from which we can well appreciate the physical insights and theoretical skills of Professor Maa.
Dah-You Maa and the many facets of modal density
Authors: Lyon Richard H.
Published: May 2013

Maa’s formula for the modal density of a room is known to everyone in acoustics. The wonderful part of it is that it so logical and direct that you don’t have to memorize or look it up - you can simply re-derive it on the spot. When it was published by Maa in 1939, there was a competing formulation by Dick Bolt that had the same volume dependent term, but was quite different in the surface area and edge length terms. But by the time of the landmark paper by Morse and Bolt in 1944 [“Sound waves in rooms”, Rev. Mod. Phys. 16(1) (1944)], Maa’s formula had won and was the relationship cited by Morse and Bolt. The uses of modal density in the response statistics of acoustical spaces and structures for the purposes of estimating impedances, energy flow, and phase statistics have grown as recognition of this fundamental property of resonant systems has grown.
Remembering one of Maa Dah-You’s last essential contributions to acoustics
Authors: Wang Jiqing
Published: May 2013

The book The fundamentals of Modern Acoustic Theory written by Maa Dah-You was published (in Chinese) in March 2004 at his age of 88. This 421 pages volume with 15 chapters covers almost all main acoustic fields from basics to many extended areas, such as linear to non-linear acoustics, common acoustical measures to active control, and the aerodynamic sound and thermo-acoustics. References listed in each chapter include many Maa's pioneering work of the related acoustic fields. This book systematically reflects his deeper understanding of acoustics theory, and further account new developments of acoustic technology. Moreover, the history of acoustics, particularly of the ancient China in the beginning sections of the first chapter, and the overview of the future developments in the afterword are also the exclusive items in this book. A well prepared exercise at the end of each chapter is most helpful for better understanding and thinking of the related topics. Therefore, it is widely used as a text or main reference book in many universities and research community in China.
A composite sound absorber with micro-perforated panel and shunted loudspeaker
Authors: Tao Jiancheng , Jiao Qijing , Qiu Xiaojun , Han Ning
Published: May 2013

Micro-perforated panel (MPP) backed by a rigid cavity is a widely used clean sound absorber; however, its application at low frequency is limited because a deep cavity is required to achieve good sound absorption in the low frequency range. In the present paper, a composite absorber composed by an MPP and a shunted loudspeaker is proposed. The loudspeaker is installed at the back wall of the air cavity and the acoustic impedance can be optimized by adjusting the parameters of the loudspeaker and the electronic components in the shunt to improve the sound absorption performance. The prediction model of such a finite-sized composite absorber is established based on the mode analysis solution and the equivalent circuit of the loudspeaker. Both numerical simulations and experiments show that the thickness of the proposed composite absorber can be much smaller than that of traditional MPP constructions.
A hybrid modal analysis for enclosed sound fields and its applications
Authors: Xu Buye , Sommerfeldt Scott D. , Leishman Timothy W.
Published: May 2013

In 1939, Dr. Dah-You Maa presented a paper entitled “The distribution of eigentones in a rectangular chamber at lower frequency ranges” [J. Acoust. Soc. Am. 10, 258 (1939)]. Since then, his interest in room acoustics has not diminished. More than six decades later, Maa proposed an idea of adding a monopole solution to the normal modal expansion to improve the accuracy in simulating the near-field sound field in rooms [D. Y. Maa, Acta Acust. (Beijing) 27, 385–388 (2002)]. Following this idea, a hybrid model that combines the free field Green’s function and a modal expansion has been proposed by the authors based on a rigorous mathematical derivation [Xu et al., J. Acoust. Soc. Am. 128, 2857–2867 (2010)]. The hybrid modal expansion can be further extended for complex sound sources by introducing the multipole expansion to the solution. In this talk, the theoretical derivation of the hybrid modal expansion will be reviewed, followed by examples demonstrating the use of the hybrid modal expansion in real-world applications.
The search for a neural basis of communication: Learning, memory, perception. and performance of vocal signals
Authors: Prather Jonathan
Published: May 2013

