bluuebunny/crossref_metadata_2025 huggingface dataset:
Vowels and intelligibility in dysarthric speech
Authors: Neel Amy T.
Published: October 2014

Diane Kewley-Port’s work in vowel perception under challenging listening conditions and in the relation between vowel perception and production in second language learners has important implications for disordered speech. Vowel space area has been widely used as an index of articulatory working space in speakers with hypokinetic dysarthria related to Parkinson disease (PD), with the assumption that a larger vowel space is associated with higher speech intelligibility. Although many studies have reported acoustic measures of vowels in Parkinson disease, vowel identification and transcription tasks designed to relate changes in production with changes in perception are rarely performed. This study explores the effect of changes in vowel production by six talkers with PD speaking at habitual and loud levels of effort on listener perception. The relation among vowel acoustic measures (including vowel space area and measures of temporal and spectral distinctiveness), vowel identification scores, speech intelligibility ratings, and sentence transcription accuracy for speakers with dysarthria will be discussed.
Vowels in clear and conversational speech: Within-talker variability in acoustic characteristics
Authors: Ferguson Sarah H. , Rogers Lydia R.
Published: October 2014

The Ferguson Clear Speech Database was developed for the first author’s doctoral dissertation, which was directed by Diane Kewley-Port at Indiana University. While most studies using the Ferguson Database have examined variability among the 41 talkers, the present investigation considered within-talker differences. Specifically, this study examined the amount of variability each talker showed among the 7 tokens of each of 10 vowels produced in clear versus conversational speech. Steady-state formant frequencies have been measured for 5740 vowels in /bVd/ context using Praat, and a variety of measures of spread will be used to determine variability for each vowel in each speaking style for each talker. Results will be compared to those of the only known previous study that included a sufficiently large number of tokens for this type of analysis, an unpublished thesis from 1980. Based on this study, we predict that within-token variability will be smaller in clear speech than in conversational speech.
Understanding speech from partial information: The contributions of consonants and vowels
Authors: Fogerty Daniel
Published: October 2014

In natural listening environments, speech is commonly interrupted by background noise. These environments require the listener to extract meaningful speech cues from the partially preserved acoustic signal. A number of studies have now investigated the relative contribution of preserved consonant and vowel segments to speech intelligibility using an interrupted speech paradigm that selectively preserves these segments. Results have demonstrated that preservation of vowel segments results in greater intelligibility for sentences compared to consonant segments, especially after controlling for preserved duration. This important contribution from vowels is specific to sentence contexts and appears to result from suprasegmental acoustic cues. Converging evidence from acoustic and behavioral investigations suggests that these cues are primarily conveyed through temporal amplitude modulation of vocalic energy. Additional empirical evidence suggests that these temporal cues of vowels, conveying the rhythm and stress of speech, are important for interpreting global linguistic cues about the sentence, such as involved in syntactic processing. In contrast, consonant contributions appear to be specific to lexical access regardless of the linguistic context. Work testing older adults with normal and impaired hearing demonstrates their preserved sensitivity to contextual cues conveyed by vowels, but not consonants. [Work supported by NIH.]
Consonant recognition in noise for bilingual children with simulated hearing loss
Authors: Nishi Kanae , Trevino Andrea C. , Rosado Rogers Lydia , Garcia Paula B. , Neely Stephen T.
Published: October 2014

The negative impacts of noisy listening environments and hearing loss on speech communication are known to be greater for children and non-native speakers than adult native speakers. Naturally, the synergistic influence of listening environment and hearing loss is expected to be greater for bilingual children than their monolingual or normal-hearing peers, but limited studies have explored this issue. The present study compared the consonant recognition performance of highly fluent school-age Spanish-English bilingual children to that of monolingual English-speaking peers. Stimulus materials were 13 English consonants embedded in three symmetrical vowel-consonant-vowel (VCV) syllables. To control for variability in hearing loss profiles, mild-to-moderate sloping sensorineural hearing loss modeled after Pittman & Stelmachowicz [Ear Hear 24, 198–205 (2003)] was simulated following the method used by Desloge et al. [Trends Amplification 16(1), 19–39 (2012)]. Listeners heard VCVs in quiet and in the background of speech-shaped noise with and without simulated hearing loss. Overall performance and the recognition of individual consonants will be discussed in terms of the influence of language background (bilingual vs. monolingual), listening condition, simulated hearing loss, and vowel context. [Work supported by NIH.].
The contribution of vowels to auditory-visual speech recognition and the contributions of Diane Kewley-Port to the field of speech communication
Authors: Richie Carolyn
Published: October 2014

