bluuebunny/crossref_metadata_2025 huggingface dataset:
Three-dimensional phononic crystals made by brazing aluminum beads to form an opal structure
Authors: Page John H. , Hu Hefei , Tanaka Yukihiro , Okada Takuro , Tamura Shin
Published: May 2008

Most phononic crystals studied so far are constructed from periodically arranged objects (e.g., spheres or rods) immersed in a continuous solid or fluid medium. In this presentation, we investigate the properties of a different type of phononic crystal made by weakly brazing 4-mm-diameter aluminum beads in a face-centred cubic array, producing a large-scale opal. In this crystal, wave propagation proceeds through the network of coupled beads rather than through the surrounding medium (air in our case), and the main band gaps that we observe result from coupled resonances of the beads rather than from Bragg scattering. Using ultrasonic techniques, we measure the transmission coefficient and the dispersion relation along the ΓL direction, and show that the lowest-frequency gap occurs between about 500 and 600 kHz - a frequency range that is just below the lowest resonance of an isolated aluminum sphere. We have also measured the group velocity, which we find to be negative in the lowest frequency band gap. Further insight into the properties of these phononic crystals is obtained by performing finite difference time domain (FDTD) calculations, in which the coupling between spheres is taken into account by allowing the spheres to overlap.
Global Trend of Precision Cleaning Technology and the Application in Hard Disk Drive Industry
Authors: Zhang Jun , Liew Yun Fook
Published: May 2008

As industry trends for hard disk drive and related electronic devices are moving toward high capacity and miniaturization. A big challenge is to find increasing the efficient ways to clean the parts which are regarded as the most stringent in cleanliness. The critical issues include the methodology for identifying the contaminants and how to choose a correct cleaning chemical and cleaning technology. Today's hard disk media, particularly the perpendicular recording media, is capable of storing up to 300GB per disk under which the distance between the recording head/disk have to be reduced to a few nanometers. It will bring more difficulties for increasing the reliability and disk capacities. On the other hand, all components within the hard disk drive are required to avoid submicron particulate contamination (<0.1μm). Megasonic and ultrasonic cleaning are the most effective and promising technologies to remove such contaminants. This presentation will discuss the technical trend and possible solution of precision cleaning and the strategy for selecting the cleaning chemicals and technologies to meet the cleanliness requirement through a case study.
Accoustic cleaning in nanoelectronics
Authors: Mertens Paul
Published: May 2008

The fabrication of state-of-the-art electronic integrated circuits involves multiple cleaning steps in which residues of the order of ten or a few tens of nm need to be removed with very high efficiency. Historically cleaning was obtained by chemical undercutting (order of a few nm). The dimensions of nanoelectronic devices become so small that the amount of substrate etching involved in a cleaning step should be kept below 0.05 nm. Therefore other cleaning mechanism, involving a mechanical force, have to be considered such as acoustic agitation of the cleaning liquid or high-velocity aerosol bombardment. These forces, however, can damage fine structures already fabricated on the substrate, particularly the “up-features.” This implies that a trade-off needs to be made between avoiding damage and obtaining an acceptable cleaning performance. This presentation includes typical hardware configurations and possible mechanisms for particle removal and use of megasonic agitation in nonaqueous liquid solutions.
Potential uses of ultrasound in the dairy ultrafiltration processes
Authors: Muthukumaran Shobha , Kentish Sandra , Ashokkumar Muthupandian , Stevens Geoff
Published: May 2008

There has been a growing interest in the industrial application of ultrasound, especially in the food industry. Power ultrasound can have a number of physical effects; it can increase turbulence through both the introduction of vibrational energy and through acoustic streaming, it can cause both particle dispersion and particle agglomeration and clean surfaces with a scouring action. Our work in this area has focused on the use of ultrasound to enhance membrane processing. Low frequency ultrasound has been used to facilitate cross flow ultrafiltration of dairy whey solutions for both during the ultrafiltration production cycle and the cleaning cycle. During the production cycle, the use of ultrasound reduces both pore blockage and the specific resistance of the fouling cake layer. This leads to higher flux rates and the potential for longer production cycles. During the cleaning cycle, ultrasound systematically increases cleaning efficiency, thus has the potential to reduce both total chemical consumption and system downtime. There was no deterioration in cleaning effectiveness or membrane condition, which implies that sonication, has not damaged the membrane itself. Similarly, there was no change in the chemical nature of soluble proteins following sonication.
Measurements and simulations on the wall shear stress from single cavitation bubbles
Authors: Dijkink Rory , Klaseboer Evert , Khoo Boo Cheong , Ohl Claus-Dieter
Published: May 2008