Brain mechanisms for communication must establish a correspondence between sensory perception and motor performance of individual signals. A class of neurons in the swamp sparrow forebrain is well suited for that task. Recordings from awake and freely behaving birds reveal that those cells express categorical auditory responses to changes in note duration, a learned feature of their songs, and the neural response boundary accurately predicts the categorical perceptual boundary measured in field studies. Extremely precise auditory activity of those cells represents not only songs in the adult repertoire but also songs of others and tutor songs, including those imitated only very few times or perhaps not at all during development. Furthermore, recordings during singing reveal that these cells also express a temporally precise auditory-vocal correspondence, and limits on auditory responses to extremely challenging tutor songs may contribute to the emergence of a novel form of song syntax. Therefore, these forebrain neurons provide a mechanism through which sensory perception may influence motor performance to enable imitation. These cells constitute the projection from a premotor cortical-like area into the avian striatum (HVCX neurons), and data from humans implicate analogous or homologous areas in perception and performance of the sounds used in speech.
Influences of perceptual continuity on everyday listening
Authors: Shinn-Cunningham Barbara , Mehraei Golbarg , Bressler Scott , Masud Salwa
Published: May 2013

In the natural environment, listeners face the challenge of parsing the sound mixture reaching their ears into individual sources, and maintaining attention on a source of interest through time long enough to extract meaning. A number of studies have shown that continuity of certain acoustic features (including pitch, location, timbre, etc.) allows the brain to group sound from one acoustic sound source together through time to form an auditory object or stream. This presentation reviews results demonstrating that auditory feature continuity has important consequences on how listeners maintain attention on a stream through time. For instance, continuity of a sound feature that a listener knows is irrelevant to the task at hand nonetheless impacts the ability to maintain auditory attention based on some other sound feature. Moreover, the influence of auditory feature continuity decreases as the time between events in a given sound stream increases. Taken together, these behavioral results support the idea that auditory attention operates on auditory objects, rather than on individual sound features, and that feature continuity has an obligatory influence on the formation of auditory streams, and therefore on how selective auditory attention allows us to communicate in everyday settings.
Experimental characterization of light emission during shock-driven cavity collapse
Authors: Anderson Phillip , Hawker Nicholas , Betney Matthew , Tully Brett , Ventikos Yiannis , Roy Ronald A.
Published: May 2013

The authors describe experimental work examining the collapse of a cavity by a strong shockwave. A millimeter size cavity is cast in Phytagel, which is then impacted by a metallic projectile accelerated by a compressed gas gun, reaching velocities up to 500 m/s. The impact generates a strong shockwave that propagates into the gel at greater than sonic velocity. Schlieren images are presented that illustrate both this process and the subsequent cavity collapse at a sub-microsecond timescale. As the shockwave reaches the cavity, it is shown to cause a rapid asymmetric collapse process characterized by the formation of a high-speed transverse jet. The pressure of the shockwave is found to be 100+ MPa as measured via a custom-built fiber-optic probe hydrophone. Previous work examining shock-driven cavity collapse observed luminescence, postulated to be due to the high-speed impact of the transverse jet on the far bubble wall; this experimental observation is replicated. Further, the light emission is characterized as a function of impact velocity and thus of shockwave pressure. This reveals that shock-driven cavity collapse shares many of the unique features that make the more widely studied SBSL-type collapse interesting.
Simulation of warm dense matter in intense bubble collapse
Authors: Tully Brett , Hawker Nicholas , Betney Matthew , Ventikos Yiannis
Published: May 2013

Previous work by the authors includes computational study of shock-bubble interaction. This work demonstrated strong compression and heating within the bubble, with gas reaching densities of order of magnitude 1 g/cc and temperatures of 10 eV. These conditions correspond to the warm dense matter regime. This paper addresses limitations of previous work through utilization of various equations of state (EOS) appropriate for the modeling of dense plasma. This is achieved through the design and implementation of a generic interface based on tabulated EOS data. Any EOS may be utilized through the framework, requiring only knowledge of pressure and energy as functions of density and temperature. The solutions to various issues such as table interpolation, tabulated change of variables, arbitrary calculation of entropy, and calculation of thermodynamic derivatives are presented. In addition, the trade-offs between CPU time, memory requirement and computational accuracy are discussed. Validation work is presented and a comparison of different EOS is also explored. The EOS used include but are not limited to the EOS for air utilized by Moss et al. (1994) to study SBSL, SESAME tabulated EOS and QEOS-type formulations. Finally, conditions attained during shock-bubble interaction are re-examined.
Use of dolphin-like pulses to enhance target discrimination and reduce clutter
Authors: Leighton Tim , Chua Gim-Hwa , White Paul R. , Tong Kenneth , Griffiths Hugh , Daniels David
Published: May 2013