Throughout her career, Diane Kewley-Port has made enduring contributions to the field of Speech Communication in two ways—through her research on vowels and through her mentoring. Diane has contributed greatly to current knowledge about vowel acoustics, vowel discrimination and identification, and the role of vowels in speech recognition. Within that line of research, Richie & Kewley-Port (2008) investigated the effects of visual cues to vowels on speech recognition. Specifically, we demonstrated that an auditory-visual vowel-identification training program benefited sentence recognition under difficult listening conditions more than consonant-identification training and no training. In this presentation, I will describe my continuing research on the relationship between auditory-visual vowel-identification training and listening effort, for adults with normal hearing. In this study, listening effort was measured in terms of response time and participants were tested on auditory-visual sentence recognition in noise. I will discuss the ways that my current work has been inspired by past research with Diane, and how her mentoring legacy lives on.
Individual differences in sensory and cognitive processing across the adult lifespan
Authors: Humes Larry E.
Published: October 2014

A recent large-scale (N = 245) cross-sectional study of threshold sensitivity and temporal processing in hearing, vision and touch for adults ranging in age from 18 through 82 years of age questioned the long-presumed link between aging and declines in cognitive-processing [Humes, L.E., Busey, T.A., Craig, J. & Kewley-Port, D. (2013). Attention, Perception and Psychophysics, 75, 508–524]. The results of this extensive psychophysical investigation suggested that individual differences in sensory processing across multiple tasks and senses drive individual differences in cognitive processing in adults regardless of age. My long-time colleague at IU, Diane Kewley-Port, was instrumental in the design, execution and interpretation of results for this large study, especially for the measures of auditory temporal processing. The methods used and the results obtained in this study will be reviewed, with a special emphasis on the auditory stimuli and tasks involved. The potential implications of these findings, including possible interventions, will also be discussed. Finally, future research designed to better evaluate the direction of the association between sensory-processing and cognitive-processing deficits will be described. [Work supported, in part, by research grant R01 AG008293 from the NIA.]
Tracking unmanned aerial vehicles using a tetrahedral microphone array
Authors: Goldman Geoffrey H. , Culver R. L.
Published: October 2014

Unmanned Aerial Vehicles (UAVs) present a difficult localization problem for traditional radar systems due to their small radar cross section and relatively slow speeds. To help address this problem, the U.S. Army Research Laboratory (ARL) is developing and testing acoustic-based detection and tracking algorithms for UAVs. The focus has been on detection, bearing and elevation angle estimation using either minimum mean square error or adaptive beamforming methods. A model-based method has been implemented which includes multipath returns, and a Kalman filter has been implemented for tracking. The acoustic data were acquired using ARL's tetrahedral microphone array against several UAV's. While the detection and tracking algorithms perform reasonably well, several challenges remain. For example, interference from other sources resulted in lower signal to interference ratio (SIR), which can significantly degrade performance. The presence of multipath nearly always results in greater variance in elevation angle estimates than in bearing angle estimates.
An ultrasonic echo characterization approach based on particle swarm optimization
Authors: Pedrycz Adam , Valero Henri-Pierre , Hori Hiroshi , Nishimiya Kojiro , Sugiyama Hitoshi , Sakata Yoshino
Published: October 2014

Presented is a hands-free approach for the extraction and characterization of ultrasonic echoes embedded in noise. By means of model-based nondestructive evaluation approaches, echoes can be represented parametrically by arrival time, amplitude, frequency, etc. Inverting for such parameters is a non-linear task, usually employing gradient-based, least-squared minimization such as Gauss-Newton (GN). To improve inversion stability, suitable initial echo parameter guesses are required which may not be possible under the presence of noise. To mitigate this requirement, particle swarm optimization (PSO) is employed in lieu of GN. PSO is a population-based optimization technique wherein a swarm of particles explores a multidimensional search space of candidate solutions. Particles seek out the global optimum by iteratively moving to improve their position by evaluating their individual performance as well as that of the collective. Since the inversion problem is non-linear, multiple suboptimal solutions exist, and in this regard PSO has a much lower propensity of becoming trapped in a local minima compared to gradient-based approaches. Due to this, it is possible to omit initial guesses and utilize a broad search range instead, which becomes far more trivial. Real pulse-echoes were used to evaluate the efficacy of the PSO approach under varying noise severity. In all cases, PSO characterized the echo correctly while GN required an initial guess within 30% of the true value to converge.
Beamspace compressive spatial spectrum estimation on large aperture acoustic arrays
Authors: Edelmann Geoffrey F. , Rogers Jeffrey S. , Means Steve L.
Published: October 2014

