bluuebunny/crossref_metadata_2025 huggingface dataset:
Spectral gaps and discrete transmission in slender tubes
Authors: Kushwaha Manvir S. , Akjouj A. , Djafari-Rouhani B. , Dobrzynski L. , Vasseur J.
Published: October 2013

Extensive band structure and transmission spectra for the longitudinal (acoustic) wave propagation in a system made up of N′ dangling side branches (DSBs) periodically grafted at each of the N equidistant sites on a slender tube are reported. A periodic pattern of large stop bands is obtained for the airy DSB on a slender tube. The emphasis is laid on the interesting result of huge gaps and discrete transmission spectrum due only to the DSB grafted at a single site (N=1) on the slender tube. Designing the system with open tubes allows achievement of lowest gap below a threshold frequency and extending up to zero—thereby providing an entirely discrete band structure and transmission spectrum. This should have important consequences for the suppression of low-frequency noise and for designing filters and transducers.
The effects of turbulence on an undular bore
Authors: Rendon Pablo Luis
Published: October 2013

The Korteweg-de Vries-Burgers (KVB) equation is the simplest equation to incorporate simultaneously the effects of nonlinearity, damping, and dispersion, and is of the form ηt+η,ηx+ηxxx=δ,ηxx. It has been found that the steady solution to this equation exhibits all the characteristics of the undular bore, with η(x,t) the surface profile. An examination of the effects of turbulence on undular bores, and of whether this turbulence can effectively alter the overall shape of the bore, is conducted through numerical simulation of the KVB equation. [The author thanks the Universidad Nacional Autonoma de Mexico for funding this project through a PAPIIT (IN116205) grant.]
Resonant aero-acoustic excitation of cavity depth modes
Authors: Efimtsov Boris M. , Golubev Alexey Yu. , Andersson Anders O.
Published: October 2013

Results are generalized of parametric experimental investigations of resonant aero-acoustic excitation of the depth modes of a flow-grazed cavity at low-subsonic (U50 m/s) and transonic flows (0.7M1.045). Special attention is given to the fundamental mechanisms responsible for this phenomenon. The conditions for these mechanisms to manifest themselves and interact are predicted. Observations are made of the relations between the characteristic flow and cavity parameters at which the maximum resonant aero-acoustic response of the cavity depth mode takes place. These relations take into account the distinction between the flow velocity and the characteristic spreading velocity of the disturbances downstream of the wall discontinuity as well as the distinction between the cavity depth and the characteristic linear scale in the Helmholtz number connected with the edge effect for sound waves in the vicinity of the cavity gap. Experimental data illustrating the reliability of these relations are presented. The extent of the Strouhal-number region where the cavity mode excitation can be treated as an aero-acoustic resonance is determined.
Single transducer parametric sound source
Authors: Epp Bastian , Mellert Volker
Published: October 2013

The idea of using the nonlinear effects of the propagation of sound waves with finite amplitude to build up highly directed sound sources is well known. In underwater acoustics, such parametric arrays are widely used. There are also attempts to build up parametric arrays in air to generate highly directed air-borne sound. To be able to use nonlinear effects in air as a carrier medium, one needs high amplitudes of the propagating sound wave. Consequently, a superposition of single sound sources (arrays) is often reported in the literature to increase the size, and so the directivity, of the sound source as well as the acheived sound pressure level. A major disadvantage of using arrays is the problem of slightly varying resonance frequencies and phase characteristics of the single source components as well as a complicated superposition of single directivity patterns. It is difficult to investigate basic properties of wave propagation and the nature of distortion in such a way. In this work, a single, high-power ultrasound transducer was used to radiate sound waves with finite amplitude, able to generate extraneous frequency components in air. Using only a single transducer, it was possible to investigate basic properties of nonlinear wave propagation in air.
Low-frequency pulse propagation over irregular terrain
Authors: Di Xiao , Gilbert Kenneth E.
Published: October 2013

The shape of a low-frequency pulse that propagates to the far side of a hill depends primarily on two physical mechanisms: a diffracted wave and a surface wave. Both of these mechanisms depend on frequency, so that the pulse shape and frequency spectrum of the received pulse differ significantly from those near the source. As is well known, diffraction acts like a low-pass filter. The surface wave has a more complicated frequency dependence that depends on the terrain, the frequency dependence of the ground impedance, and the near-ground sound speed profile. For a given range, these parameters determine a frequency band that contains most of the surface wave energy. The surface wave thus functions as a low-frequency bandpass filter. The combined effects of diffraction and surface wave propagation on pulse propagation over a hill are studied using a parabolic equation model and a piecewise linear mapping to flatten the irregular terrain. Numerical calculations are presented for selected terrain features, ground impedances, and sound speed profiles. [Research supported by the U.S. Army Armament, Research Development and Engineering Center (ARDEC).]
Analysis of the effect of small currents on ray stability and caustic formation in a layered moving fluid medium
Authors: Bergman David
Published: October 2013

