bluuebunny/crossref_metadata_2025 huggingface dataset:
Ultrasonic calibration of MEMS (microelectromechanical systems) microphone arrays.
Authors: Knisely Katherine E. , Singh Victor , Grosh Karl , Dowling David R. , Yonak Serdar H.
Published: March 2010

Ultrasonic microphone arrays have applications in imaging, scale model testing, and sound source localization. For these applications MEMS ultrasonic microphones are emerging as a natural choice because of their small size, low cost, and robustness to manufacturing treatments and environmental conditions. In this talk an array is constructed using commercially available Knowles SiSonic[TM] capacitive MEMS microphones. The absolute sensitivity and phase calibrations of the array elements in the 1–60-kHz frequency range are obtained through the use of replacement calibration and time-frequency analysis. For these results, a modulation in sensitivity and phase measurements for frequency ranges over 15 kHz is observed under certain conditions. The origin of this modulation is discussed as well as various techniques for its removal. Relative element calibrations are crucial for array performance and are obtained through a variety of sound source configurations, which vary in size and location relative to the array.
Comparison of single-frequency monopulse techniques that mimic the results of multiple-frequency, single-aperture interferometry.
Authors: Brogan Daniel S. , Chamberlin Kent A.
Published: March 2010

The capability to resolve targets from each other and from noise is desirable in all target detection and tracking systems. In a previous presentation [G. R. Cutter, Jr. and D. A. Demer, J. Acoust. Soc. Am. 126, 2232 (2009)] it was shown that single-frequency, multiple-aperture interferometry could be used to resolve multiple targets that could not be resolved using single-frequency, single-aperture interferometry. This was accomplished by using the phase differences from several pairs of subarrays with different subarray spacing to mimic the operation of a multiple-frequency, single-aperture interferometric system. The differential-phase technique is one type of phase-comparison monopulse used in target bearing estimation. The current research compares the results of single-frequency, multiple-aperture interferometry to those of other single-frequency phase-comparison and amplitude-comparison monopulse techniques by using multiple receive subarrays and/or varying the receive arrays’ elements’ weights.
The Science and Human Rights Coalition of the AAAS (American Association for the Advancement of Science): The interface between the human rights and scientific communities.
Authors: Wyndham Jessica
Published: March 2010

In July 2005, AAAS held a 2-day conference of scientific associations and human rights organizations to explore ways in which the scientific community could become more directly engaged in human rights. Born of a conference recommendation, the AAAS Science and Human Rights Coalition was officially launched in January 2009. Forty-two scientific membership organizations have since joined the Coalition which aims to build bridges and opportunities for collaboration within the scientific community and between the scientific and the human rights communities. Introducing the Coalition, Ms. Wyndham will answer four questions: What does the Coalition aim to accomplish? How will it do this? Who can join? How does it operate? One year young, the growing number of Coalition members reflects the commitment of scientific organizations to addressing human rights. This, however, is just the beginning. Building on the achievements to-date and lessons learnt throughout the course of the first year, Ms. Wyndham will outline the Coalition’s plans for making a real and meaningful contribution to the realization of human rights.
Novel computer aided design of labial flue pipes.
Authors: Moss Brian , Lewis Elfed , Leen Gabriel , Bremer Kort , Niven Andrew
Published: March 2010

A labial flue pipe is a well known tone generator, which is familiar and easily recognizable as the organ pipes seen in many concert halls and churches. However, the design and understanding of the sounding mechanism of such pipes are fraught with difficulty. Traditionally labial pipes are constructed from age-old lookup tables that are closely guarded intellectual secrets. This paper discusses a novel computer program that facilitates the design and construction of such labial flue pipes. The computer program allows almost all aspects of the labial flue pipe design to be varied, the resultant frequency is generated, and in addition the Ising efficiency number is provided. Furthermore, a discussion is included related to the fact that even though an Ising number greater than 3 indicates that a pipe is overblown, the fundamental tone is still predominant. A comparison will also be made between a CFD simulation of the labial flue pipe jet mechanism and smoke trail plots typical of such analysis.
On the role of acoustics in the Vedic Hindu tradition and philosophy.
Authors: Prasad M. G. , Rajavel B.
Published: March 2010

Acoustics has received a very high importance in the Vedic Hindu tradition and philosophy of ancient India. The Vedic literature which refers to both the Vedas and the subsequent literature based on Vedas emphasize that the roles of sound both as source signal and as hearing are important. This work presents the various ways such as chants, Sanskrit language, vocal, and instrumental music in which acoustics has played a major role.
Inertial cavitation threshold dependence on high static pressures.
Authors: Bader Kenneth, B. , Mobley Joel , Raymond Jason , Gaitan D. Felipe
Published: March 2010