For large aperture sonar arrays, the number of acoustic elements can be quite sizable and thus increase the dimensionality of the l1 minimization required for compressive beamforming. This leads to high computational complexity that scales by the cube of the number of array elements. Furthermore, in many applications, raw sensor outputs are often not available since computation of the beamformer power is a common initial processing step performed to reduce subsequent computational and storage requirements. In this paper, a beamspace algorithm is presented that computes the compressive spatial spectrum from conventional beamformer output power. Results from CALOPS-07 experiment will be presented and shown to significantly reduce the computational load as well as increase robustness when detecting low SNR targets. [This work was supported by ONR.]
Improvement of the histogram in the degenerate unmixing estimation technique algorithm
Authors: Mukae Junpei , Ishida Yoshihisa , Murakami Takahiro
Published: October 2014

A method of improving the histogram in the degenerate unmixing estimation technique (DUET) algorithm is proposed. The DUET algorithm is one of the methods of blind signal separation (BSS). The BSS framework is to retrieve source signals from mixtures of them without a priori information about the source signals and the mixing process. In the DUET algorithm, the histogram of both the direction-of-arrivals (DOAs) and the distances is formed from the mixtures which are observed using two sensors. And then, signal separation is achieved using time-frequency masking based on the histogram. Consequently, the capability for the DUET algorithm strongly depends on the performance of the histogram. In general, the histogram is degraded by the reverberation or the reflection of the source signals when the DUET algorithm is applied in the real environment. Our approach is to improve the histogram of the DUET algorithm. In our method, the phase component of the observed mixture at each frequency bin is modified by those at the neighboring frequency bins. The proposed method gives the sharper histogram in comparison with the conventional approach.
Start point estimation of a signal in a frame
Authors: Ota Anri , Ishida Yoshihisa , Murakami Takahiro
Published: October 2014

An algorithm for start-point estimation of a signal from a frame is presented. In many applications of speech signal processing, the signal to be processed is often segmented into several frames, and then the frames are categorized into speech and non-speech frames. Instead, we focus on only the frame in which the speech starts. To simplify the problem, we assume that the speech is modeled by a number of complex sinusoidal signals. When a complex sinusoidal signal that starts in a frame is observed, it can be modeled as multiplication of a complex sinusoidal signal of which the length is infinite and a window function that has finite duration in the time domain. In the frequency domain, the spectrum of the signal of the frame is given by the shifted spectrum of the window function. Sharpness of the spectrum of the window function depends on the start point of the signal. Hence, the start point of the signal is estimated by the sharpness of the observed spectrum. This approach can be extended to the signal that consists of a number of complex sinusoidal signals. Simulation results using artificially generated signals show the validity of our method.
Examination and development of numerical methods and algorithms designed for the determination of an enclosure’s acoustical characteristics via the Schroeder Function
Authors: Possing Miles
Published: October 2014

A case study was conducted to measure the acoustical properties of a church auditorium. While modeling the project using EASE 2.1, some problems arose when attempting to determine the reverberation time using the Schroder Back Integrated Impulse Function within EASE 2.1. An auxiliary investigation was launched aiming to better understand the Schroeder algorithm in order to produce a potentially improved version in MATLAB. It was then theorized that the use of a single linear regression is not sufficient to understand the nature of the decay, due to the non-linearity of the curve, particularly during the initial decay. Rather, it is hypothesized that the use of numerical methods to find instantaneous rates of change over the entire initial decay along with a Savitsky-Golay Filter could possibly yield much more robust, accurate results when attempting to derive the local reverberation time from reflectogram data.
A modified direction-of-arrival estimation algorithm for acoustic vector sensors based on Unitary Root-MUSIC
Authors: Shen Junyuan , Li Wei , Guo Yuanming , Chen Yongjue
Published: October 2014

A novel method applying for direction-of-arrival(DOA) using acoustic vector sensors(AVS) based on Unitary Root-MUSIC algorithm(URM) is proposed in this paper. AVS array has a characteristic named coherence principle of sound pressure and velocity, which can significantly improve the detection performance of DOA by reducing the influence of white Gaussian noise. We apply this characteristic and the extra velocity information of AVS to construct a modified covariance matrix. In particular, the modified covariance matrix need not extend the dimension in calculation of AVS covariance matrix which means saving the computing time. In addition, we combine the characteristics of modified matrix with URM algorithm to design a new algorithm, which can minimize the impact of environment noise and further reduce computational complexity to a lower order of magnitude. So the proposed method can not only improve the accuracy of DOA detection but also reduce the computational complexity, compared to the classic DOA algorithm. Theory analysis and simulation experiment show that the proposed algorithm for AVS based on URM can significantly improve the DOA resolution in low SNR ratios and few snapshots.
Evaluating microphones and microphone placement for signal processing and automatic speech recognition of teacher-student dialog
Authors: Brady Michael C. , D'Mello Sydney , Blanchard Nathan , Olney Andrew , Nystrand Martin
Published: October 2014