Cavitation bubbles collapsing near a solid boundary manifest a jetting effect towards the closest rigid boundary. After impacting on the boundary the jet spreads out radially thereby shearing the surface. It is speculated that cavitation bubbles clean through the resulting shearing forces, e.g. they drag adherent contaminants through the radial spreading jet. Although the cleaning through bubbles is used in a wide set of technologies, for example for pre-cleaning surgical equipment, for bio-film removal in medical and drinking water applications, and in ultraclean processing of semiconductor wafers, very limited experimental studies on the wall shear stress exist. In an effort to shed light on the shear forces caused by no spherical oscillating cavitation bubbles close to boundaries we implement a constant temperature anemometer together with high-speed imaging to elucidate the fluid dynamics near the boundary. Additionally, we perform potential flow simulations with a boundary element method. The boundary layer is resolved with a convolution integral to the solution of Stokes' first problem. In general we find the generation of wall shear stress both in experiment and simulations during the growth and collapse of the bubble. Yet, very high shear stress is generated once the jet impacts and spreads on the boundary.
Propagation in a dispersive medium: a phase-space approach
Authors: Cohen Leon
Published: May 2008

We derive exact moments for pulse propagation in a dispersive medium. These moments are not only inherently interesting but clarify the validity of a recently proposed approximation scheme for wave propagation. The approximation method for pulse propagation is based on the Wigner position-wave/number representation and is very accurate, easy to apply, and moreover is physically illuminating. In particular one obtains the evolved approximate Wigner distribution from the initial Wigner distribution by a simple linear translation in phase space. Propagation with damping is also taken into account. We will show that the reason for the high accuracy of the approximation is that the important low order moments are exactly given by the approximation and that these low order moments preserve very well the basic shape of the pulse. Moreover, now that we understand why the approximation method works well, the approximation can be systematically improved. We give a number of specific examples of exactly calculable moments to illustrate the method and we compare exact and approximate moments. Research supported by the AFOSR.
On the binding of successive tones: Implicit versus explicit pitch comparisons
Authors: Demany Laurent , Pressnitzer Daniel , Semal Catherine
Published: May 2008

Listeners were presented with sound sequences in which one pure tone (T) was followed by a set (S) of five synchronous or asynchronous pure tones 550 cents apart. In a “present/absent” condition, T was either identical to a randomly selected component of S or halfway in frequency between two components, and listeners had to indicate if T was present in S or not. In an “up/down” condition, T was 100 cents below or above a randomly selected component of S, and listeners had to identify the direction of the frequency shift. When the components of S were asynchronous, the present/absent task was easier than the up/down task. When the components of S were synchronous, the opposite trend was observed. In case of asynchrony, the components of S could be heard out individually, so listeners presumably compared explicit (conscious) pitch percepts to make their judgments. In case of synchrony, the components of S were difficult to hear out individually; apparently, pitch comparisons were then made implicitly by “frequency-shift detectors” (Demany and Ramos, 2005) which, we argue, participate in auditory scene analysis. It seems that such detectors relate automatically consecutive tones, and are generally less efficient for comparisons between nonconsecutive tones.
Concurrent sound segregation and peripheral frequency resolution
Authors: Micheyl Christophe , Keebler Michael V. , Oxenham Andrew J.
Published: May 2008

In everyday environments where multiple sound sources coexist, listeners often have to follow a harmonic source of interest (e.g., someone's voice) among other such sources (e.g., other talkers). Here, we review psychoacoustical findings suggesting that this form of auditory scene analysis is related to the degree of peripheral “resolvability” of harmonics in the auditory system. We present experimental results, which indicate a systematic relationship between harmonic resolvability and the ability of normal-hearing listeners to “track” (or discriminate) changes in the fundamental frequency (F0) of a target harmonic complex mixed with another (interferer). We show that poor harmonic resolvability is associated not only with poor performance, but also with an inability to take advantage of normally potent cues for concurrent sound segregation, including F0 differences, laterality differences, and onset-offset asynchronies. While it remains unclear whether this relationship is causal, these findings may have important implications for the design of artificial auditory-scene-analysis systems, and for our understanding of the listening difficulties experienced by hearing-impaired listeners in whom peripheral frequency resolution is reduced. [Work supported by NIDCD R01DC05216.]
Signal duration modulates age differences in neuromagnetic brain activity associated with concurrent sound segregation
Authors: Alain Claude , McDonald Kelly
Published: May 2008