In studies of underwater sound it is usually sufficient to treat the medium of propagation as a static inhomogeneous medium. Including current and time-dependent changes adds to the overall understanding of the behavior of sound in underwater environments. The quiescent medium approach combines the movement of the fluid with the local sound speed to create an effective sound speed profile. Recently it has been demonstrated that the effects of small-scale fluid motion on the propagation of sound in the ocean can be of the same order of magnitude as that of the static sound speed profile [J. A. Colosi, J. Acoust. Soc. Am. 119, 705–708 (2006)]. The stability of the acoustic ray equation in a moving fluid medium depends on the first and second derivatives of the sound speed profile and fluid velocity [D. R. Bergman, Waves Random Complex Media 15(4), 417–435 (2005)]. In this presentation the effects of weak but rapidly changing currents on the stability and intensity of the acoustic field are investigated for environments with low Mach number. The main result will show that in real underwater environments small currents cannot be ignored when modeling or analyzing the propagation of sound.
Influence of attack transient and decay times of percussive sounds on the echo threshold
Authors: Savitala Hari V. , Braasch Jonas
Published: October 2013

The human ability to localize a direct sound source in the presence of reflected sounds is well known as the precedence effect. The echo threshold defines the time delay of the reflection, above which the reflection becomes audible as a separate event. For musical instruments, various attack transient and decay times affect the echo threshold, but often a constant echo threshold of 50 ms is assumed for music. The aim of this study is to determine in more detail how the parameters of a test sound affect the echo threshold. For this purpose, 12 adult listeners with normal hearing participated in a psychoacoustic experiment to determine the echo threshold for various percussive and nonpercussive instrument sounds. In the headphone-based experiment, the lead was placed with an interaural time difference (ITD) of ±300 μs, while the lag was presented from the opposite side. Interstimulus interval (ISI) values of 0, 1, 2, 5, 10, 25, 50, and 100 ms were tested. The results of the experiment showed a large variation of the echo threshold. The smallest value of 7.50 ms was found for the castanet and the largest of 75.00 ms for a free-reed organ pipe.
Modulation detection thresholds for cochlear implants: Dependence on stimulation site and stimulus level
Authors: Pfingst Bryan E. , Burkholder Rose A. , Thompson Catherine S. , Xu Li
Published: October 2013

Amplitude modulation of pulse trains is the strategy of choice for transmission of temporal-envelope information in most modern auditory prostheses. Previous studies by Fu, Colletti, and Shannon have shown high negative correlations between modulation detection thresholds (MDTs) (measured at only one site in the electrode array) and speech recognition with cochlear or brainstem auditory prostheses. The objective of this study was to characterize MDTs across the parameter space used in cochlear prosthesis speech processors: all usable stimulation sites at multiple stimulus levels. Across-site patterns of MDTs varied considerably from subject to subject. In general, sites with the poorest MDTs were located in the basal third of the scala-tympani electrode array, but this pattern did not hold for all subjects or all stimulus conditions. The mean MDT for two levels (30% and 70% of dynamic range) was highly correlated with the mean for five levels. Mean MDTs at these two levels averaged over four adjacent electrodes in any of five segments of the electrode array were negatively correlated with speech recognition scores. This study suggests that in order to adequately characterize a subject’s modulation detection acuity, it is necessary to sample MDTs at multiple sites and levels. [Work supported by NIH/NIDCD.]
Influence of visual stimuli in the sound quality evaluation of loudspeaker systems
Authors: Karandreas Alex , Christensen Flemming
Published: October 2013

There is currently an increasing demand to evaluate sound quality attributes of products and to understand to what extent they influence a user’s overall impression, since there is usually more than one modality stimulating this evaluation. The present study uses loudspeakers as an example and evaluates the overall impression in relation to hearing and vision. In order to quantify the bias that the image of a loudspeaker has on the sound quality evaluation done by a naive listening panel, loudspeaker sounds of varied degradation are coupled with positively or negatively biasing visual input of real loudspeakers, and in a separate experiment by pictures of the same loudspeakers. In order to choose loudspeakers that provide a sufficient range of visual bias a preliminary visual-only experiment has been conducted. From the ongoing experiments it is possible to evaluate how much the auditory perception of a loudspeaker can be biased from visual input and study how the two modalities interact. Results from the experiments are presented.
Simulation of borehole acoustic measurements with adaptive finite elements for the accurate and efficient assessment of rock formation properties
Authors: Michler Christian , Demkowicz Leszek , Torres-Verdin Carlos
Published: October 2013