We evaluate a variety of audio recording techniques for a project on the automatic analysis of speech dialog in middle school and high school classrooms. In our scenario, the teacher wears a headset microphone or a lapel microphone. A second microphone is then used to collect speech and related sounds from students in the classroom. Various boundary microphones, omni-directional microphones, and cardioid microphones are tested as this second classroom microphone. A commercial microphone array [Microsoft Xbox Kinect] is also tested. We report on how well digital source-separation techniques work for segregating the teacher and student speech signals from one another based on these various microphones and placements. We also test the recordings using various automatic speech recognition engines for word recognition error rates under different levels of background noise. Preliminary results indicate one boundary microphone, the Crown PZM-30, to be superior for the classroom recordings. This is based on its performance at capturing near and distant student signals for ASR in noisy conditions, as measured by ASR error rates across different ASR engines.
Strawman outline for a standard on the use of passive acoustic towed arrays for marine mammal monitoring and mitigation
Authors: Thode Aaron
Published: October 2014

There is a perceived need from several U.S. federal agencies and departments to develop consistent standards for how passive acoustic monitoring (PAM) for marine mammals is implemented for mitigation and regulatory monitoring purposes. The use of towed array technology is already being required for geophysical exploration activities in the Atlantic Ocean and the Gulf of Mexico. However, to date no specific standards have been developed or implemented for towed arrays. Here, a strawman outline for a ANSI standard is presented (http://wp.me/P4j34t-a) to cover requirements and recommendations for the following aspects of towed array operations: initial planning (including guidelines for when PAM is not appropriate), hardware, software, and operator training requirements, real-time mitigation and monitoring procedures, and required steps for performance validation. The outline scope, at present, does not cover operational shutdown decision criteria, sound source verification, or defining the required detection range of the system. Instead of specifying details of towed array systems, the current strategy is to focus on the process of defining the required system performance for a given application, and then stepping through how the system hardware, software, and operations should be selected and validated to meet or exceed these requirements. [Work supported by BSEE.]
Towards a standard for the measurement of underwater noise from impact pile driving in shallow water
Authors: Dahl Peter H. , Theobald Pete D. , Robinson Stephen P.
Published: October 2014

Measurements of the underwater noise field from impact pile driving are essential to the address environmental regulations in effect in both Europe and North America to protect marine life. For impact pile driving in shallow water there exists a range scale R* = H/tan(Θ) that delineates important features in the propagation of underwater sound from impact pile driving, where Θ is the Mach angle of the wavefront radiated into the water from the pile and H is water depth. This angle is about 17o for many steel piles typically used, and thus R* is approximately 3H. For range R, such that R/R* ~ 0.5 depth variation in the noise field is highest, more so with peak pressure than with sound exposure level (SEL); for R/R* > 1 the field becomes more uniform with depth. This effect of measurement range can thus have implications on environmental monitoring designed to obtain a close-range datum, which is often used with a transmission loss model to infer the noise level at farther range. More consistent results are likely obtained if the measurement range is at least 3H. Ongoing standardization activities for the measurement and reporting of sound levels radiated from impact pile driving will also be discussed.
Importance of metrics standardization involving the effects of sound on fish
Authors: Halvorsen Michele B.
Published: October 2014

Reporting accurate metrics while employing good measurement practices is a topic that is gaining awareness. Seemingly a simple and expected task, however when reading current and past literature, reporting sound metrics utilized is often not met. It is clear that increased awareness and development of standardization of acoustic metrics is necessary. When reviewing previously published literature on the effects of sound on fish, it is often difficult to fully understand how metrics were calculated leaving the reader to make assumptions. Furthermore, the lack of standardization and definition decreases the amount of data and research studies that could be directly comparable. In a field that has paucity of effects of sound on fish, this situation underscores the importance and need for standardization.
Developments in standards and calibration methods for hydrophones and electroacoustic transducers for underwater acoustics
Authors: Robinson Stephen P. , Foote Kenneth G. , Theobald Pete D.
Published: October 2014

If they are to be meaningful, underwater acoustic measurements must be related to common standards of measurement. In this paper, a description is given of the existing standards for the calibration of hydrophones and electroacoustic transducers for underwater acoustics. The description covers how primary standards are currently realized and disseminated, and how they are validated by international comparisons. A report is also given of the status of recent developments in specification standards, for example within the International Organization for Standardization (ISO) and the International Electrotechnical Commission (IEC). The discussion focuses on the revision of standards for transducer calibration, and the inclusion of extended guidance on uncertainty assessment, and on the criteria for determining the locations of the acoustic near-field and far-field. A description is then provided of recent developments using non-traditional techniques such as optical sensing, which may lead to the next generation of standards. A report is also given of the status of recent developments in and of a number of current initiatives to promote best measurement practice.
All clicks are not created equally: Variations in high-frequency acoustic signal parameters of the Amazon river dolphin (Inia geoffrensis)
Authors: Trone Marie , Balestriero Randall , Glotin Hervé , David Bonnett E.
Published: October 2014