Listeners' ability to parse concurrent sounds is a prerequisite in solving the cocktail party problem. Here, we examine whether deficits in periodicity coding can account for older adults' difficulties in understanding speech in noise. In two experiments, we measured auditory evoked fields (AEFs) while young and older adults listened to complex sounds that either had all of their harmonics in tune or had the third harmonic mistuned by 4 or 16% of its original value. For each participant, the AEFs were modeled with a pair of dipoles in auditory cortex and the effects of age on the resulting source waveforms were examined. Older adults were less likely to report hearing the mistuned harmonic as a separate sound than young adults, but only for short duration signals (50 ms). This age difference was paralleled by reduced neuromagnetic activity indexing the processing of the mistuned harmonic. For longer duration signals (e.g., 200 ms), older adults show comparable amplitude, but a delay in latency. These findings show an age-related decline in concurrent sound perception based on harmonicity, which could partly be alleviated by increasing signal duration. The implications of these findings for speech perception in noise will be discussed.
Auditory scene analysis emerges from a distributed yet integrated network
Authors: Cusack Rhodri , Pressnitzer Daniel
Published: May 2008

Three architectures that have been explicitly or implicitly proposed for auditory scene analysis were evaluated: (1) that some single module is responsible for scene analysis after feature extraction; (2) that scene structure is elaborated through feed-forward stages of auditory processing; and (3) that scene analysis is integrated yet distributed. It is argued that the available evidence points towards the third model. Against (1), scene analysis is influenced by features that are best represented across a broad range nodes in the auditory network, and transmitting this information to a single module would not be parsimonious. Furthermore, neurophysiology has shown a range of nodes reflect scene analysis, ranging from the cochlear nucleus, through auditory cortex to the parietal cortex. Against both (1) and (2) is the high degree of reciprocal interaction between the behavioural derivation of features and scene analysis, and the integration evident from the exclusivity of scene organisations. In contrast, model (3) naturally predicts these observations and is compatible with auditory anatomy, which is not “feed forward” or hublike. Finally, we highlight outstanding issues important to a fuller understanding of scene analysis, such as the neural code for perceptual groups and the mechanism for extraction of statistical regularities.
What can binocular rivalry tell us about auditory streaming
Authors: Denham Susan , Winkler István , Gyimesi Kinga , Stefanics Gábor
Published: May 2008

Although sensory systems share the common goal of building accurate representations of the environment, differences in the physical nature of stimuli from different modalities seem to argue against similar processing strategies. Nevertheless, our experiments have revealed surprisingly deep parallels between the dynamics of perceptual organisation in auditory streaming and binocular rivalry. These parallels provide evidence for an interpretation of auditory streaming in terms of rivalry between competing temporal associations, which helps to explain the distribution of perceptual switching with respect to stimulus features, strong differences between first and steady-state phases in ongoing switching, the nonmonotonic dependence of switching rate on stimulus “strength,” the range of relationships between phase duration and stimulus features, and the existence and distribution of “transition” phases during which subjects simultaneously experience what are usually thought to be mutually exclusive perceptual states.
Temporal pitch processing by cochlear implant users
Authors: Carlyon Robert P. , Kong Ying-Yee , Lynch Cathy , Deeks John
Published: May 2008

Modern cochlear implants (CIs) convey fundamental frequency (F0) information using a purely temporal code. However, temporal pitch processing - as measured by the discrimination of the changes in the rate of a pulse train applied to a single electrode - is often worse than in normal hearing (NH) at low pulse rates, and usually deteriorates dramatically at rates above about 300 pps. We will describe evidence that at low pulse rates, the value of the pitch perceived can be affected by refractory effects at the level of the auditory nerve (AN), and can also be influenced by small (< 0.5 dB) changes in stimulus level. In contrast, the deterioration in rate discrimination at high rates appears to be impervious to a wide range of manipulations that would be expected to strongly affect the representation of the stimulus at the level of the AN. One example of this is the finding that the variation in rate discrimination performance with baseline rate for single-pulse-per-period pulse trains correlates, across listeners, with discrimination of different rates of sinusoidal amplitude modulation imposed on a 5000-pps carrier. The implications of these findings for attempts to improve pitch coding in CIs will be discussed.
Parameters affecting the precedence-effect with cochlear implants
Authors: Seeber Bernhard U. , Hafter Ervin R.
Published: May 2008

Monaural spectral and temporal cues as well as binaural cues provide the information to analyze the auditory scene. Subjects using bilateral cochlear implants (CIs) have limited access to those cues, particularly to interaural time differences (ITDs) and to spectral information. Nevertheless, many subjects can localize sounds in quiet based on the evaluation of interaural level differences (ILDs) but this might be impaired by the presence of reflections. We studied the precedence effect with patients wearing bilateral CIs and found that localization was abolished in most subjects when the echo was present, but some subjects showed dominance of the leading sound on localization. The reasons for the different outcomes were studied with simulations of CIs. Using a noise-band vocoder simulation, subjects could not fuse lead and lag into a single object and they localized both separately. With a sinusoidal vocoder, lead-lag fusion and the precedence effect could be obtained provided that frequencies of the carrier sinusoids were matched across ears. Interaural phase of the carrier played only a limited role. We conclude that precedence of ongoing sounds can solely be based on ILDs and envelope-ITDs and that proper place matching of CI-electrodes would help the analysis of concurrent sounds.
Modeling responses of brainstem neurons to electrical stimuli
Authors: Colburn H Steven , Chung Yoojin , Brughera Andrew , Zhou Yi
Published: May 2008