Borehole acoustic measurements are routinely used to probe in situ properties of rock formations. The numerical simulation of such measurements involves numerous outstanding challenges, such as the modeling of large spatial variations of rock porosity and anisotropy, material diffractions, source-sensor effects, and the need for a reflectionless truncation of the computational domain. To simulate borehole acoustic measurements, a new axisymmetric finite-element formulation in the frequency domain has been developed and successfully tested. The formulation couples acoustic phenomena within the fluid-filled borehole with elasticity in both the probed rock formations and the measurement tool. One of the central components of our formulation is the automatic adaptivity of element size and approximation order of the finite-element spatial discretization. No user interaction is necessary to control the adaptivity. Such a formulation enables accurate simulation results in the presence of large contrasts of material properties. A novel combination of the perfectly matched layer enhanced with finite-element adaptivity effectively truncates the computational domain. Several challenging benchmark problems are presented that confirm the accuracy, reliability, and efficiency of our method to reproduce complex waveforms excited by monopole and dipole sources in a fluid-filled borehole that penetrates layered media.
Perfectly matched layers for the numerical approximation of the radiation boundary condition in structural acoustic finite element tools
Authors: Zampolli Mario , Tesei Alessandra , Jensen Finn B. , Blottman John B.
Published: October 2013

The Berenger perfectly matched layer (PML) is one of a variety of numerical techniques for approximating the Sommerfeld radiation boundary condition. In acoustics, the PML is a nonphysical finite thickness layer of fluid or solid material that surrounds the physical computational domain of interest and acts as an anisotropic absorber of the outgoing waves. The absorption coefficients in the PML are such that the acoustic energy is dissipated selectively only in the direction perpendicular to the interface between the PML and the physical domain. The study presented here shows that the appropriate choice of PML absorbing functions can have a beneficial effect on the convergence of frequency domain acoustic finite-element tools. In the context of elastic target scattering computations, it is shown how the PML can be applied in direct contact with the wet surface of convex elastic targets. Other examples from low-frequency geoacoustic benchmarking applications demonstrate the adaptability of the PML to a variety of geometries and to problems where the radiation condition must be imposed across inhomogenous layers. The numerical results obtained with the finite-element/PML tool are compared to results from finite-element/infinite-element codes and to results from parabolic equation models.
Sound radiation from finite plates excited by a space-time stream of random impulses
Authors: Akamatsu Katsuji , Yamaguchi Takao , Kanazawa Junichi
Published: October 2013

Sound radiation from baffled finite elastic plates that are subjected to impulses occurring at random points on the surface, at random intervals, and with random strength is analyzed. The plate response is obtained in the Stieltjes integral form of the plate impulse response function with respect to the occurrence of impulses. The total sound power radiated from the plate is formulated according to Heckl’s approach. Explicit expressions for the expected value of the plate response and the radiated sound power are derived for the case in which the stream of impulses is uncorrelated. An approximate solution for the radiated sound power is obtained by assuming light damping of plates and by neglecting modal coupling effects. For comparison, the exact and approximate solutions are evaluated numerically for a plate with constant loss factors. The analysis is applied to the prediction of rainfall noise by expressing the expected value of the exciting force in terms of the size of raindrops, their terminal velocity, and the rainfall rate.
Dynamic analysis of shuttle bus generated vibration in an airport rental car center with an elevated roadway
Authors: Phillips James
Published: October 2013

A finite-element analysis (FEA) model was developed for a new rental car center at an international airport in the United States. Incorporated into the structure was an elevated roadway for shuttle buses transporting rental car customers. Of concern was possible vibration transmitted into occupied portions of the structure from shuttle buses. Measurements were conducted on an existing elevated roadway at the airport in order to determine dynamic force inputs to be applied in the FEA model developed to predict the vibration response inside the rental car center. Impulse response measurements were conducted and then the vibration response of the roadway was measured during shuttle bus passbys. The results of these tests were used to derive the dynamic forces generated by the moving buses on the structure. A finite-element analysis (FEA) model of the elevated roadway was also developed. This model was used to verify the derived dynamic force levels and to verify the modeling approach applied to the rental car center.
Parametric relations for porous foams and their use in optimization of structural acoustic performance of multilayer trim components
Authors: Lind Eleonora , Goransson Peter
Published: October 2013

Lightweight porous acoustic multilayer trim components have traditionally been specified in terms of sound absorption and sound transmission loss performance targets. Importantly, the material specification of the trim component developed only for absorption and sound transmission loss may be suboptimal in terms of, e.g., sound radiation behavior. This highlights the necessity for accurate, computationally efficient, and robust simulation method, which should form an integral part of a multidisciplinary optimization tool. In addition, for such optimization to be physically meaningful the design parameters used should be based on relations between microstructural dimensions and properties, and the corresponding macroscopic parameters describing porous materials. This paper discusses recent results from research focused on the parametrization of porous foams, i.e., continuous links between micro-dimensions, elasticity, density, and flow resistivity, used in higher order 3-D finite element simulations of multilayer components for structural-acoustic applications. In addition to the microdimensions, also the thicknesses of individual layers have been used as design parameters, with the overall weight as the design objective. Acoustic and vibration targets, velocity of the radiating surface, radiation efficiency, transmission, etc., as well as upper and lower bounds of the microdimensions, have been used as constraints.
Adult age differences in the use of envelope cues to identify noise-vocoded words with a varying number of frequency bands
Authors: Sheldon Signy , Pichora-Fuller Kathy , Schneider Bruce
Published: October 2013