The quality and quantity of acoustical data available to researchers are rapidly increasing with advances in technology. Recording cetaceans with a 500 kHz sampling rate provides a more complete signal representation than traditional sampling at 96 kHz and lower. Such sampling provides a profusion of data concerning various parameters, such as click duration, inter-click intervals, frequency, amplitude, and phase. However, there is disagreement in the literature in the use and definitions of these acoustic terms and parameters. In this study, Amazon River dolphins (Inia geoffrensis) were recorded using a 500 kHz sampling rate in the Peruvian Amazon River watershed. Subsequent spectral analyses, including time waveforms, fast Fourier transforms and wavelet scalograms, demonstrate acoustic signals with differing characteristics. These high frequency, broadband signals are compared, and differences are highlighted, despite the fact that currently an unambiguous way to describe these acoustic signals is lacking. The need for standards in cetacean bioacoustics with regard to terminology and collection techniques is emphasized.
Acoustical terminology in the Sonar Modelling Handbook
Authors: Holden Andrew
Published: October 2014

The UK Sonar Modelling Handbook (SMH) defines the passive and active Sonar Equations, and their individual terms and units, which are extensively used for sonar performance modelling. The new Underwater Acoustical Terminology ISO standard, which is currently being developed by the ISO working group TC43/SC3/WG2 to standardize terminology will have an impact on the SMH definitions. Work will be presented comparing the current SMH terminology with both the future ISO standard and other well-known definitions to highlight the similarities and differences between each of these.
The definitions of “level,” “sound pressure,” and “sound pressure level” in the International System of Quantities, and their implications for international standardization in underwater acoustics
Authors: Ainslie Michael A.
Published: October 2014

The International System of Quantities (ISQ), incorporating definitions of physical quantities and their units, was completed in 2009 following an extensive collaboration between two major international standards organizations, the International Organization for Standardization (ISO) and the International Electrotechnical Commission (IEC). The ISQ encompasses all SI units as well as selected units outside the SI such as the byte (including both decimal and binary multiples), bel, neper, and decibel. The ISQ, which includes definitions of the terms “level,” “sound pressure,” and “sound pressure level,” is presently being used to underpin an underwater acoustics terminology standard under development by ISO. For this purpose, pertinent ISQ definitions are analyzed and compared with alternative standard definitions, and with conventional use of the same terms. The benefits of combining IEC and ISO definitions into a single standard, solving some long-standing problems, are described. The comparison also reveals some teething problems, such as internal inconsistencies within the ISQ, and discrepancies with everyday use of some of the terms, demonstrating the need for continued collaboration between the major standards bodies. As of 2014, the ISQ is undergoing a major revision, leading to a unique opportunity to resolve these discrepancies.
From the sound up: Reverse-engineering room shapes from sound signatures
Authors: Boning Willem , Bassuet Alban
Published: October 2014

Typically, architects and acousticians design rooms for music starting from a model room shape known from past experience to perform well acoustically. We reverse the typical design process by using a model sound signature to generate room shapes. Our method builds off previous research on reconstructing room shapes from recorded impulse responses, but takes an instrumental, design-oriented approach. We demonstrate how an abstract sound signature constructed in a hybrid image source-statistical acoustical simulator can be translated into a room shape with the aid of a parametric design interface. As a proof of concept, we present a study in which we generated a series of room shapes from the same sound signature, analyzed them with commercially available room acoustic software, and found objective parameters for comparable receiver positions between shapes to be within just-noticeable-difference ranges of each other.
Achieving acoustical comfort in restaurants
Authors: Battaglia Paul
Published: October 2014

The achievement of a proper acoustical ambiance for restaurants has long been described as a problem of controlling noise to allow for speech intelligibility among patrons at the same table. This simplification of the acoustical design problem for restaurants does not entirely result in achieving either a sensation of acoustical comfort or a preferred condition for social activity sought by architects. In order to more fully study the subjective impression of acoustical comfort a large data base from 11 restaurants with 75 patron surveys for each (825 total) was assembled for analysis. The results indicate that a specific narrow range of reverberation time can produce acoustical comfort for restaurant patrons of all ages. Other physical and acoustical conditions of the dining space are shown to have little to no consistent effect on the registration of comfort. The results also indicate that different subjective components of acoustical comfort—quietude, communication, privacy—vary significantly by age group with specific consequences for the acoustical design of restaurants for different clienteles.
500-seat theater in the city of Qom; Computer simulation vs. acoustics measurements
Authors: Azad Hassan
Published: October 2014