Building on the earlier success of using twin inverted pulses to supress bubble scatter and so reduce clutter when detecting targets in bubbly water (proven in ship wakes in field trials), the same nonlinear processing scheme is generalized to make use of dolphin-like pulses. Their performance in reducing clutter and enhancing target discrimination is demonstrated in the laboratory, and the opportunities for using the same scheme to improve the detection of hidden electronic devices or semiconductor devices by radar are discussed.
Attenuation of sound in water through collections of very large bubbles with elastic shells
Authors: Lee Kevin M. , Wilson Preston S.
Published: May 2013

The ultimate goal of this work is to accurately predict the attenuation through a collection of large (on the order of 10 cm radius) tethered encapsulated bubbles used in an underwater noise abatement system. Measurements of underwater sound attenuation were performed during a set of lake experiments, where a low-frequency compact electromechanical sound source was surrounded by different arrays of encapsulated bubbles with various individual bubbles sizes and void fractions. The measurements are compared with an existing predictive model [J. Acoust. Soc. Am. 97, 1510–1521 (1995)] of the dispersion relation for linear propagation in liquid containing encapsulated bubbles. Although the model was originally intended to describe ultrasound contrast agents, it is evaluated here for large bubbles, and hence low frequencies, as a design tool for the underwater noise abatement system, and there is fairly good quantitative agreement between the data and the model. Refinements to the model to incorporate multiple scattering effects, which may be important at high void fractions, via an effective medium approach [J. Acoust. Soc. Am. 111, 168–173 (2002)] and comparison with the data will also be discussed. [Work supported by Shell Global Solutions.]
Dynamic quantitative ultrasound imaging of mimicked breast lesions during shear wave propagation to emphasize differences in tissue statistical backscatter properties
Authors: Alavi Marzieh , Destrempes Francois , Montagnon Emmanuel , Cloutier Guy
Published: May 2013

The main motivation in this study was to increase the accuracy of the breast tissue characterization by combining quantitative ultrasound (QUS) with ultrasound (US) dynamic elastography. To demonstrate that, an agar-gelatin breast mimicking phantom with two inclusions containing the same density of agar (US scatterers) but different proportions of gelatin corresponding to different mechanical properties was made. Transient plane shear waves (SW) at 200 Hz were transmitted through the phantom while the displacement of scatterers was imaged at 5 MHz with an ultrafast imaging technique. With segmented inclusions, the reciprocal (beta parameter) of the effective density of scatterers of a general distribution model of the echo envelope and its normalized range (normalized by the mean of beta during SW propagation) were estimated for each inclusion. The results showed that the relative difference of beta between the surrounding medium and both inclusions A and B were 55.6% (A) and 0.9% (B), respectively, whereas differences (in %) of the beta normalized range were 46.2% (A) and 52.6% (B), respectively. The static value beta failed to distinguish inclusion B from the surrounding; however, the dynamic range of beta succeeded in that task for the two inclusions. Thus, dynamic QUS might add information to QUS in a static framework.
Shear wave elastography for characterizing muscle tissue in myofascial pain syndrome
Authors: Turo Diego , Otto Paul , Gebreab Tadesse , Armstrong Katherine , Gerber Lynn H. , Sikdar Siddhartha
Published: May 2013