This talk describes modeling efforts to understand brainstem neural responses to electrical cochlear stimulation. Our approach is to combine brainstem models developed for acoustic stimulation with descriptions of auditory-nerve (AN) responses to electric stimulation. Predictions for the behavior of neurons at several levels of the brainstem are compared to available physiological data. Specifically, the AN model predicts differences in discharge probability, degree of phase-locking, and adaptation of discharge rate of the AN response to stimuli, inlcuding both electrical and acoustical cases. Predicted neural responses are generated for several brainstem nuclei, including cell populations in the cochlear nucleus (CN), the superior olive (SOC) and the inferior colliculus (IC). Specific attention is given to two special topics: 1) responses to the simultaneous stimulation of multiple electrodes, and 2) the effects of amplitude modulation on the tuning of binaural neurons to interaural time delay (ITD). [Work supported by US National Institutes of Health grants DC04663, DC00100, and DC05775 (Delgutte, PI).]
Higher sensitivity of human auditory nerve fibers to positive electrical currents
Authors: Macherey Olivier , Carlyon Robert P. , Van Wieringen Astrid , Deeks John , Wouters Jan
Published: May 2008

Most contemporary cochlear implants (CIs) stimulate the auditory nerve with trains of amplitude-modulated, symmetric biphasic pulses. Although both polarities of a pulse can depolarize the nerve fibers and generate action potentials, it remains unknown which of the two (positive or negative) phases has the stronger effect. Animal experiments have shown that cathodic (negative) current flows are more effective than anodic (positive) ones in eliciting neural responses, and this finding has motivated the development of novel speech-processing algorithms. Here we show electrophysiologically and psychophysically that the human auditory system exhibits the opposite pattern, being more sensitive to anodic stimulation. We measured electrically evoked compound action potentials in CI listeners for phase-separated pulses, allowing us to tease out the responses to each of the two opposite-polarity phases. At an equal stimulus level, the anodic phase yielded the larger response. This finding was corroborated by a measure of psychophysical masking patterns and may relate to a particular orientation of the nerve fibers relative to the electrode or to a substantial degeneration/demyelination of the peripheral processes. Potential applications to improve CI speech-processing strategies are discussed.
The response of a complex dynamical system to a complex external forcing
Authors: McCoy John
Published: May 2008

The response of a dominant grounded spring/mass to which are attached a large number of significantly less massive sprung masses, when subject to an external forcing acting on the dominant mass, is revisited. Distinguishing the new study is a focus on the net force that the attached systems cause to act on the dominant mass and the derivation of a semi-analytic, time domain representation of the system operator that transforms the history of the external force acting on the adorned dominant spring/mass to the history of an effective force that acts on the unadorned dominant spring/mass. The mathematical framework provided by the concept of an "effective force" acting on the unadorned simple system gives additional insight to a commonly accepted concept of an "effective dissipation" modifying the unadorned simple system, when subject to simple external force histories; i.e., force histories with limited time extents or with limited bands of spectral content. Of greater practical interest is the mathematical framework allows investigation of external force histories that are not simple.
Dynamic response of an insonified sonar window
Authors: Hull Andrew J.
Published: May 2008

This talk derives and evaluates an analytical model of an insonified sonar window in contact with an array of Tonpilz transducers operating in receive mode. The window is fully elastic so that all wave components are present in the analysis. The output of the model is a transfer function of a transducer element output voltage divided by input pressure versus arrival angle and frequency. This model is intended for analysis of sonar systems that are to be built or modified for broadband processing. The model is validated at low frequency with a comparison to a previously derived thin plate model. Once this is done, an example problem is studied so that the effects of higher order wave interaction with acoustic reception can be understood. It was found that these higher order waves cause multiple nulls in the region where the array detects acoustic energy and that their locations in the arrival angle-frequency plane can be determined. The effects of these nulls in the beam patterns of the array are demonstrated.
Circumferential admittance approach for vibro-acoustic prediction of a submerged cylindrical shell with rib aperiodicity and transversal bulkheads
Authors: Maxit Laurent
Published: May 2008