There is an under construction 500-seat Theater in Qom city in Iran in which I was part of the acoustics design team. We went through a different steps of the acoustics design using Odeon software packages which enabled us to go back and forth in design process and make proper improvement while we were suffering from having limitations on material choice. Fortunately the theater is being built so after a while it would be feasible to do acoustics measurements with the help of Building and Housing Research Center (BHRC) in Iran as well as subjective evaluation during the very first performances. This paper is aimed to juxtapose the results of computer simulation and acoustics measurement and make a comparison in between to see if there are any discrepancies.
Visualization of auditory masking for firefighter alarm detection
Authors: Farmer Casey , Abbasi Mustafa Z. , Wilson Preston S. , Ezekoye Ofodike A.
Published: October 2014

An essential piece of firefighter equipment is the Personal Alert Safety System (PASS), which emits an alarm when a firefighter has been inactive for a specified period of time and is used to find and rescue downed firefighters. The National Institute for Occupational Safety and Health (NIOSH) firefighter fatality reports suggest that there have been instances when the PASS alarm is not audible by other firefighters on the scene. This paper seeks to use acoustic models to measure the sound pressure level of various signals throughout a structure. With this information, a visual representation will be created to map where a PASS alarm is audible and where it is masked by noise sources. This paper presents an initial audibility study, including temporal masking and frequency analysis. The results will be compared to auralizations and experimental data. Some other potential applications will be briefly explored.
Investigations on acoustical coupling within single-space monumental structures using a diffusion equation model
Authors: Sü Gül Zühre , Xiang Ning , Çalışkan Mehmet
Published: October 2014

Sound energy distributions and flows within single-space rooms can be exploited to understand the occurrence of multi-slope decays. In this work, a real-size monumental worship space is selected for investigations of non-exponential sound energy decays. Previous field tests in this single-space venue indicate multi-slope formation within such a large volume and the multiple-dome upper structure layout. In order to illuminate/reveal the probable reasons of non-exponential sound energy decays within such an architectural venue, sound energy distributions and energy flows are investigated. Due to its computational efficiency and advantages in spatial energy density and flow vector analysis, a diffusion equation model (DEM) is applied for modeling sound field of the monumental worship space. Preliminary studies indicate good agreement for overall energy decay time estimations among experimental field and DEM results. The energy flow vector and energy distribution analysis indicate the upper central dome-structure to be the potential energy accumulation/ concentration zone, contributing to the later energy decays.
History of high intensity focused ultrasound, Bill and Frank Fry and the Bioacoustics Research Laboratory
Authors: O'Brien William , Dunn Floyd
Published: October 2014

1946 is a key year in the history of HIFU. That year, sixty-eight years ago, the Bioacoustics Research Laboratory was established at the University of Illinois. Trained in theoretical physics, William J, (Bill) Fry (1918–1968) left his graduate studies at Penn State University to work at the Naval Research Laboratory in Washington, DC on underwater sound during World War II. Bill was hired by the University of Illinois in 1946, wanting to continue to conduct research activities of his own choosing in the freer university atmosphere. Like Bill, Francis J. (Frank) Fry (1920–2005) went to Penn State as well as the University of Pittsburgh where he studied electrical engineering. Frank joined Bill at the University of Illinois, also in 1946, having worked at Westinghouse Electric Corporation where his division was a prime contractor on the Manhattan Project. Floyd Dunn also arrived at the University of Illinois in 1946 as an undergraduate student, having served in the European Theater during World War II. The talk will recount some of the significant HIFU contributions that emerged from BRL faculty, staff, and students. [NIH Grant R37EB002641.]
Transforming ultrasound basic research in to clinical systems
Authors: Sanghvi Narendra T. , Franklin Thomas D.
Published: October 2014

In late 1960s, Robert F. Heimburger, MD, Chief of Neurosurgery at Indiana University School of Medicine, started collaborating with William J. Fry and Francis J. Fry at Interscience Research Institute (IRI) in Champaign, IL. and treated brain cancer patients with HIFU. In 1970, Dr. Heimburger and Indiana University School of Medicine (IUMS) invited IRI to join IUMS and Indianapolis Center For Advanced Research, Inc. (ICFAR). In 1972, a dedicated Fortune Fry Research Laboratory (FFRL) was inaugurated to advance ultrasound research relevant for clinical use. In the ‘70s, an automated computer controlled, integrated B-mode, image-guided HIFU system (“the candy machine”) was developed that successfully treated brain cancer patients at IUMS. HIFU was found to be safe for the destruction of brain tumors. Later a second-generation brain HIFU device was developed to work with CAT or MR images. In 1974, the FFRL developed a first cardiac real-time, 2-D ultrasound scanner. Prof. H. Feigenbaum pioneered this imaging technique and formed “Echocardigraphy Society.” In 1978, an automated breast ultrasound system was successfully developed led to form Labsonics, Inc. that proliferated 300 scanners in 4 years. In 1986, the Sonablate system to treat prostate cancer was developed. The Sonablate has been used worldwide.
The development of high intensity focused ultrasound in Europe, what could we have done better?
Authors: ter Haar Gail
Published: October 2014