Older adults with good audiograms have difficulty understanding speech in noise. Age-related differences have been found on some temporal processing measures such as gap detection; however, older adults are believed to have well-preserved ability to use envelope cues to identify words. Following Shannon et al. (1995), we used noise-vocoded speech such that the amplitude envelope of speech was retained in frequency bands but filled with noise, thereby obliterating fine structure cues within each band. In experiment 1, younger and older listeners heard a list of words. Each word was presented first with one vocoded frequency band, and the number of bands was incremented until the listener correctly identified the word. The average number of bands required for correct identification was found to be identical for both age groups. In experiment 2, both age groups identified words in four blocked noise-vocoded conditions (16, 8, 4, and 2 bands). Younger adults outperformed older adults. Although older adults were as able as younger adults to use envelope cues cumulatively in experiment 1, they were less able to use these cues without the benefit of repetition.
Energy suppression of steady-state portions of vowels while maintaining the energy of consonants better improves speech intelligibility for elderly listeners in reverberation
Authors: Miyauchi Yusuke , Arai Takayuki
Published: October 2013

In a reverberant environment, overlap-masking renders speech perception difficult. Arai itet al. proposed that energy suppression of steady-state portions of speech improves speech intelligibility with young subjects [Acoust. Sci. Technol. 23, 229–232 (2002)]. Audibility degrades with age. Therefore, reverberation can be a more critical barrier to speech perception by elderly listeners. To investigate the effect of suppressing reverberation for elderly people, a listening test was conducted with 25 elderly subjects (mean age: 73.3 years) using the following three types of speech: (a) speech without energy suppression of steady-state portion, (b) speech with the suppression of steady-state portions of speech, and (c) speech with the suppression only of vowels, while maintaining the energy of consonants. As expected, speech intelligibility by method (b) (46.7%) was markedly improved from (a) (42.2%), but method (c) (52.0%) improved it more. Because the energy of consonants is less than that of vowels, overlap-masking attendant on previous vowels would largely affect perception of subsequent consonants. These results suggest that suppressing the steady-state portions of vowels while maintaining the energy of consonants serves elderly persons well to improve speech intelligibility in a reverberant environment. [Work supported by JSPS.KAKENHI (16203041).]
Description of mouth shapes used in traditional Japanese sign language by comparison to those of speech utterance
Authors: Fukuda Yumiko , Hiki Shizuo
Published: October 2013

Mouth shapes used in traditional Japanese sign language were described systematically through comparison to those in uttered speech. A framework of mouth shape descriptions that had been proposed by the present authors included (1) projection of mouth shapes of Japanese vowels and semi-vowels to the lip articulation plane of the three-dimensional IPA vowel chart and (2) grouping of discriminating consonants in lip-reading on the IPA articulation matrix. It was transferred to a coordinate consisting of contractions of muscles involved in mouth movement. In traditional Japanese sign language, the mouth shape, when added as a part of facial expression, modifies meaning of the word conveyed by a hand gesture. Those mouth shapes were extracted from conversations among native signers and were symbolized by their respective degrees of closing, rounding, and pulling corners, and by categories borrowed from Japanese vowels and some consonants. Using the extended framework, those mouth shapes that were elected for visual discrimination ranged more widely than those elected to discriminate phonemes aurally. Video pictures were retrieved according to their symbols from an electronic dictionary of 250 basic sign words and 1500 exemplary sentences of their meanings, as edited by the Japanese sign language research group, NRCD.
Acoustic correlates of primary motor speech disorders in children during oral and hand tasks
Authors: Peter Beate , Stoel-Gammon Carol
Published: October 2013

Primary child speech disorders have been subclassified variously. One proposed subtype, childhood apraxia of speech (CAS), results from motor programming deficits in speech production, and associated characteristics include limited phoneme inventory, variability in production, and difficulty with speech prosody, e.g., monostress and monopitch. The purpose of this study was to identify acoustic correlates of oral and limb movements, derived from the hypothesis of a generalized timing deficit, that characterize this disorder subtype differentially from other disorder subtypes. Twelve children (age 4;7 to 6;6), of which eight were labeled with CAS, and 12 age- and gender-matched peers participated in a variety of oral motor and hand tasks. K-means clustering generated three disorder subgroups in the sample of participants with speech disorders, two of which were populated with CAS labels. Distinguishing characteristics included accuracy of vowel duration in a nonword imitation task, accuracy of clap interval duration in a clapped rhythm imitation task, rate of syllable repetition, durations of unstressed but not stressed vowel durations, and breath group duration. Taken together, these results are consistent with the view that deficits in timing accuracy, movement speed, and breath support underlie at least one subtype of idiopathic speech disorder.
Changes of voice characteristics under noisy conditions: Acoustic and fiberscopic studies of school teachers
Authors: Kobayashi Noriko , Masaki Takashi
Published: October 2013