The circumferential admittance approach is presented to predict the vibro-acoustic behaviour of an externally fluid loaded shell with aperiodic stiffeners and transversal bulkheads. It consists to consider the circumferential spectral displacements of the shell and to use a partitioning of the structure such that the fluid loaded shell constitutes one subsystem and the stiffeners/bulkheads constitute others subsystems. Each subsystem is characterised independently by the circumferential admittances. For the stiffeners/bulkheads, a standard FE code is used to calculate these admittances with a shell description. The admittances of the fluid load shell are estimated by a spectral approach. Developments are proposed in this paper to improve the numerical convergence of the spectral approach. The final assembly is obtained using the continuity relations. Excitations like mechanical forces, acoustic diffuse fields or plane waves can be easily taken into account. For submarine applications, this model permits to study the noise radiated by the pressure hull and to analyse the acoustic target strength of the shell with its internal frames. The whole submarine pressure hull can be described with the assumption that it is extended by a cylindrical rigid baffle. Numerical validations are presented and results at several kHz are shown for a submarine application.
A 'concatenation' study of wave energy transport among three coupled plates
Authors: Wolff Nick , Weaver Richard
Published: May 2008

We address the problem of constructing predictions for the diffuse transport and distribution of mean spectral acoustic energy density in an undamped complex linear structure, without appeal to SEA and by taking information gleaned from short time cost-effective, direct numerical simulations (DNS). A numerical model is constructed consisting of three finite plates coupled by sets of springs. Coupling is chosen to be sufficiently strong to avoid localization and sufficiently weak that the notion of substructures remains valid. Numerical simulations of transient responses to each of several sources are carried out; responses are recorded at each of several receivers. The resulting time-domain diffuse waveforms are analyzed for the work done at each source, and the transient spectral energy density at each receiver at times shortly after the action of the sources. This information is used in a 'concatenation' ansatz to predict the evolution and distribution of spectral energy density at these receivers at later times. The resulting prediction is compared favorably with the actual energy densities observed in the DNS at late times. It is described how the results also apply directly to the more conventional but closely related problem of damped systems and/or steady state input. Work supported by NSF CMS-0528096.
Modification of an interface parameter between sub-system and vehicle: case of a fan system attached to the front end of a car
Authors: Mapagha Saul , Martin Vincent
Published: May 2008

The car industry, as well as many others, is constantly undergoing modifications to comply security and comfort regulations. Among the major causes for concern is the integration of sub-systems on vehicles. In fact, when a sub-system runs on a vehicle, it can lead to vibrations on the body of the car and thus acoustical radiations, causing trouble for the passengers, and also outside the vehicle. The frequency response functions (FRF)-based substructuring technique and impedances coupling methods are used to predict the forces entering the vehicle from those measured on a test bed. When a given constraint on the forces entering the vehicle is not satisfied, the car supplier may choose to modify the interface parameters between the substructures. In this paper an analytical approach to filter the entering forces by elastic suspension is proposed. An application is given by way of numerical experiments on a fan system attached to the front end of a car, both subsystems being identified through measurements.
Efficient computation of the radiated sound power of vibrating structures using a modal approach
Authors: Roy Nicolas , Lapi Marc
Published: May 2008

The optimal design of structures in terms of noise control is of great interest in many fields such as automotive, aerospace, and naval. Since design strategies require simple design parameters, the sound power is often used to characterize the sound radiated from a structure. The CTSN in collaboration with TOP MODAL has recently developed a software tool to efficiently compute the radiated sound power of a vibrating structure using modes obtained by finite element analysis. The modal analysis has been enhanced to take into account the presence of fluid cavities and damping elements via the introduction of residual modes. The radiated sound power is computed from the volume velocities of the vibrating surface using a lumped parameter method requiring no explicit modeling of the acoustic medium. The modal contributions to the total sound power may also be calculated. The underlying methods are described including the improved coupled fluid-structure modal analysis and the computation of the radiated sound power. Next an overview of the tool's architecture is presented. Finally, several industrial applications are presented to illustrate the features and interest of the tool.
The structural acoustic properties of stiffened shells
Authors: Luan Yu
Published: May 2008

Plates stiffened with ribs can be modeled as homogeneous isotropic or orthotropic plates, and modeling such an equivalent plate numerically with, say, the finite element method is, of course, far more economical in terms of computer resources than modelling the complete, stiffened plate. This is important when a number of stiffened plates are combined in a complicated structure composed of many plates. However, whereas the equivalent plate theory is well established there is no similar established theory for stiffened shells. This paper investigates the mechanical and structural acoustic properties of curved shells with stiffening ribs. Finite element simulations and experimental data will be compared and discussed.
Acoustic radiation from vibrating panels subject to high-frequency broadband excitation
Authors: Bliss Donald B. , Franzoni Linda P.
Published: May 2008