The creation of a bubble in a liquid is energetically favorable when the liquid is subject to a net/an overall tension less than of its saturated vapor pressure. This gain is offset by the creation of a liquid-vapor/gas interface, and only bubbles larger than some critical size will spontaneously grow. The magnitude of the acoustic pressure required to produce a cavity of the critical size is termed the cavitation threshold of the liquid. The dependence of the cavitation threshold on hydrostatic pressure has previously been reported up to 130 bars in terms of electrical power applied to the acoustic driving transducer. These measurements used a standing wave set-up in a stainless steel spherical resonator (24.1 cm outer diameter, 1.9 cm thick) with a Q > 10 000 when fluid loaded. This work will extend into higher pressure regimes and will be used to extrapolate the cavitation threshold at 1 kbar.
Investigation of acoustic streaming in the cochlea.
Authors: Aho Katherine , Vejar Elaine , Thompson Charles
Published: March 2010

In this work we will analyze acoustic streaming in the cochlea. A model will be developed to examine the steady flow induced by the harmonic excitation of a fluid in a rigid two-dimensional waveguide. The strength of the flow will depend on the time-phase between the axial velocity and the boundary velocity. It was found in previous work that the amplitude of the flow is directly related to the streaming Reynolds number. These results will be used to compare with Bekesy’s observations.
Development and validation of a computational model of bone-conducted sound transmission for improved hearing protection design.
Authors: Clavier Odile H. , Wismer Margaret , Wilbur Jed , Dietz Anthony , O’Brien William
Published: March 2010

Bone-conducted sound is the limiting factor in current hearing protection for very high-noise environments such as those encountered by maintainers of military aircraft. The University of Illinois has developed a three-dimensional acoustic wave propagation model for the computation of sound transmission into, around, and through fluid and solid-shell bodies. Creare has implemented experimental techniques to validate the computational results. Three cases are presented. The first is a fluid sphere, for which experimental, computational, and analytical results were obtained. The second is a solid-shell, fluid-filled sphere, for which experimental and computational results were obtained to determine the effect of the solid shell. The third is a representative human head with and without passive hearing protection. For this case, the computational model and an experimental head simulator were both developed from a computed tomography scan of a living human head. The head simulator was built around a rapid-prototype instrumented skull using silicon organs and simulated tissue. Experimental, computational, and analytical results were all in good agreement for the fluid sphere test case. While general agreement was obtained for the other two cases, specific discrepancies in the results are outlined and limitations of the models are discussed.
Diassociating spectral and temporal influences in an AM-QFM (amplitude modulated and quasi-frequency modulated) discrimination task.
Authors: Borucki Ewa , Bruce Berg
Published: March 2010

This study investigated temporal and spectral influences in a task used to investigate the bandwidths of phase sensitivity. Subjects discriminated amplitude modulated (AM) tones and quasi-frequency modulated (QFM) tones in a 2IFC. An adaptive threshold procedure was used to estimate modulation depth needed to make the discrimination as a function modulation frequency for a 2000-Hz carrier. Threshold functions were often nonmonotonic, with nonmonotonicities observed at higher-modulation frequencies. This is likely due to the effects of distortion products creating salient spectral cues. When stimulus duration is decreased from 200 to 50, 20, or 10 ms, thresholds for low-modulation frequencies decreased to near-chance levels, whereas thresholds in the region of nonmonotonicities were largely unaffected. The decrease in stimulus duration appears to hinder the listener’s ability to use temporal cues in order to discriminate between AM and QFM, whereas spectral information derived from distortion product cues appears to be resilient. [Work supported by NSF BCS-07464003.]
Fundamental frequency and pitch shift discrimination.
Authors: Yost William A. , Brown Christopher A. , Walling Farris
Published: March 2010

The pitch-shift of the residue has been used as an argument for the importance of temporal fine structure in pitch perception. Discrimination of a change in fundamental frequency (F0) for harmonic complexes was compared to discriminating a change in the shifted frequency (f) in pitch-shift of the residue stimuli. Patterson and Wightman [J. Acoust. Soc. Am. 59, (1976)] showed that a linear relationship exists between matched pitch and the frequency of the lowest spectral component for pitch shift of the residue stimuli (stimuli for which the spacing between adjacent spectral components is constant, but all components in the complex are shifted up or down in frequency). Obtaining pitch matches for pitch shift of the residue stimuli is a difficult task because of the ambiguous nature of the pitch shift of the residue in several conditions. Using the two discrimination experiments allows for an estimate of the slopes relating matched pitch to the frequency of the lowest component. Slopes estimated from the discrimination experiments will be compared to those obtained by the previous authors to determine if the discrimination experiments provide a valid way to predict the matched pitch of pitch-shift of the residue stimuli. [Research supported by NIDCD.]
Extraction of fundamental pitch using Euclid’s algorithm implemented through iterative demodulation.
Authors: Kumaresan Ramdas , Li Jiun-yee , Cariani Peter
Published: March 2010