It has been known that incidence of vocal pathology was high in school teachers as they had to use a great amount of voice and loud phonation in noisy school settings with the children’s loud voices. Vocal hygiene programs and voice therapy are effective treatments for these cases. The final goal of the treatment should be to obtain the skills to use efficient and healthy phonation even in abusive and noisy environments. In our study, two school teachers with vocal nodules, two speech therapists who were trained to keep proper phonation even in noisy environments, and ten college students with no laryngeal pathology (control group) spoke eight sentences in two environments: a quiet environment and one with meaningful multitalker babble (MMB). Acoustic analyses and fiberscopic examination revealed higher sound pressure and F0 levels and laryngeal constriction in MMB for the teachers before therapy and the students, whereas two speech therapists kept similar F0 levels and the nonconstricted laryngeal condition in both environments. A teacher who had voice therapy produced less constricted voice with MMB than the pretherapy recording. The results suggested the efficacy of voice therapy with MMB for teachers to obtain efficient and healthy voice in their noisy work environments.
Effect of speech task on intelligibility in speakers with dysarthria
Authors: Bunton Kate
Published: October 2013

There is some evidence that speech intelligibility varies across speech task for speakers with dysarthria. Primary differences have been related to the linguistic level of the material being evaluated (e.g., single words, sentences, conversation). While conversational speech is the most socially valid context for evaluating speech intelligibility, quantifying intelligibility in conversation limits its clinical use. The present study assessed intelligibility in speakers with dysarthria across four speech production tasks: single words, sentence production, passage reading, and spontaneous speech. Speakers varied with regard to the type of dysarthria, including hypokinetic, hyperkinetic, spastic, and mixed. Results show that all speakers were significantly less intelligible during spontaneous speech than the other structured speech production tasks (e.g., single words, sentences, and reading). Mean differences varied across speech severity level but did not vary across dysarthria type. These data indicate that speech production task is an important variable to consider during the evaluation of dysarthria and highlight the importance of developing techniques to measure the intelligibility of conversational speech. [Work supported by NIH Grant R03 DC005902.]
A speech communication aid system for total laryngectomies using voice conversion of body transmitted artificial speech
Authors: Nakamura Keigo , Toda Tomoki , Saruwatari Hiroshi , Shikano Kiyohiro
Published: October 2013

There are several problems associated with using existing electrolarynxes. For example, the loud volume of the device itself might disturb smooth interpersonal communication, and its generated speech is also unnatural. To improve the quality of speech communication using such a medical device, this paper proposes a novel speech communication aid system for total laryngectomies. This system detects articulated speech caused by a new sound source as an alternative to the existing electrolarynx through the soft tissues of the head with a nonaudible murmur (NAM) microphone attached to the surface of the skin [Nakajima et al, Proc. Interspeech 2005, pp. 293–296 (2005)]. The new sound source outputs signals of extremely low energy that cannot be heard by people near the speaker. Such body-transmitted artificial speech is converted to a more natural voice by statistical voice conversion [T. Toda and K. Shikano, Proc. Interspeech 2005, pp. 1957–1960 (2005)]. The performance of the proposed system is evaluated in terms of objective and subjective measures using body-transmitted artificial speech simulated by a non-disabled speaker. Experimental results show that body-transmitted artificial speech is consistently converted to a much more natural and intelligible voice. [Work supported by SCOPE-S.]
Development of a hands-free electrolarynx for persons whose larynx have been removed
Authors: Hashiba Mitsuo , Sugai Yasunori , Izumi Takashi , Ino Shuichi , Ifukube Tohru
Published: October 2013

An electrolarynx is a useful speech-substitute device for patients who have lost laryngeal function. Unfortunately, the voices produced by a conventional electrolarynx are completely flat, unlike a human voice. For that reason, we developed a pitch-controlled electrolarynx that allows patients to control a voice intonation using their exhalation [N. Uemi and T. Ifukube, ‘‘Design of a new electrolarynx having a pitch control function,’’ IEEE RO-MAN’94, 198–203 (1994)]. This pitch-controlled electrolarynx has been manufactured and more than 3000 patients have used the device since 1998. However, many patients have strongly requested that we design a hands-free electrolarynx so that they can perform daily life functions and work in their office without their hands occupied by the electrolarynx control during use. In response to that request, we designed some models of a hands-free electrolarynx that is attachable to the patient’s neck. Furthermore, by comparing the models based on evaluation tests regarding its sound quality and usability, a thermo-plastic brace was found to be the best for attachment to the neck. Although our hands-free electrolarynx has not yet been equipped with the pitch-control function, it was ascertained that it is useful for taking notes while talking on the phone, and so on.
Collagen injection as an adjunct to arytenoid adduction
Authors: Kimura Miwako , Nito Takaharu , Tayama Niro , Chan Roger
Published: October 2013