Radiation from a baffled panel subject to high-frequency broadband excitation is studied. If the panel flexural waves are subsonic, the radiation comes from edges. For supersonic waves the entire surface is important. Characterization of the radiation is surprisingly straightforward in the high-frequency broadband limit, with simple analytical expressions for directivity patterns. For subsonic waves, a series expansion of the surface velocity wave number transform, convergent in the radiating region, can be reinterpreted physically as singularity functions along the panel perimeter, namely the delta function and its derivatives. This interpretation leads to monopole, dipole, and higher-order edge radiators with relative strengths that depend on surface-wave Mach number. Both propagating and evanescent structural waves contribute to the radiation. A proper physical explanation of the radiation is provided both in physical space and transform space. The common interpretation of edge radiation in terms of uncancelled volumetric sources is not correct; the higher order edge singularities are very significant. For low structural damping, this approach leads to a very simple way to calculate the radiated field. The relationship between radiation and structural power flow is potentially useful in energy-intensity based prediction methods with application to vehicle interior noise. (Sponsor NSF)
Reconstruction of Rayleigh-Lamb dispersion spectrum based on noise obtained from an air-jet forcing
Authors: Larose Eric , Roux Philippe , Campillo Michel
Published: May 2008

The time-domain cross-correlation of incoherent and random noise recorded by a series of passive sensors contains the impulse response of the medium between these sensors. By using noise generated by a can of compressed air sprayed on the surface of a plexiglass plate, we are able to reconstruct not only the time of flight but the whole waveforms between the sensors. From the reconstruction of the direct A0 and S0 waves, we derive the dispersion curves of the flexural waves, thus estimating the mechanical properties of the material without a conventional electromechanical source. The dense array of receivers employed here allow a precise frequency-wave number study of flexural waves, along with a thorough evaluation of the rate of convergence of the correlation with respect to the record length, the frequency, and the distance between the receivers. The reconstruction of the actual amplitude and attenuation of the impulse response is also addressed in this paper [Larose et al, J. Acoust. Soc. Am 122 (2007)].
Automatic testing of speech understanding
Authors: Francart Tom , Moonen Marc , Wouters Jan
Published: May 2008

Speech reception tests are commonly administered by manually scoring the oral response of the test subject. This requires a test supervisor to be continuously present, which can be avoided by having the subject type the response on a computer keyboard and automatically scoring the response. However, spelling errors may then be counted as recognition errors, and hence influence the test results. We demonstrate an autocorrection approach based on two scoring algorithms to cope with spelling errors. The first algorithm deals with sentences and is based on word score. The second algorithm deals with single words and is based on phoneme score. Both algorithms are evaluated with a corpus of typed answers based on three different Dutch speech materials. The percentage of differences between the autocorrection algorithm and the manual correction was calculated, in addition to the mean difference in speech recognition threshold between automatic and manual scoring. The sentence correction algorithm performed at higher accuracy than commonly obtained with these speech materials. The word correction algorithm performed better than the human operator. Both algorithms can be used in practice and allow speech reception tests with open set speech materials over the internet.
A comparison between two methods for speech transmission quality assessment with noise simulation at receiver's side
Authors: Wältermann Marcel , Côté Nicolas , Raake Alexander , Möller Sebastian
Published: May 2008

Two auditory experiments are presented that were designed in order to compare two different methods for the assessment of speech quality when noise at listener's side is present: (1) Headphone presentation with samples pre-recorded over a head and torso simulator, and (2) test participants located in a noisy environment, rendered by a 4.1 loudspeaker system. Real-world types of background noise were employed as well as a variety of user terminals. The correlation between corresponding overall quality scores amounts to r = 0.9. Hence, a headphone presentation can replace a loudspeaker noise simulation in principle, leading to the consequence that a simpler experimental set-up is sufficient for assessing the speech quality when noise is present at the receiver's side. However, there exist certain differences between both quality scores. For example, in most cases clean wideband (50-7000 Hz) and standard narrow-band (300-3400 Hz) speech are rated slightly lower if the samples are presented over a headphone and noise is simultaneously present. Since the separation between speech and noise is potentially easier when the noise is played back over loudspeakers, participants may mainly judge the speech rather than the overall quality, leading to higher scores for the clean channel conditions.
Predicting speech quality under noise in wideband speech transmission
Authors: Ramirez Juan-Pablo , Raake Alexander , Wältermann Marcel , Möller Sebastian
Published: May 2008

Prediction of users' opinion of speech quality in telecommunication is of a major issue in speech transmission planning. For this purpose, reliable models, such as the E-model (ITU-T Rec. G.107, 2005) were developed for narrow-band (NB) transmission. Meanwhile, telephony using the Internet Protocol (VoIP) has enabled an enlargement of the transmission band, introducing wideband (WB; 50-7000 Hz and beyond) to every day communication. The quality improvement thanks to the bandwidth extension was found to be 29%, as compared with NB [3]. Consequently, an updated planning model is required to assess quality in WB speech transmission. We performed two listening only tests to study the quality impairment due to signal attenuation, different noise types and level and various codecs. The tests showed that user's quality perception in case of WB was similar to NB for the noisy conditions. However, considerably higher judgments were given to WB than to NB for the low-noised conditions. With increasing noise level, the quality of NB and WB become more and more alike, and quality differences between codecs decrease. Following the lines of thoughts of the E-model, we propose a model assessing the quality impairment in WB speech transmission resulting from noise and/or signal attenuation.
Influence of informational content of background noise on speech quality evaluation for VoIP application
Authors: Leman Adrien , Faure Julien , Parizet Etienne
Published: May 2008