Strong periodicity or “virtual” pitches are heard at the fundamentals (F0s) of harmonic complexes. As an alternative to both spectral pattern matching and autocorrelation-like analysis, we propose a novel mechanism that extracts the pitch directly from the signal by iterative demodulation and low-pass filtering. In effect the mechanism computes the greatest common divisor (gcd) of component signal harmonic numbers, using a Euclid-like iterative subtraction algorithm. For example, F0 of 560- and 320-Hz tones is 80 Hz, obtained by iteratively subtracting smaller frequencies from larger ones until they converge, e.g., 560−320=240 Hz, 320−240=80, 240−80=160, and 160−80=80. This algorithm can be adapted to compute pitches of harmonic complexes. Consider two tones, 560 and 320 Hz. As in Euclid’s algorithm, the tones are iteratively multiplied and low-pass filtered to obtain the first (240-Hz), second (160-Hz), and third (80-Hz) difference tones. At the end only F0 remains. For inharmonic tones, incomplete demodulation results if there is a lower-frequency limit to the process. The mechanism converges on one pitch rather than multiple subharmonics. Such repeated multiplication and low-pass operations could conceivably be implemented in the ascending auditory pathway via phase-locked spike trains with synchrony capture, successive cross-CF convergences, low-pass modulation tunings, and cumulative decline of phase-locking at successive stations. [Work supported by AFOSR FA9550-09-1-0119.]
Neural timing nets for fundamental frequency (F0)-based auditory scene analysis.
Authors: Cariani Peter , Kumaresan Ramdas
Published: March 2010

Neural timing nets are idealized networks of delay lines, coincidence detectors, and adaptive processing elements that operate in the time domain on temporally coded signals to compare, extract, and separate auditory objects [Cariani, Neural Networks 14, 737–753 (2001); J. New Music Res. 30, 107–135; IEEE Trans. Neural Netw. 15(5) (2004)]. Timing nets constitute an alternative, potential mode of neural signal processing in which information resides in neural signals rather than in patterns of activated elements. Recurrent timing nets with delay loops act as dense arrays of recursive, comblike filters to effect a period-by-period analysis that builds up and separates component invariant time patterns with different fundamentals (F0s). Using both linear and nonlinear processing rules, the latter were used to process and separate synthetic double vowels, running speech, and polyphonic musical excerpts, with varying results. Relations to, and combinations with, processing strategies based on autocorrelation and all-order interspike intervals, adaptive comb filtering, correlogram duplex analysis, cancellation, Fourier zero-crossings, and demodulation, with and without prior bandpass filtering, are discussed. Bottom-up/top-down mechanisms for dynamic facilitation of lower level temporal processing loops are also considered. [Work supported by AFOSR FA9550-09-1-0119.]
Vocal pitch regulation depends on the baseline voice F0 feedback: An ERP study for investigating the role of auditory feedback for voice pitch error correction.
Authors: Behroozmand Roozbeh , Larson Charles
Published: March 2010

Voice production and control require neural interactions between the vocal motor and auditory mechanisms. The comparison between the incoming auditory feedback and the predicted sensory input (efference copies) from a self-produced vocalization allows the detection of feedback mismatches for voice error detection and correction. However, the sensitivity of the audio-vocal system for feedback mismatch detection seems to depend on the extent of feedback deviation from the predicted vocal output. The present study investigated this effect by examining event-related potentials (ERPs) in response to a +100-cent voice feedback pitch perturbation stimulus while the extent of pre-stimulus (baseline) feedback pitch deviation was randomly manipulated at 0, 50, 100, 200, and 400 cents. Results showed that the neural responses to +100-cent pitch-shift stimuli grew systematically larger as the extent of pre-stimulus baseline pitch deviation became smaller. This finding suggests that the extent of disparity between the predicted and incoming sensory feedback of self-produced voice can affect the neural tuning processes that adjust the sensitivity of the audio-vocal mechanisms for voice pitch error detection and correction. Lower sensitivity to larger feedback mismatches may imply robustness against the disruptive effect of highly deviant or externally generated sounds during vocal production and control.
Molecular analysis of the effect of onset asynchrony in the identification of a rudimentary sound source in noise.
Authors: Lutfi Robert A. , Liu Ching-Ju , Stoelinga Christophe N. J.
Published: March 2010

The threshold for detecting a target in noise is often greater when the target is gated on simultaneously with the noise than when it is gated on after some delay [Zwicker, E. (1965). J. Acoust. Soc. Am. 37, 653–663]. One explanation is that the perceptual principle of grouping causes the target and noise with simultaneous onsets to be perceived as a single sound source. This idea was tested for a rudimentary sound source using perturbation analysis. In a two-interval, forced-choice procedure listeners identified as target the impact sound produced by the larger of two stretched membranes. The noise on each presentation was the impact sound of a variable-sized plate. Grouping predicts that the decision weights on the noise should be positive when target and noise have simultaneous onsets, but that they should approach zero when target and noise are gated on asynchronously. This prediction was confirmed when the noise preceded the target by a fixed interval (100 ms), but not when it followed the target by the same interval or when either interval was selected at random on each presentation (100 or −100). [Research supported by NIDCD Grant 5R01DC006875-05.]
Level-dependent changes in perception of temporal envelope cues.
Authors: Wang Xin , Ahlstrom Jayne B. , Horwitz Amy R. , Dubno Judy R.
Published: March 2010