Dysphonia associated with laryngeal paralysis may be identified in the short term postoperatively or may develop years after successful arytenoid adduction. In selected cases, collagen injection of vocal fold improves phonation after arytenoid adduction. Our experience with the use of collagen injection to supplement arytenoid adduction is reported. Forty patients were treated with transoral collagen injection into the atrophic vocal fold after arytenoid adduction at The University of Tokyo from January 1990 to December 2005. These patients all had a diagnosis of unilateral vocal fold paralysis. Voice quality (GRBAS scale) and the other examinations for vocal function demonstrated measurable improvement after treatment. After arytenoid adduction, maximum phonation time (MPT) improved from 5 to 11 s and mean flow rate (MFR) as an aerodynamic examination improved from 772 to 322 ml/s. Voice quality, especially roughness, improved from 1.8 to 0.8 after collagen injection. MPT (from 11 to 14 s) and MFR (from 322 to 233 ml/s) showed overall improvement after collagen injection. We will report the ease with which transoral collagen injection was accomplished and how better voice quality was attained without the more extensive surgery. In the session techniques of anesthesia, injection, and patient selection are discussed.
Evaluation of a photonasography
Authors: Amelot Angélique , Honda Kiyoshi , Maeda Shinji , Crevier-Buchman Lise , Basset Patricia
Published: October 2013

A photonasographic (PNG) technique was tested to evaluate velopharyngeal port (VP) opening/closing during speech. This system uses a photodetector placed in the pharynx via a nostril and a fiberscope in another nostril pointing to the back wall of the nasopharynx. With this setting, the light is reflected from the walls, passes through the open port, and is sensed by the photodetector. In a pilot experiment, a native French speaker read sequences C1V1C1 V2C1V1, where C1=/n/ or /t/ and V1,2=/a/, /i/, /A/, /E/, or /O/, as /tatA ta/. The obtained PNG data compare qualitatively well with the velum position data, on the same corpus, derived from the naso-fiberscopic video images. When NPG detects a VP opening, the fiberscope data correspondingly indicate a lowering of the velum and inversely. PNG, as well as the velum position data, show that VP is open at the onset of a nasal phoneme and is closed just after the offset of that nasal phoneme. Moreover, VP is not completely closed during the first and second phoneme in the sequence [ninini] and [nanana], indicating their nasalization. The combination of a well-set fiberscopic light-source and photodetector appears to give a good estimation of the degrees of VP opening.
Lip protrusion and tongue position in French vowels produced by blind speakers and sighted speakers
Authors: Menard Lucie , Leclerc Annie , Aubin Jerome , Brasseur Annie
Published: October 2013

The influence of visual experience in speech production was investigated through a study of the French vowels /i y u a/. Articulatory and acoustic recordings of ten repetitions of the vowels in CVC syllables embedded in carrier sentences were conducted in three consonant environments (/b d g/) and two prosodic conditions (neutral and under contrastive emphasis). Six congenitally blind adults and six sighted adults were recorded. All participants were native speakers of Canadian French and had no history of speech disorder. The audio signal, lip movement, and tongue shapes were recorded using a digital camera and an ultrasound system. The minimal (for /i a/) or maximal (for /y u/) horizontal position of the upper lip was tracked and the front-back position of the tongue was measured for each vowel. Formant frequencies were extracted using LPC analyses. Results show that, despite similar acoustic differences, the rounded and unrounded vowels are less differentiated along the protrusion dimension for blind speakers compared to sighted speakers. It is suggested that this variability is related to a trade-off between lip protrusion and tongue position. Significant interaction effects between consonant environment and vowel category are discussed for each speaker group. [Work supported by SSHRC, NSERC.]
The effects of the false vocal folds on laryngeal pressures and flows
Authors: Sheng Li , Scherer Ronald C. , MingXi Wan , SuPin Wang , LiYun Qi
Published: October 2013

As a laryngeal constriction, the false vocal fold (FVF) gap (the distance between the medial edges of the FVFs) may create important effects on phonation by altering the pressures within and flows through the larynx. The computational code FLUENT was used to examine the effects on pressures and flows for FVF gaps ranging from 0.02 to 2.06 cm, for three glottal angles (uniform and convergent/divergent 40 deg) and two minimal glottal diameters (0.04 and 0.06 cm), for constant subglottal pressure (8 cm H2O). The specific design of the FVFs followed prior literature. Results suggest three important ranges of FVF gaps: (1) when the FVF gap was 1.5–2 times the minimal glottal diameter, pressures were lower throughout the larynx, and flows were higher through the larynx (less flow resistance); (2) for smaller FVF gaps, intralaryngeal pressures increased and flows decreased; (3) for greater FVF gaps, pressures and flows were unaffected. Also, (4) the divergent glottal angle gave the greatest flows, (5) flow separation locations for the divergent glottis moved downstream as the FVF gap decreased, and (6) the FVFs straightened the flow for narrow gaps. Thus, the status of the FVFs needs to be considered for voice production and synthesis.
Coherent noise processing and geoacoustic inversion
Authors: Gerstoft Peter , Huang Chen-Fen , Hodgkiss William S.
Published: October 2013