Myofascial pain syndrome (MPS) affects 85% of chronic pain sufferers in a specialty pain center. Neck and low-back are commonly affected by MPS. Myofascial trigger points (MTrPs) are characteristic findings of MPS and are palpable tender nodules in the muscles of symptomatic subjects. Mechanical characterization of MTrPs and surrounding tissue can offer important insight about the pathophysiology of the MPS, which is currently poorly understood. In this study, we propose an inexpensive technique, based on ultrasound shear wave elastography, to objectively measure mechanical properties of MTrPs and surrounding tissue in the upper trapezius. In an ongoing clinical study, we recruited 34 subjects: 12 healthy controls, 10 with not spontaneously painful MTrPs (latent) and 12 with symptomatic chronic neck pain (>3 months) and active (spontaneously painful) MTrPs. Shear wave elastography was performed on the upper trapezius of all subjects using the Ultrasonix RP system and an external vibrator. Voigt’ s model was used to estimate shear modulus G and viscosity μ of the interrogated tissue. Preliminary analysis demonstrates that symptomatic muscle tissue in subjects with neck pain is stiffer (G = 8.40 ± 8.31 kPa, mean ± standard deviation) compared to muscle in control subjects (G = 2.86 ± 2.48 kPa) (p < 0.05), and that active MTrPs are more viscous (μ = 17.10 ± 9.46 Pa*s) than surrounding tissue (μ = 10.59 ± 5.96 Pa*s). Latent MTrPs (μ = 24.31 ± 12.72 Pa*s) and surrounding tissue (μ = 20.09 ± 7.48 Pa*s) are more viscous than normal tissue (μ = 10.96 ± 4.17 Pa*s).
Speckle tracking in multiple-scattered pressure field
Authors: Kalkan-Savoy Ayse , Thompson Charles
Published: May 2013

Speckle tracking using ultrasound B-scan image sequences to quantify myocardial strain has the potential to become a standard method in echocardiography. A pressure field for a modulated Gaussian pulse traveling in inhomogeneous media is constructed. This model is based on computation of pressure field using Kirchoff integral formulation in frequency domain. Each pressure term of Neumann Series is converted into time domain and used in computation of Pade approximants. Pressure field in time and space is constructed utilizing Pade approximants and includes multiple-scattering effects. The scatterers of interest are large enough in size and contrast to contribute to strong and multiple-scattering. The signals at transducer point for each time frame are analyzed and speckles are tracked utilizing an optical flow algorithm. A velocity field which can be utilized to quantify strain is computed and visualized. The efficiency of this method will be discussed.
Virtual prototyping of condenser microphones using the finite element method for detailed electric, mechanic, and acoustic characterization
Authors: Herring Jensen Mads Jakob , Sandermann Olsen Erling
Published: May 2013

Recent development and advances within numerical techniques and computers now enable the modeling, design, and optimization of many transducers using virtual prototypes. Here, we present such a virtual prototype of a Brüel & Kjær Type 4134 condenser microphone. The virtual prototype is implemented as a model using the finite element method with COMSOL Multiphysics and includes description of the electric, mechanic, and acoustic properties of the transducer. The acoustic description includes thermal and viscous losses explicitly solving the linearized continuity, Navier-Stokes, and energy equations. The mechanics of the diaphragm are modeled assuming a pre-stressed membrane, electrostatic attraction forces, and acoustic loads. The model includes electric description of the active and passive capacitances of the microphone cartridge as well as an external circuit model representing the preamplifier. Different modes of the system are studied, including the important first rocking mode of the membrane. The model has no free fitting parameters and results in the prediction of the frequency dependent sensitivity, capacitance, and mechanical impedance. The model results show good agreement with measured data.
Directional robustness of an in-situ, dual dipole omni microphone array for hearing instruments
Authors: Burns Thomas
Published: May 2013

The three critical factors for providing stable directional performance for typical microphone arrays used in hearing aids include the relative sensitivity and phase between the microphones in addition to the placement of the hearing instrument behind the user's ear. A directional system is robust if these factors can operate over a wide range of levels without degrading the directional performance. In this study, dual dipole microphones were arranged symmetrically around an omnidirectional microphone such that all inlets were collinear. Compared to an endfire array, whether it be a delay-and-sum or Blumlein configuration, this dual-dipole-omni array is remarkably more robust, yielding very little degradation in the Directivity Index for the aforementioned critical factors varying as much as +/− 3 dB, +/− 30 ms, and the directional axis of the hearing instrument varying +/− 20 degrees on the ear.
Numerical study of the cross-talk effects in acoustical transducer arrays and correction
Authors: Bybi Abdelmajid , Assaad Jamal , Hladky-Hennion Anne-Christine , Benmeddour Farouk , Grondel Sébastien , Rivart Frederic
Published: May 2013