Level-dependent changes in speech recognition may reveal effects of basilar-membrane nonlinearities on temporal envelope fluctuations. It is hypothesized that, as a result of the compressive effects of the active cochlear mechanism, the “effective” magnitude of speech envelope fluctuations will be reduced as speech level increases from lower (more linear) to conversational (more compressive) regions. With further increases from conversational levels (to a more linear region), temporal envelope fluctuations will become more pronounced. Accordingly, speech recognition will be maximized at conversational levels due to the optimal “flattened” envelope and then decline at lower or higher levels. To test these assumptions, speech recognition scores were measured as a function of level for adults with normal hearing. Speech stimuli were spectrally degraded using “noise vocoder” processing so that perceptual effects of modifications to the speech temporal envelope can be revealed. As vocoded speech level increased, background noise level also increased, maintaining a fixed signal-to-noise ratio to minimize sensation-level effects on speech recognition scores. Discussion will focus on level-dependent effects for different speech stimuli and the role of nonlinearities on perception of temporal envelope cues. [Work supported by NIH/NIDCD]
Auditory channel weights for consonant recognition in normal-hearing listeners.
Authors: Apoux Frederic , Healy Eric W.
Published: March 2010

The present study evaluated the relative contribution of various regions of the frequency spectrum to consonant recognition in normal-hearing listeners. The method used in this study was specifically designed to provide an estimate of the importance of each band (i) consistent with the frequency resolution of the auditory system and (ii) irrespective of the location of information elsewhere in the spectrum. Speech stimuli were divided into 30 adjacent bands with each band corresponding approximately to the bandwidth of an auditory filter. Listeners were presented with a subset of bands to avoid ceiling effects. The importance of each band was derived from the drop in performance observed when that particular band was omitted. The spectral location of the bands was always chosen randomly except for the one band whose importance was being assessed. The results indicated a fairly homogeneous contribution of all 30 bands to consonant recognition (i.e., a flat auditory channel importance function) with only the five lowest bands (below 300 Hz) having lesser weight. In contrast, additional analyses revealed a non-uniform contribution of the bands to the transmission of voicing, manner and place of articulation. [Work supported by NIDCD.]
Noise and vibration transmission reduction using multi-element multi-path structures.
Authors: Palas Cassidy , Bliss Donald
Published: March 2010

Principles of structural acoustics are utilized in novel ways to cancel the transmission of sound and vibration through multi-element flexible barriers. Configurations analyzed include two and three layered plates with elastic interconnections. The substructures have different wave propagation properties and boundary conditions. Both continuous and periodic discrete elastic coupling methods are examined. This research demonstrates that flexibility and controlled resonant behavior can be used to block vibration and sound transmission, even with low structural damping. The main strategies utilized are structural wave cutoff with multi-element multi-path (MEMP) structures and relative phase changes due to boundary reflections. Examples of acoustic transmission loss through panel barriers are presented, and the potential advantages and possible shortcomings of the approach are evaluated. Practical configurations for layered sound reduction materials include designs allowing multiple substructural plates to produce radiation on a given surface, leading to net cancellations of transmitted sound in certain frequency ranges. Experimental results show vibration transmission reduction for several configurations. The work has particular application to the reduction in vehicle interior noise and addresses the need for good acoustic performance of lighter weight flexible structures.
Building information management: An acoustics consultant’s perspective.
Authors: Markham Benjamin
Published: March 2010

Architecture and engineering firms use building information management (BIM) software applications increasingly in the design process; similarly, building owners create BIM models for existing facilities both for internal use and to aid future tenants, facilities managers, and designers in planning for maintenance programs, restoration, or renovation. As for other design and engineering disciplines, these tools provide powerful opportunities to the acoustics community. The primary benefits to architectural acoustics consultants fall into two broad categories: first, in the construction process by providing a means of organizing the specifications for materials and constructions, and second, in the design process, by establishing a framework of data and acoustical attributes that in the future could potentially aid in acoustical calculation and analysis.
Building information modeling with revit architecture and acoustical coordination.
Authors: LePage Katherine
Published: March 2010

The adoption of BIM has provided a huge shift within the architectural design community. It provides an opportunity for more thoughtfully designed buildings as well as comprehensive coordination with consultants. This presentation will cover the basics of how programs like revit architecture are being used by the architectural community in general and more specifically for acoustical coordination. Topics covered will include visualization, basic modeling, consultant coordination, and construction administration. The work of GWWO Inc./Architects will be used as a basis for providing real world examples of how the program has been used through all stages of design and construction. The 70 000-sft2 Daniel Z. Gibson Performing Arts Center at Washington University will be a featured project. The center provides a home for both the Music and Drama Departments and includes state of the art teaching and performance spaces. Challenges for the project included transforming an existing 600 seat house theater into a more intimate 440 seat theater with a balcony level, designing an acoustically pleasing 200 seat recital hall while minimizing the sound focusing of a concave glass wall along one side of the hall and the addition of a 175 seat flexible experimental theater.
Error estimation due to sample size effects of in situ surface impedance measurements.
Authors: Brandao Eric , Fulco Erico , Lenzi Arcanjo , Tijs Emiel
Published: March 2010