Ocean acoustic noise can efficiently be processed to extract Green function information from noise [Roux et al., J. Acoust. Soc. Am. (2004), Siderius et al., ibid. (2006)]. By cross-correlating the ambient noise field from two sensors, it is possible to extract the impulse response between the two sensors including bottom and subbottom bounces. When this noise processing is used on a vertical array, it can give valuable information about the subbottom near the array. This information will be then be used to constrain a classical geoacoustic inversion procedure where we use a distant towed source to obtain the geoacoustic bottom parameters.
Geoacoustic inversion based on both acoustic pressure and particle velocity
Authors: van Leijen A. Vincent , Hermand Jean-Pierre , Smith Kevin B.
Published: October 2013

Conventional inversion schemes for environmental assessment depend on an objective function that exploits amplitude or phase information of acoustic pressure data alone. This work investigates the potential of vector sensors for geoacoustic inversion by defining an objective function that also takes into account acoustic particle motion. Calculations are performed on synthetic broadband data for a shallow water environment (South Elba) with an optimization scheme based on different metaheuristics. Differences in the inversion process, including sensitivity of the cost function to environmental parameters and convergence speed of the optimization algorithm, are presented by comparing inversion results for a sparse pressure-only array and a vector sensor array.
Mismatch effects on passive matched-field, thick-sediment inversion
Authors: DelBalzo Donald R. , Linzell Robert S. , Pflug Lisa A.
Published: October 2013

Acoustic field predictions in shallow-water areas can be severely limited by environmental uncertainty, especially in sediment property characteristics. Direct methods of obtaining geophysical data are expensive, so inverse methods, which invert signals from controlled active sources, have recently become popular. This work assesses the impact of environmental mismatch on low-frequency (<100 Hz) matched-field correlations on broadband signals from surface ships with unknown source levels at unknown ranges. A range-staggered technique is employed to create a range-dependent sediment description, with ever-increasing confidence as additional ships are analyzed. Matched-phase techniques are applied in a simulated shallow-water environment with a single vertical array and high signal-to-noise ratios. The simulations indicate significant potential for passive, range-dependent, thick-sediment characterizations out to ranges of tens of water depths in shallow water, despite reasonable mismatch conditions in the water and in the sediments. [Work sponsored by SPAWAR.]
To obtain geo-acoustic parameters by the acoustic inversion technique using subbottom profiler
Authors: Wang Chung-Wu , Lee Cheng-Che
Published: October 2013

The fast sound speed at sea surface in the water surrounding Taiwan causes strong refraction. If the water depth is shallower than 5000 m, the sound ray will reflect at the bottom, causing severe bottom losses. Therefore, geo-acoustic parameters are important for acoustic studies in the water surrounding Taiwan. In the past, researchers used single-pulse sonar of 3.5 kHz to obtain geo-acoustic parameters by the spectral ratio technique. Now the sediment exploration has already advanced and used the Chirp Sonar system on Taiwanese Research Vessels such as Ocean Researcher I (OR I) and Ta-Kuans TOPAS system. This paper presents the results of attenuation coefficients in ocean bottoms using the acoustic inversion technique on the chirp sonar data collected by OR I. The results are compared with the core data, and the error is within 15%.
The resolution of a nonlinear ocean acoustic inverse problem
Authors: Odom Robert I. , Ganse Andrew A.
Published: October 2013

For a strictly linear inverse problem the model resolution matrix is a simple product of the linear system matrix and its generalized inverse. How much the model resolution differs from the identity is an indication of bias in the inverted model. Nonlinear inverse problems lead to higher order resolution operators, which allow one to quantify the effect of ignoring the nonlinearity in the inverse problem. In addition, the trade-off between resolution and variance is examined for the nonlinear inverse problem. Ocean acoustic inverse problems for ocean and/or bottom structure are intrinsically underdetermined, because some continuous quantity such as sound speed or temperature is estimated from a finite number of data values, and the model is generally discretely parametrized to be numerically tractable. Discretizing the model has the positive effect of regularizing an intrinsically ill-posed problem. However, discretizing an intrinsically nonlinear problem extracts a penalty. Higher order resolution operators may contribute even when they should be identically zero. (For example, in a quadratically nonlinear model, cubic and higher order resolution operators should be zero, but won’t be zero as a result of the discretization.) Modal propagation in an ocean acoustic waveguide is used to illustrate the issues. [Work supported by ONR.]
Calibration of consonant perception in room reverberation
Authors: Ueno Kanako , Kopco Norbert , Shinn-Cunningham Barbara
Published: October 2013

Many studies of sound perception often assumed that our auditory sensory processes are relatively static, rather than plastic. However, in everyday environments, we naturally and fluidly compensate for interfering effects of background noise and room reverberation. In order to investigate how listeners calibrate auditory perception to such acoustic interference, a listening experiment was performed to measure the effect of sudden changes of reverberation on the identification of consonants. Test sounds were generated by convolving two types of binaural room impulse responses (BRIRs) measured in large real rooms with speech tokens. As a control condition, pseudo-anechoic BRIRs with negligible reverberation energy were used. Listeners were asked to identify the consonant in a vowel-consonant target. The target was preceded by a carrier phrase consisting of vowel-consonant pairs from the same talker. In some cases, the target and carrier phrase were processed by the same BRIRs, while in others the BRIR’s processing target and carrier differed. Consistent effect of calibration was observed in one of the simulated rooms, but not in the other, suggesting that the ability to compensate for the effects of reverberation depends on the specific pattern of reverberation produced in a given room. [Work supported by AFOSR and NSF.]
Sentence context influences vowel perception in reverberant conditions
Authors: Wotton Janine , Welsh Kristin , Smith Crystal , Elvebak Rachel , Haseltine Samantha , Shinn-Cunningham Barbara
Published: October 2013