Cross-talk in acoustical transducer arrays is an undesirable phenomenon which decreases seriously the performances of these sensors. Indeed, when one element of the array is driven, it generates parasitic displacement fields at the radiating surfaces of the neighboring elements, which changes the directivity of the antenna. To well understand this phenomenon a transducer array similar to those used in medical imaging and NDT applications is modeled by finite element method. The research work, investigated systematically the effects of the cross-talk. First, it inspected the acoustical and mechanical cross-talk throughout the propagating medium and the filling material. Second, it studied the influence of the matching layer and the backing on the acoustical performances of the transducer. It is shown that the filling material and the matching layer are the major factor contributing to this phenomenon. In order to cancel the cross-talk a correction method previously developed by the author has been used. This solution consisted in applying adapted electrical voltages on each neighboring element of the active one in the purpose to reduce the displacement field on their active surface. This method was tested numerically and the obtained results clearly demonstrated its ability to reduce the cross-talk.
Objective evaluation of sound quality for attacks on robust audio watermarking
Authors: Nishimura Akira , Unoki Masashi , Kondo Kazuhiro , Ogihara Akio
Published: May 2013

Various attacks on robust audio watermarking have been proposed. Reversible signal processing attacks, such as sampling frequency conversion, degrade sound quality of the distributed watermarked audio (stego audio) and disturb extraction of hidden data so that copyright detection systems using automated crawling are invalidated. Reversible signal processing of the attack can recover sound quality of the degraded audio data. In order to prove validity and security of audio watermarking system, analysis of the presumed attacks or reversible signal processing on stego audio, is required. However, these attacks on audio signal also degrade sound quality of commercial music where such pieces of music are considered to be not suitable for appreciation. Therefore, degradation of sound quality induced by various attacks should be taken into account to decide if the intensity of the attacks are realistic. In this study, objective audio quality measurement (PEAQ) was applied to the audio signals including typical perceptual coding, MP3, tandem MP3, MPEG4AAC, and reversible signal processing of sampling frequency conversion, noise addition, frequency shift, bandpass filtering, and echo addition. The results indicate requirements for robustness and criteria of the attacks on high quality and robust audio watermarking technology.
The effect of firefighting protective equipment on head related transfer functions
Authors: Suits Joelle I. , Argo Theodore F. , Wilson Preston S. , Ezekoye Ofodike A. , Champlin Craig A.
Published: May 2013

Personal Alert Safety System (PASS) devices are used in the fire service to locate trapped or injured personnel. When a firefighter becomes incapacitated, the device emits an audible alarm to help rescue teams locate the downed firefighter. These devices have been successful, but there are still cases in which PASS is not effective, and the present project seeks to provide science-based guidance for improvements to PASS. One part of this complex problem is the effect of the protective equipment (helmet, eye protection, hood, coat) that is worn by firefighters on hearing. Since this has not previously been studied, it has not been accounted for in the current design of the PASS signal. To address this deficiency, head related transfer function (HRTF) measurements have been taken with a KEMAR acoustic mannequin wearing various combinations of the aforementioned equipment. Results indicate a reduced received level at the ear when the full complement of gear is worn, as might be expected, potentially causing a reduced detection range. In addition, the helmet and eye protection devices cause significant disruption of the normal HRTF patterns, which could potentially interfere with localization. [Work supported by U.S. Department of Homeland Security Assistance to Firefighters Grants Program.]
Back scattering attenuators (silencers)
Authors: Rosenhouse Giora
Published: May 2013

Back scattering silencers are a kind of “sonic crystals.” The frequency range of “sonic crystals” modeling includes the whole audio range and ultrasound, up to phonon waves. Here we concentrate on applications in the audio domain. However, the present investigation is applicable at the higher frequencies as welļ because of the high scalability of the system. The paper defines and analyses 2-D and 3-D back scattering silencers, made of arrays of rigid or soft obstacles (cylinders, spheres or prolate/oblate spheroids for example), in order to attenuate plane waves by multiple scattering along a wave guide. This effect is strong especially at certain band-gaps along the frequency domain. Each obstacle reflects secondary waves that are partially reflected. Thus, along each row of the array, the sound waves lose a certain amount of the energy that adds to the total amount of attenuation. Specifically, the paper analyses back scattering silencers built of meshes of either cylinders or prolate spheroids, where each obstacle is located symmetrically within a fluid cell and each cell is identical to the others.