The boundary element method (BEM) is used to predict the effect of the sample size on in situ impedance measurements on a locally reactive sample at normal incidence. The BEM model is based on a direct measurement of the acoustic impedance above the sample (with a combined sound pressure and particle velocity sensor). As the sound source is placed close to the sensor, the acoustic waves cannot be considered plane anymore. Because of that, two models to obtain the surface impedance based on the acoustic field are presented and compared. It is also shown that the sample size, the thickness of the sample, and the positions of the source and receiver affect the measured surface impedance and the calculated absorption coefficient. Those effects are parametrized in a way that the user of the in situ measurement setup may have an estimate of the required sample size for a desired frequency range with a given error allowance.
Classroom acoustics in the developing world: A student project to develop simple assessments and treatments.
Authors: McNeill Nathan , Blevins Matthew , Drott Emily , Kremer Alex , Kusch Mark , Pluhar Brian , Ringer Ashley , Sargent Valerie , Bolton J. Stuart
Published: March 2010

An inadequate educational system can be crippling to a developing country struggling to advance in the modern world. Poor classroom acoustics are a significant factor that can contribute to an inadequate education. Classroom acoustics is a mature field of research in the developed world; however, the classroom conditions and resources found in the developing world present new research challenges. A special project course was developed at Purdue University to give undergraduate students the opportunity to learn about and address classroom acoustics problems in underdeveloped regions. The goal of this special project course was to develop an evaluation and design manual that can be distributed to teachers and educational officials in those regions. The purpose of the manual is to help identify acoustic problems in classrooms, provide suggestions of simple noise control solutions, and outline best practices for the design of new classrooms. The manual specifically addresses problems related to excessive reverberation, poor signal-to-noise ratios, and noise transmission from outside the classroom. The ultimate intent of this project was to increase awareness of the unique classroom acoustic conditions in the developing world.
Some preliminary investigations on the use of a diffusion equation model for room-acoustic parameters prediction.
Authors: Escolano José , Navarro Juan M. , López José J. , Xiang Ning
Published: March 2010

Over the very recent years, an alternative model to describe sound fields in enclosures, based on the mathematical theory of diffusion, has been proposed. This model is based on the concept of sound particles traveling along straight lines and striking walls or scattering objects existing in a room. The scope of this work is to present some preliminary investigations in the prediction of objective room-acoustics parameters, such as reverberation time, clarity, and definition through the use of a diffusion equation. This paper will discuss modeling accuracy of the room-acoustics parameters using some numerical examples. Moreover, some issues concerning the numerical implementation by means of finite difference schemes for the diffusion equation are addressed.
Inverse problem of recovery of poroelastic parameters of cancellous bone by inversion of transmitted ultrasonic data.
Authors: Fellah Zine El Abiddine , Ogam Erick , Masson Catherine , Gilbert Robert
Published: March 2010

This investigation addresses the inverse problem (IP) of the retrieval of the macroscopic poroelastic parameters of cancellous bone using transmitted ultrasonic waves. The IP is solved by minimizing the objective functional (OF), which is a measure of the discrepancy, in the least squares sense, between the trial and measured data pertaining to the transmitted USW. In this study the Biot–Johnson–Koplik was employed as the interaction model (IM). A close examination of the OF computed using this IM and measured ultrasonic data transmitted through the cancellous bone showed that it exhibited several local minima and maxima. Most off-the-shelf iterative nonlinear least squares optimization algorithms (such as the Levenberg–Marquadt or Nelder–Mead simplex methods) could not retrieve a nonambiguous solution. A simple method for resolving this problem without ambiguity by using two distinct transducer central frequencies is proposed. Finally the recovered acoustic parameters are compared with those obtained using classical methods.
Quantitative evaluation of articular cartilage using a minimally invasive ultrasound method.
Authors: Viren Tuomas , Saarakkala Simo , Jurvelin Jukka S. , Tyräs Juha
Published: March 2010

Quantitative ultrasound imaging is a potential method for the evaluation of the integrity of articular cartilage. A quantitative minimally invasive intra-articular ultrasound (IAUS) technique has been developed for in situ and in vivo diagnostics of articular cartilage [Virén et al. Ultrasound Med. Biol. 35, 1546–1554, (2009)]. In IAUS, a high-frequency (40 MHz) intravascular ultrasound device is used and reflection coefficient (R), integrated reflection coefficient (IRC), apparent integrated backscatter coefficient (AIB), and ultrasound roughness index (URI) are calculated for cartilage. A set of experiments was conducted for the validation of the technique. For example, significant differences were observed in ultrasound parameters between intact, spontaneously and surgically repaired rabbit articular cartilage. The parameters revealed abnormal surface roughness and internal collagen structure by decreased R and IRC and increased values of URI and by increased AIB. Furthermore, the poor integration of the repair and surrounding native tissue was visible in the ultrasound images. As a reference, histological images revealed that repaired tissue exhibited abnormally organized collagen network, rough surface and lower collagen, and proteoglycan contents. Based on the present experimentation using IAUS, tissue structure and properties, not diagnosable by present clinical techniques, may be quantitatively evaluated with the minimally invasive ultrasound technique.
Temperature characteristics of electret silicon microphones.
Authors: Yasuno Yoshinobu
Published: March 2010