Sentences recorded with a Mid-western accent were convolved with head-related impulse responses that included different room reverberation conditions. The stimuli were presented binaurally through headphones in an echo-attenuated chamber and subjects (n=23) typed the sentences they heard. The target word was one of a vowel pair (cattle/kettle, jam/gem, gas/guess, past/pest) embedded as the second word in one of three sentence types. The neutral sentence provided little context for the word. Target words in sentences that provided strong contextual cues could be congruent or incongruent with the expectations of the subject, for example, ‘‘The cattle/kettle grazed in the meadow.’’ Subjects made significantly more errors in the incongruent sentences compared to the neutral (Wilcoxon=3.572 p<0.05) or congruent sentences (Wilcoxon=3.56 p<0.05). When the target word was in a congruent sentence subjects performed equally well in reverberant or pseudo-anechoic conditions (Wilcoxon=1.298) but they made more errors in the reverberant condition for both neutral (Wilcoxon=3.359, p<0.05) and incongruent sentences (Wilcoxon=2.241, p<0.05). Results suggest that reverberation may cause listeners to rely more heavily on linguistic context to determine word meaning. [Work supported by NOHR, AFOSR.]
Quantifying the effects of room acoustics on speech intelligibility in very low signal-to-noise ratio environments
Authors: Whittington Jarrod E. , Bradley John S.
Published: October 2013

The intelligibility of speech transmitted from closed offices to adjacent spaces is strongly affected by the signal-to-noise ratio at the receiver position and the acoustical characteristics of the spaces involved. Previous studies have suggested that the effect of room acoustics on speech intelligibility in closed offices and rooms is negligible and can be ignored (as with intelligibility quantifiers such as the articulation index). The purpose of this study is to show that in conditions of very low signal-to-noise (i.e., when high speech privacy is a primary concern), the influence of room acoustics rises dramatically. To this end, multiple subjects were given tests of speech intelligibility in simulated sound fields. Speech samples were presented to subjects with seven levels of signal-to-noise and four different reverberation times. The results from these tests show that as reverberation time increases, speech intelligibility decreases much more sharply for very low (−8 dB) signal-to-noise situations than in higher (+10 dB) signal-to-noise situations. This suggests an important relationship between room acoustics and speech privacy/security.
The effect of a preprocessing approach improving speech intelligibility in reverberation considering a public-address system and room acoustics
Authors: Hodoshima Nao , Arai Takayuki , Svensson Peter
Published: October 2013

This study evaluates a preprocessing approach for reducing reverberation effects when the input signal is not ideal, dry speech, but rather a realistic speech signal picked up by a close microphone in a room. And this study shows how the situation affects the input and the further approach compared to a dry signal as the input. Steady-state suppression, as described by Arai et al. [Acoust. Sci. Technol. 23, 229–232 (2002)], was used as a preprocessing approach that processes a speech signal before it is radiated from loudspeakers. A lecture was simulated in two different halls (reverberation times of 1.2 and 1.8 s) in which public address systems were installed. The simulation software CATT-Acoustic was used and impulse responses were calculated for the input to the preprocessing approach and for a listener position. Stimuli for a syllable identification test were prepared by convolving speech signals with the calculated impulse responses. Speech signals were given with and without steady-state suppression. The inclusion of natural and electroacoustical impulse responses makes the study of steady-state suppression more realistic and tests its robustness. [Work supported by JSPS (176911).]
Improving speech intelligibility in reverberant rooms
Authors: Winker Douglas F. , Hixson Elmer L.
Published: October 2013

Speech intelligibility in rooms with long reverberation times has long been a problem for acousticians. This problem is exacerbated when the option for acoustic treatments is not an option for a variety of reasons. In this study, a constant-beamwidth, wide-bandwidth (CBWB) loudspeaker array was used to improve speech intelligibility in two multipurpose rooms where treatment was not an option. A CBWB loudspeaker array was designed with independent beam pattern control on both the horizontal and vertical axes. A wide horizontal beam pattern and narrow vertical pattern were designed to achieve constant coverage from 500 Hz to 4 kHz. The array incorporates four separate arrays, two with nested elements, and linear-phase FIR filters to maintain constant beamwidth over the three octaves of interest. Room simulations were conducted with CATT-Acoustic and the array was implemented and compared to a source with a more conventional coverage pattern. The STI method was used to measure speech intelligibility at several positions throughout the rooms. The study showed an improvement in speech intelligibility and a reduction in measured reverberation times.