With the rise of mobile communication systems, the background noise in the speaker's environment and its interaction with VoIP network impairment affects speech quality perception. This effect should be taken into account in nonintrusive models in order to improve accuracy of end user perception measurement. The aim of this study is to determine the impact of information contained in background noise (background speech, environmental sources) on speech quality perception. A subjective test on the speech quality perception in real network conditions has been done comparing the effect of stationary background noise mainly due to transmission equipment (electrical humming or blowing noises) with the effect of nonstationary environmental noise (public place, traffic noise, background conversation). Interactions between these different background noise condition and realistic network impairments (coders and packets loss) were also studied. The speech quality was evaluated through the Mean Opinion Score determined from an Absolute Category Rating method.
Speech processing and dialect variation in the American Midwest
Authors: Clopper Cynthia G.
Published: May 2008

The Midwestern United States is divided into two dialect regions: North and Midland. The Northern dialect is characterized by the clockwise rotation of the low and low-mid vowels in the acoustic-phonetic vowel space. The Midland dialect is characterized by back vowel fronting and the merger of the low back vowels. A series of experiments exploring the effects of dialect variation on spoken word recognition has revealed systematic linguistic processing costs associated with these dialect-based vowel system differences for Midwestern listeners. While word recognition performance in noise is equally accurate for both dialects, the patterns of errors that listeners produce reveal systematic confusions between phonologically different vowels that overlap in the acoustic-phonetic space across the two dialects. In addition, speeded lexical classification performance is highly accurate for both dialects, but performance is slower for Northern vowels than Midland vowels of the same phonological class. The results from these studies suggest that even in the absence of striking intelligibility differences between the two dialects, processing costs associated with the perception of dialect variation can be observed using traditional speech perception and spoken word recognition tasks.
Analysis of total vowel space areas in three regional dialects of American English
Authors: Fox Robert A. , Jacewicz Ewa
Published: May 2008

The acoustic characteristics of vowel systems in different varieties of American English are greatly affected by regional variation. Given the significant positional differences of vowels within the acoustic space across regional dialects of English, one should expect that the size and extent of the vowel space is also affected by this type of variation. Traditionally, the size of the acoustic vowel space has been measured as the triangular area defined by the three corner vowels. An obvious weakness of this approach is that it underestimates the actual "working space" of vowel system in that the onsets and/or offsets of other vowels are often found outside this triangular area. This paper proposes a procedure to estimate the area of a complete vowel space, taking into account dynamic formant pattern of all vowels and diphthongs. Complete vowel space areas are calculated for individual speakers and compared across three distinct regional varieties (representing Inland North, Southern Appalachian, and a Midland variety). The comparison also examines possible changes to vowel space area as a function of speaker generation (younger and older adults).
Stability and variation in the pronunciation of French: A corpus-based approach
Authors: Durand Jacques , Nguyen Noel
Published: May 2008

This contribution relates to the project "La Phonologie du Français Contemporain" (PFC), the largest and most ambitious survey of modern French ever conceived. The PFC project involves over thirty researchers from a variety of countries and aims at the recording, partial transcription and phonetic analysis of over 600 speakers from the francophone world. It aims at a broad coverage of varieties of contemporary French by selecting groups of speakers from approximately 60 different locations in the francophone world. So far around 600 speakers have already been recorded in various parts of the francophone world beside France, including Belgium, Burkina Fasso, Canada, Côte d'Ivoire, Louisianne, and Switzerland. The recordings involve formal and informal conversation, the reading aloud of a word-list and a passage. Current work conducted on the database ranges from auditory exploitation of the data to automatic speech recognition via classical experimental phonetics. We address theoretical and empirical issues related to liaison as well as vowel systems, and examine sociogeographical variation and changes in progress in these respects within French, using a variety of quantitative methods for the empirical assessment of phonological hypotheses.
The temporal location of rms peak in coarticulated vowels
Authors: Jacewicz Ewa , Fox Robert A.
Published: May 2008