The electret condenser microphone (ECM) has become an important component for various consumer equipment systems because of its stable sensitivity and frequency characteristics and its success in achieving small size and high sensitivity. A previous report described the method to design a microphone with stable temperature characteristics. The quality of the silicon material for the diaphragm was pointed out as important for improving robustness. Recently, microphones using micro-electro-mechanical systems (MEMS) have been practically applied and widely used for mobile equipment, such as cellular phones. The major reason for adopting MEMS is that a re-flow soldering process can be used in production, in addition to having desirable features such as being small and thin. However, almost no electret types have been commercialized since a guaranteed 300°C heat resistance is required. This report presents a new trial electret silicone microphone that contains a SiO2 thin-film electret. The report also evaluates the heat resistance of a SiO2 electret, the temperature stability of a silicon diaphragm, and the temperature change in sensitivity and frequency characteristics compared with those of a conventional ECM and the microphone for measurement.
Electret condenser microphones in hearing aids.
Authors: Warren Daniel , Schafer David , Konde Kiran
Published: March 2010

As sensors small, high-gain devices, hearing aid microphones must be tiny, have high signal-to-noise ratios, be relatively insensitive to vibration from the nearby speaker, and operate at low hearing aid battery voltages. The subminiature backplate-electret condenser microphone was developed in the early 1970s specifically for hearing aids, with a very lightweight mechanical system for lower vibration sensitivity and somewhat improved signal-to-noise compared to previous designs. Although the design has been refined over several generations, the basic technology of electret films has proven to be very robust over the last 40 years, enabling continual size reductions while maintaining high signal-to-noise ratios. Noise generation and vibration sensitivity continue to be competitive dimensions for hearing aid microphones. Some of the mechanisms for noise generation and vibration pickup are intrinsic to the design of these microphones. Other mechanisms may be caused by interaction between the microphones and the hearing aid system of which they are a part. In this presentation, we will discuss both types of mechanisms and some design guidelines for minimal noise application of microphones in hearing aids.
Extending and abstracting sitar acoustics in performance.
Authors: Bahn Curtis R.
Published: March 2010

There is a very close relationship between the acoustic design of instruments of the Hindustani classical music tradition and the structure of raga. The sitar and other related string instruments have sympathetic strings which are tuned to prominent notes of a raga. These strings resonate sympathetically in performance, creating a rich and hauntingly beautiful sonority. In creating a computer extended electronic sitar, the first step was to model and extend this resonant quality of the instrument, allowing for new tunings and extended resonances not possible with a traditional instrument. The extended sitar also uses performance analysis and a variety of sensors to control various signal processing techniques that further extend the quality of the instrument. This sonic analysis and subsequent sound synthesis/computer generated compositional schemes relate directly to structures of raga. This presentation will detail the development and performance of the computer extended sitar in relationship to traditional raga structure and in new contexts not directly related to traditional notions of the sitar.
Modeling of acoustic propagation to capture finer-scale features of bounding objects.
Authors: Collier Sandra L. , Alberts W.C. Kirkpatrick , Noble John M.
Published: March 2010

One of the limiting factors of performing high-resolution modeling of outdoor sound propagation in the time domain is the treatment of the boundary conditions. Impedance ground boundary conditions have been successfully modeled in the time domain for a variety of porous ground types via the use of recursive convolution methods; however, the applications were for a flat ground. The modeling of sound propagation around buildings (or other obstacles) is usually accomplished by drastically simplifying the building geometry, thereby failing to account for fine-scale, or even some larger-scale, details. Experiments conducted at a fire training facility found that external building features, such as facets and fire escapes, can effect the sound field significantly enough so that standard acoustic diffraction models fail to accurately predict the sound field. Here we develop numerical models that allow us to capture the effects of finer-scale features. Such treatments are necessary for the development of source localization algorithms in urban or other multifarious environments.
Effects of modulated noise on detection of formant-like stimuli.
Authors: Nelson Peggy B. , Wojtczak Magdalena , Nie Yingjiu
Published: March 2010

One problem reported by listeners with hearing impairment (HI) might be that formant transitions in noise are not as salient as for listeners with normal hearing (NH). We hypothesize that a reduced ability to perceive formant transitions in noise could contribute to reduced masking release seen in HI listeners. In this study, we used pure-tone glides and harmonic complex simulations of formant transitions (shifts in intensity increments across frequency range of a formant transition over the transition duration) to test this hypothesis. Pure-tone glides were 30- or 60-ms long and were centered at 1250 or 2500 Hz, frequencies representing second formants in speech. They were presented in isolation or with unmodulated flanking tones (simulating F1 and F3) in quiet, steady noise, and gated noise. NH listeners were only slightly better than HI listeners at detecting the single gliding tone in quiet. Flanking tones adversely affected performance of the HI but not NH listeners. Performance of the NH listeners was somewhat better in gated than in steady noise, whereas the HI listeners performed similarly in steady and gated noise. The role of perception of formant transitions in masking release will be discussed. [Work supported by NIDCD R01-DC008306.]
Psychometric properties of the coordinate response measure corpus with various types of background interference.
Authors: Eddins David A. , Liu Chang
Published: March 2010