The clinical uptake of HIFU has been disappointingly slow. This despite its promise as a minimally invasive, ultimately conformal technique. It may be instructive to look at the way in which this technique has evolved from its early days with an eye as to whether a different approach might have resulted in its more rapid acceptance. Examples will be drawn from HIFU’s development in the United Kingdom.
The development of MRI-guided focused ultrasound at Brigham & Women’s Hospital
Authors: McDannold Nathan
Published: October 2014

The Focused Ultrasound Laboratory was created in the Department of Radiology at Brigham & Women’s Hospital in the early 1990’s, when Ferenc Jolesz invited Kullervo Hynynen to join him to collaborate with GE Medical Systems to develop MRI-guided Focused Ultrasound surgery. This collaboration between Dr. Hynynen, an experienced researcher of therapeutic ultrasound, Dr. Jolesz, who developed MRI-guided laser ablation, and the engineers at GE and later InSightec, with their decades of experience developing MRI and ultrasound systems, established a program that over two decades produced important contributions to HIFU. In this talk, Nathan McDannold, the current director of the laboratory, will review the achievements made by the team of researchers, which include the development of the first MRI-guided FUS system, the creation of the first MRI-compatible phased arrays, important contributions to the validation and implantation of MR temperature mapping and thermal dosimetry, the development of an MRI-guided transcranial system, and the discovery that ultrasound and microbubbles can temporarily disrupt the blood–brain barrier. This output of this team, which led to clinical systems that have treated tens of thousands of patients at sites around the world, is an excellent example of how academic research can be to the clinic.
What have we learned about shock wave lithotripsy in the past thirty years?
Authors: Zhong Pei
Published: October 2014

Shock wave lithotripsy (SWL) has revolutionized the treatment of kidney stone disease since its introduction in the early 1980s. Considering the paucity of knowledge about the bioeffects of shockwaves in various tissues and renal concretions 30 years ago, the success of SWL is a truly remarkable fest on its own. We have learned a lot since then. New technologies have been introduced for shock wave generation, focusing, and measurement, among others. In parallel, new knowledge has been acquired progressively about the mechanisms of stone comminution and tissue injury. Yet there are still outstanding issues that are constantly debated, waiting for resolution. In this talk, the quest for a better understanding of the shockwave interaction with stones and renal tissue in the field of SWL will be reviewed in chronological order. Focus will be on stress waves and cavitation for their distinctly different (for their origin), yet often synergistically combined (in their action), roles in the critical processes of SWL. This historical review will be followed by a discussion of the recent development and future prospects of SWL technologies that may ultimately help to improve the clinical performance and safety of contemporary shock wave lithotripters. [Work supported by NIH through 5R37DK052985-18.]
Educating mechanical engineers in the art of noise control
Authors: Hansen Colin
Published: October 2014

Acoustics and noise control is one of the disciplines where the material that students learn during a well-structured undergraduate course, can be immediately applied to many problems that they may encounter during their employment. However, in order to find optimal solutions to noise control problems, it is vitally important that students have a good fundamental understanding of the physical principles underlying the subject as well as a good understanding of how these principles may be applied in practice. Ideally, they should have access to affordable software and be confident in their ability to interpret and apply the results of any computer-based modelling that they may undertake. Students must fully understand any ethical issues that may arise, such as their obligation to ensure their actions do not contribute to any negative impact on the health and welfare of any communities. How do we ensure that our mechanical engineering graduates develop the understanding and knowledge required to tackle noise control problems that they may encounter after graduation? This presentation attempts to answer this question by discussing the process of educating undergraduate and postgraduate mechanical engineering students at the University of Adelaide, including details of lab classes, example problems, text books and software developed for the dual purpose of educating students and being useful in assisting graduates solve practical noise control problems.
Advanced methods of signal processing in acoustics
Authors: Culver R. Lee , Xiang Ning
Published: October 2014

Signal processing is applied in virtually all areas of modern acoustics to extract, classify, and/or quantify relevant information from acoustic measurements. Methods range from classical approaches based on Fourier and time-frequency analysis, to array signal processing, feature extraction, computational auditory scene analysis, and Bayesian inference, which incorporates physical models of the acoustic system under investigation together with advanced sampling techniques. This talk highlights new approaches to signal processing recently applied in a broad variety of acoustical problems.
Nearfield acoustic measurements of triggered lightning using a one-dimensional microphone array
Authors: Dayeh Maher A. , Evans Neal
Published: October 2014