The acoustic energy peak of the vowel, here defined as the rms peak, occurs typically before the temporal vowel midpoint. This study examines whether and how the location of the rms peak varies as a function of consonantal context and vowel duration. Four measures of rms peak location were explored: its distance (in ms) from vowel onset and from CVC-word onset (absolute measures) and its relative position (ranging from 0 to 100%) within the vowel and within the CVC-word (relative measures). Analysis of eight American English vowels produced in ten consonantal contexts by twenty speakers shows that the two relative measures yielded the most comparable and consistent patterns. The relative location of rms peak varied significantly as a function of consonantal context. It occurred earlier (i.e., closer to vowel or word onset) in the context of fricatives (and not stops), voiced consonants (as opposed to voiceless), and labials (as opposed to alveolars). The temporal location of rms peak was also closer to the vowel onset (or word onset) for long vowels as opposed to short. Explanation of these patterns will be provided.
An acoustic investigation into coarticulation and speech motor control: high vs. low frequency syllables
Authors: Herrmann Frank , Whiteside Sandra P. , Cunningham Stuart
Published: May 2008

Levelt & Wheeldon's concept of the mental syllabary (1994) suggests that articulatory routines for syllables are stored in the form of gestural scores in a library. Cholin et al. (2006) found a syllable frequency effect on naming latency of nonsense words and interpret this as supporting evidence for the existence of such a syllabary. This paper presents a data set from a project investigating speech motor learning and coarticulation. Acoustic-based coarticulation measures, such as F2 locus equations, are used to indirectly determine the degree of gestural overlap in articulatory movements. Greater degrees of coarticulation can be indicators of cohesive articulatory gestures and therefore, motor speech skill. It is hypothesised that high frequency syllables, as well trained articulatory gestures, should exhibit a greater degree of coarticulation, less variation, and shorter duration times than low frequency syllables. Fifteen native speakers of English listened to, and repeated 16 mono-syllabic stimuli which belonged to either of two categories: high and low frequency (CELEX). Acoustic measurements included the transitions of the first three formants (C-to-V coarticulation), transition times and syllable duration. Patterns of coarticulation were quantified using F2 locus equations and the absolute changes in formant frequency. The results are presented and discussed.
Automatic data enhancement for language identification using voice generation
Authors: Lawson Aaron , Linderman Matthew , Carlin Michael , Stauffer Allen
Published: May 2008

Approaches to LID require very large sets of training (often greater than six hours per language) for accurate results. This study looks at ways of automatically reducing the amount of training data required to train a LID model, while maintaining or increasing accuracy. Initial experiments found that speaker density, i.e., the number of speakers per time unit, had a very dramatic influence on the accuracy of models (absolute increase of 15%). In order to accomplish the goal of increasing the number of speakers available for LID training without having to collect additional audio, the STRAIGHT algorithm was used to synthesize "novel" speakers for use in training language models for a LID system. The mean pitch and vocal tract length of the speaker in each LID training file was scaled to generate four additional voices per original speaker to artifically augment the training data. The resulting models yielded an improvement of 10% over the baseline system (from 80% to 90%). This study shows that automatically generated speakers have a great impact on the accuracy of LID models, in this case an initial training set of 15 minutes per language was augmented and performed comparably to a set trained with six hours.
Channel mitigation approach for automatic language identification
Authors: Stauffer Allen , Lawson Aaron
Published: May 2008

A major obstacle to overcome in language identification (LID) performance is the impact of varying channel conditions. In house experiments show that LID performance drops between 10-12% across channels. The focus of this project is to mitigate the impact of channel conditions and artifacts on LID, and to provide an understanding of how channel robust models may be created by combining data from across corpora. Our approach involved creating composite cross-channel language models from multiple corpora that were tested with data from three corpora whose results were compared to results obtained from same-channel and pure cross-channel experiments. Our hypotheses were that 1) same-channel models would be the most accurate, 2) purely cross-channel models would be considerably less accurate, and 3) composite model accuracy would fall in between that of the same-channel and cross-channel models. Results were surprising: while pure cross-channel tests performed the worst, with an average of 11% loss in accuracy, composite models outperformed matched condition models in tests by 0.5% to 15% for all four combinations tested. These findings provide evidence that cross-channel augmentation does not just make for more robust LID models across channel conditions, but improves model robustness within channel conditions as well.
Newborn pain cry analysis based on pitch frequency tracking
Authors: Bard Delphine , Runefors Per , Arnbjörnsson Einar
Published: May 2008

The aim of the newborn pain cry analysis is to test the hypothesis that cry can be used as a tool to detect signs of nocioceptive pain. Previous studies applying signal processing techniques to analyze the sound of these cries have been done. The subject of this paper is to adapt and improve the original method with the help of new signal processing methods. The pitch frequency is extracted from the waveform of the recorded babies' cries using time domain methods. The fluctuations of this parameter are analyzed in terms of jitter. In particular, a sliding buffer approach is presented, as well as an improvement of the Average Mean Difference Function (AMDF). Comparison between original and news results has been done.