The coordinate response measure (CRM) corpus has gained broad acceptance as a research tool for investigating speech intelligibility in background competition and has been widely used in studies of informational masking. Beneficial properties of the CRM include closed set format, limited linguistic content, and inclusion of target items that promote sound stream segregation based on the voice-identify of the target talker. Extensive research has demonstrated the psychometric characteristics of CRM target-word identification in the presence of one, two, or three competing talkers from the same corpus as well as long-term speech-shaped noise (LTSSN). Percent correct identification varies non-monotonically with signal-to-background ratio for one and two CRM-talker interference and is monotonic when the number of talkers exceeds 2 or is LTSSN. The CRM has even broader applicability when used with other types of background competition. The purpose of this study is to establish the psychometric characteristics of CRM target-word identification in various backgrounds with the goal of being able to determine when it is appropriate or not to use adaptive threshold procedures with the CRM corpus. Psychometric functions and thresholds based on adaptive tracking for identification of CRM target words presented in multi-talker babble and cafeteria noise will be discussed.
Position and height asymmetries in hiatus resolution: Are they phonetically driven phenomena?
Authors: Kang Hijo
Published: March 2010

Hiatus is not preferred in many languages and is resolved in various ways. Two typological patterns have been found in hiatus resolution. When hiatus is resolved by a weakening (deletion or gliding) of one vowel, (1) V1 is more likely to be the target than V2 [Casali (1996)] and (2) high vowels are more likely to be the target than non-high vowels [Rosenthall (1997)]. If these patterns are due to human articulatory and/or auditory mechanisms [Ohala (1993)], the patterns would be expected to reflect variations in ordinary speech. In this study Korean CV1V2 nonce words produced by six Korean speakers are analyzed to test whether V1 and high vowels in a realized hiatus are actually weakened in terms of duration. The results showed that V1 is shorter than V2 irrespective of speech rate [F(1,5)=12.90, P<0.02], which was not found in CVCV or CVGV words. It was also found that high vowels are shortened in fast speech to a more degree than non-high vowels [F(1,5)=7.16, P<0.05]. The formant analysis showed that vowel-to-vowel coarticulation effects are greater in CV1V2 than in CVCV, which means that vowels in hiatus are more vulnerable to misperception and consequently, sound change.
Vowel-inherent spectral change in isolated vowels and consonant vowel consonants.
Authors: Assmann Peter F. , Nearey Terrance M. , Kiefte Michael
Published: March 2010

To study the interaction of vowel inherent spectral change (VISC) and consonant context on vowel formant patterns, we recorded 15 vowels /i ɪ eɪ ε æ ʌ ɑ ɔ oU U u ɝ ɑɪ ɑU ɔɪ/ in each of 14 consonant environments (pVp, pVb, bVb, bVp, tVt, tVd, dVt, dVd, kVk, kVg, gVk, gVg, hVd, and V) spoken by 10 men and 10 women from the north Texas region. Analysis of vowel formant frequency trajectories confirmed the reliable presence of VISC across talkers for a range of consonant environments. Vowels showing stable patterns of VISC included those acknowledged as diphthongs in most North American English dialects, /ɑɪ/, /ɑU/, /ɔɪ/, /eɪ/, /oU/, as well as /ɪ/, with more variable movement patterns for /ε/, /æ/, and /u/. Of particular interest were cases where the formants F1 and/or F2 showed a “switchback” pattern of movement, with initial movement in the expected direction for the vowel (as observed in isolated vowels) and subsequently toward the“locus” for the final consonant. Parallel analyses of VISC in three dialects (north Texas, Nova Scotia, and Alberta) are currently underway and will be reported at the meeting along with statistical modeling results [Nearey (this meeting)].
Improved representation of variance in measures of vowel merger.
Authors: Hall-Lew Lauren
Published: March 2010

Current measures of vowel merger, such as the Euclidean distance between averages, have only been able to capture some of the variability between two given vowel clusters. Reliance on averages obscures the amount of variability within a given vowel class, while other techniques, such as calculating distance between minimal pairs, rely on few tokens per speaker. Hay et al. (2006) introduced an alternative approach that accounts for the variability between two vowel clusters, taking formant values as input, rather than averages. The measure is the Pillai-Bartlett statistic [Baayen (2008)], an output of a multivariate analysis of variance (MANOVA), which represents the proportion of one variance that can be predicted by another variance. A higher Pillai value indicates a lower degree of overlap between two vowel clusters in F1/F2 space. Since the value is derived from a MANOVA, Pillais can account for known internal factors influencing the production of merger, such as phonological environment, thereby reducing the need to obtain minimal pair lists. This talk argues for using Pillais as measures of merger by comparing results from low back merger in California English [Hall-Lew (2009)] with the analysis of front vowel merger by Hay et al. (2006).
The effect of increased loudness on anticipatory coarticulation of steady-state formants in a vowel-consonant-vowel context for patients with Parkinson’s disease.
Authors: Kamphaus Jennifer A. , Sussman Joan E. , Stathopoulos Elaine T. , Huber Jessica E. , Richardson Kelly C. , Boyarsky Brit A.
Published: March 2010