For the first time, acoustic signatures from rocket-triggered lightning are measured by a 15 m long, one-dimensional microphone array consisting of 16 receivers, situated 79 m from the lightning channel. Measurements were taken at the International Center for Lightning Research and Testing (ICLRT) in Camp Blanding, FL, during the summer of 2014. We describe the experimental setup and report on the first observations obtained to date. We also discuss the implications of these novel measurements on the thunder initiation process and its energy budget during lightning discharges. Challenges of obtaining measurements in these harsh ambient conditions and their countermeasures will also be discussed.
Sonic boom noise exposure inside homes
Authors: Klos Jacob
Published: October 2014

Commercial supersonic overland flight is presently banned both nationally and internationally due to the sonic boom noise that is produced in overflown communities. However, within the next decade, NASA and industry may develop and demonstrate advanced supersonic aircraft that significantly mitigate the noise perceived at ground level. To allow commercial operation of such vehicles, bans on commercial supersonic flight must be replaced with a noise-based certification standard. In the development of this standard, variability in the dose-response model needs to be identified. Some of this variability is due to differing sound transmission characteristics of homes both within the same community and among different communities. A tool to predict the outdoor-to-indoor low-frequency noise transmission into homes has been developed at Virginia Polytechnic Institute and State University, which was used in the present study to assess the indoor exposure in two communities representative of the northern and southern United States climate zones. Sensitivity of the indoor noise level to house geometry and material properties will be discussed. Future plans to model the noise exposure variation among communities within the United States will also be discussed.
Evaluation of the effect of aircraft size on indoor annoyance caused by sonic booms
Authors: Loubeau Alexandra
Published: October 2014

Sonic booms from recently proposed supersonic aircraft designs developed with advanced tools are predicted to be quieter than those from previous designs. The possibility of developing a low-boom flight demonstration vehicle for conducting community response studies has attracted international interest. These studies would provide data to guide development of a preliminary noise certification standard for commercial supersonic aircraft. An affordable approach to conducting these studies suggests the use of a sub-scale experimental aircraft. Due to the smaller size and weight of the sub-scale vehicle, the resulting sonic boom is expected to contain spectral characteristics that differ from that of a full-scale vehicle. To determine the relevance of using a sub-scale aircraft for community annoyance studies, a laboratory study was conducted to verify that these spectral differences do not significantly affect human response. Indoor annoyance was evaluated for a variety of sonic booms predicted for several different sizes of vehicles. Previously reported results compared indoor annoyance for the different sizes using the metric Perceived Level (PL) at the exterior of the structure. Updated results include analyses with other candidate noise metrics, nonlinear regression, and specific boom duration effects.
Effects of secondary rattle noises and vibration on indoor annoyance caused by sonic booms
Authors: Rathsam Jonathan
Published: October 2014

For the past 40 years, commercial aircraft have been banned from over-land supersonic flight due to the annoyance caused by sonic booms. However, advanced aircraft designs and sonic boom prediction tools suggest that significantly quieter sonic booms may be achievable. Additionally, aircraft noise regulators have indicated a willingness to consider replacing the ban with a noise-based certification standard. The outdoor noise metric used in the certification standard must be strongly correlated with indoor annoyance. However, predicting indoor annoyance is complicated by many factors including variations in outdoor-to-indoor sound transmission and secondary indoor rattle noises. Furthermore, direct contact with vibrating indoor surfaces may also affect annoyance. A laboratory study was recently conducted to investigate candidate noise metrics for the certification standard. Regression analyses were conducted for metrics based on the outdoor and transmitted indoor sonic boom waveforms both with and without rattle noise, andincluded measured floor vibration. Results indicate that effects of vibration are significant and independent of sound level. Also, the presence or absence of rattle sounds in a transmitted sonic boom signal generally changes the regression coefficients for annoyance models calculated from the outdoor sound field, but may not for models calculated from the indoor sound field.
Smoothed particle hydrodynamics simulation of sound reflection and transmission
Authors: Zhang YongOu , Zhang Tao , Ouyang Huajiang , Li TianYun
Published: October 2014

Mesh-based methods are widely used in acoustic simulations nowadays. However, acoustic problems with complicated domain topologies and multiphase systems are difficult to be described with these methods. On the contrary, Smoothed Particle Hydrodynamics (SPH), as a Lagrangian method, does not have much trouble in solving these problems. The present paper aims to simulate the reflection and transmission of sound waves with the SPH method in time domain. Firstly, the linearized acoustic equations are represented in the SPH form by using the particle approximation. Then, one dimensional sound reflection and transmission are simulated with the SPH method and the solutions are compared with the theoretical results. Finally, the effects of smoothing length and neighboring particle numbers on the computation are discussed. The errors of sound pressure, particle velocity, and change of density show that the SPH method is feasible in simulating the reflection and transmission of sound waves. Meanwhile, the relationship between the characteristic impedance and the reflected waves obtained by the SPH simulation is consistent with the theoretical result.