The current investigation examines the effect of increased loudness on anticipatory coarticulation of the vowels [i, ɑ, u] on the preceding schwa vowel for patients with Parkinson’s disease. Four speakers, three males and one female with speech severity levels between 8% and 49%, are involved in pre-treatment, treatment, and post-treatment conditions. During treatment, participants use an auditory device which plays speech-babble into one ear while they are speaking. The speech-babble competition results in an automatic loudness increase (Lombard effect) in the talker wearing the device. Participants produced target nonsense CVC words embedded in carrier sentences at comfortable loudness levels and at an average of 3 dB above that level. The steady-state F1, F2, and F3 formant frequencies of the preceding [ə] were analyzed to determine whether increased loudness results in increased anticipatory coarticulation from the [i, ɑ, u] vowels in the CVC words, as treatment progresses. Results using steady-state formant frequency measures from this study will be compared with previous studies that used F2 trajectory measures [Tjaden and Sussman (2006); Weismer et al. (1995); Tjaden (2003)]. (Work supported by NIH 1R01DC009409-01)
The surface wave model of phonation threshold pressure and physical models of the vocal fold mucosa: Theory and experiment.
Authors: Fulcher Lewis P. , Scherer Ronald C.
Published: March 2010

One of the simpler manifestations of the role of the vertical phase difference in transferring energy from the glottal airflow to the motion of the vocal folds is the surface wave model (SWM) developed by Titze in 1988. He predicted that the phonation threshold pressure (PTP) should decrease as the glottal half width decreases and as the vocal fold thickness increases, and increase as the tissue damping increases, if no vocal tract is present. Further, his treatment of the SWM predicts that for a given glottal half width, converging glottal shapes should have a higher PTP than diverging glottal shapes. Although some of the experiments done with physical models of the vocal fold mucosa support these predictions, the measurements of the angle dependence of the PTP [Chan et al., “Further studies of phonation threshold pressure in a physical model of the vocal fold mucosa,” J. Acoust. Soc. Am. 101, 3722–3727 (1997)] do not follow the expected trend. A recent re-examination of the SWM found that the diverging-converging question is sensitive to the value of the entrance loss coefficient. Values of the coefficient near 1.37 seem to remove the discrepancy with the data. [Work supported by NIH R01DC03577.]
An acoustic and electroglottographic study of Cantonese tone.
Authors: Yu Kristine M. , Lam Hiu-Wai , Li Shing-Yin
Published: March 2010

Cantonese is a tone language with six tones (traditionally described as high level, high rising, mid level, mid-low falling, mid-low rising, and mid-low level), and also three “stopped” tones in CVC words, not studied here. The interest of this study was (1) how different acoustic cues can be used to classify the different tones, particularly in connected speech and (2) the interaction of voice quality and tone in Cantonese. Twelve speakers were recorded producing minimal pairs as in Wong [(2006)] in isolation, in isolated disyllables, and in sentence-medial disyllables, where the disyllables ranged over all possible 36 bitones in the language; all speakers also made electroglottographic (EGG) recordings. Acoustic measures were F0, F0′, cepstral peak prominence, and harmonic amplitudes H1* and H2*, H1*-H2*, H1*-A1*, H1*-A2*, H1*-A3*, and H2*-H4*. EGG measures were closed quotient and peak of increasing contact. Measures were made automatically using VOICESAUCE and EGGWORKS. Preliminary results indicate variability in the presence of creak in the realization of the lowest tone (mid-low falling) as well as variability in the presence of creak at the low f0 regions of the rises.
Improving anti-submarine warfare tracking performance by incorporating classification information.
Authors: Mortensen William H. , Krout David W. , McLaughlin Jack
Published: March 2010

Prior research has shown that more accurate classification of contact amplitudes can improve tracking performance. In this work, we augment standard, contact-based trackers with a classifier run on features of the received time series from which the contacts were extracted. Ground truth information from benchmark datasets is used in a flexible simulation framework built around the sonar simulation toolset (SST) to generate simulated target and clutter returns. The simulated returns are used as input to the classifier and the contacts from the benchmark datasets as input to the tracker. The classification information provides an additional input to the association step in the probabilistic data association (PDA) and joint PDA trackers, and to the probability of target for each contact in the Bayesian tracker. The results show that even relatively poor classification can make a noticeable improvement in tracking performance. [This work was funded by the U.S. Office of Naval Research, Contract No. N00014-01-G-0460, Delivery Order #36.]