bluuebunny/crossref_metadata_2025 huggingface dataset:
Acoustic wave propagation in cubic piezoelectric semiconductor plates
Authors: Collet Bernard
Published: May 2008

Piezoelectric materials can be either dielectric or semiconductors. An acoustic wave propagating in a piezoelectric crystal is usually accompanied by an electric field. When the crystal is also a semiconductor, the electric field produces currents and space charges, resulting in dispersion and acoustic loss. The interaction between a travelling acoustic wave and mobile charges in piezoelectric semiconductors is currently called the acoustoelectric effect. It was shown experimentally and proved theoretically that an acoustic wave travelling in a piezoelectric semiconductor can be amplified by the application of an initial dc electric field. Piezoelectric semiconductors devices often have structural design of plates or rods. Here we study the thin piezoelectric semiconductor plates. Two-dimensional equations for coupled extensional, flexural and thickness-shear motions are obtained systematically from the three-dimensional equations by retaining lower order terms in power series expansions in the plate thickness coordinate. The two-dimensional equations are specialized to crystals of cubic (43m) symmetry. Propagation of extensional, flexural and thickness-shear waves and their amplification by a dc electric field are analyzed.
Ultrasonic control of the adhesion quality of two aluminium sheets
Authors: Taifi Naima , Faiz Bouazza , Moudden Ali , Maze Gerard , Decultot Dominique , Izbaim Driss
Published: May 2008

In this work, we present two ultrasonic methods allowing to control the quality of adhesion of two aluminium sheets with the same thickness. These methods are based on the analysis of the ultrasonic signals retrodiffused by the Al/glue/Al structure. The first method consists in controlling the behaviour of the mode of the sheets which is splitted. Two parameters controlling the transfers were allowed to characterize any type of adhesion: good, bad and intermediate. The second method is based on control of the width of the mode of the adhesive. The representation of this width by Argan diagram allows to control the quality of adhesion from the measurement of the diagram diameter.
Acoustic tomography of the internal wave-ssociated fluctuations in the lower atmosphere
Authors: Chunchuzov Igor , Kulichkov Sergey , Perepelkin Vitaly , Ziemann Astrid , Arnold Klaus , Kniffka Anke
Published: May 2008

The two different schemes of acoustic tomography of the atmospheric boundary layer (ABL) were used in the field experiments conducted near Melpitz (Germany) and Zvenigorod (Russia). The mesoscale effective sound speed fluctuations (periods 1min-1h) averaged over different acoustic ray paths were retrieved from the fluctuations of sound travel time between sources and receivers. It was found that a major contribution to the retrieved fluctuations comes from the wind speed fluctuations. By using a coherence analysis of the retrieved and measured wind speed fluctuations in the spatially distanced points the wave like fluctuations with periods of 16-20min, 8-10min, 4-5min, 1-2min have been filtered, and their horizontal translation velocities and scales have been estimated. Similar periods were also found in the variations of the vertical turbulent fluxes of momentum and heat near ground. The mechanism of origination of these periods in the observed fluctuations is proposed. The effect of the wind shear variations induced by internal waves on the turbulence intensity was observed. This effect showed a substantial role played by internal waves in the origin of an intermittency of turbulence in the stably stratified ABL. This work was supported by RFBR, grants 06-05-64229, 05-05-64973, 07-05-91555.
Numerical simulation of a thermoacoustic wave amplification
Authors: Hireche Omar , Weisman Catherine , Baltean-Carlès Diana , Bauwens Luc , François Maurice-Xavier , Le Quere Patrick
Published: May 2008

We performed a numerical study of the thermal and physical phenomena occurring in thermoacoustic wave generators. The goal of the simulation is to predict the amplification due to thermoacoustics of a wave initially of small amplitude. Therefore, we focus on the stack and the two heat exchangers, which we call the active cell, which is acoustically compact. The resonator area is split into two parts: the active cell, in which heat transfer takes place, and a resonator, in which the flow is acoustic. The flow in the two-dimensional active cell can be approximated as a low Mach number viscous, conductive flow, subjected to spatially uniform pressure fluctuations. This model is formally derived using asymptotic expansions in terms of Mach number. The focus here is heat transfer between two successive stack plates. The two-dimensional time-dependent problem resulting from this model is solved numerically. Outside the active cell, flow in the resonator is described by a reversible acoustic one-dimensional model. The coupling between the two zones is obtained by matching the velocity at the interface. The acoustics in the resonance tube can be solved using the d'Alembert solution, relating velocities at the interface to velocity values at an earlier time.
Effect of an obstacle on Rayleigh acoustic streaming cells
Authors: Moreau Solenn , Baillet Helene , Valière Jean-Christophe
Published: May 2008

Acoustic streaming has harmful consequences on thermoacoustic machines behaviour because of the associated heat transfers. A preliminary study was carried out in order to study the effect of an obstacle on the Rayleigh cells to help in understanding the role of such phenomena in thermoacoustic machines. An obstacle was introduced in a half-wavelength cylindrical wave guide to study its effects on acoustic streaming. The obstacle was placed at various positions along the wave guide axis and experiments were carried out at various acoustic levels. The axial streaming velocity was measured using Laser Doppler Velocimetry (LDV). It was observed that adding an obstacle in the streaming pattern modifies the latter and that new streaming vortices appear in the vicinity of the obstacle. When the obstacle position approaches a maximum of the Rayleigh streaming velocity the number and the amplitude of acoustic streaming vortices at the ends of the obstacle increase. Similar tendencies were observed when the acoustic velocity amplitude was increased. Because streaming in the vicinity of the obstacle end is complex and has a high amplitude, heat effects can be expected to be important and complex at the ends of the thermaoacoustic stack where heat exchangers are located.
Experimental study of the thermoacoustic effect using infrared thermography
Authors: Feuillet Vincent , Penelet Guillaume , Lotton Pierrick , Camberlein Lionel
Published: May 2008

The thermal metrology used to study experimentally the thermoacoustic effect is almost always based on thermoelectric junctions. This kind of measurement is intrusive and provides a few information in space. This paper presents an experimental set-up where temperature measurement on the thermoacoustic stack is acquired by infrared thermography. This measurement provides more information in space and time to study complex physical phenomena, such as heat transfer through both ends of the stack, non linear edge effects, or optimal spacing between stack and heat exchangers. The experimental set-up consists of a half-wavelength resonator with a square cross section closed by a rigid wall at one end and coupled to an electrodynamic loudspeaker at the other end. The thermoacoustic core is either a single plate or a stack of parallel plates made of Kapton. Temperature measurements are carried out along the stack by the use of an infrared camera. The acoustic pressure is measured by a microphone flush-mounted on the wall at the exit of the resonator. First results show the essential role played by the edge effects and the heat generated by viscous losses.
The contribution of temporal fine structure information to the intelligibility of speech in noise
Authors: Hopkins Kathryn , Moore Brian
Published: May 2008

Temporal fine structure (TFS) information in speech may be particularly useful when listening to a target in a background that contains temporal 'dips'. A change in TFS may allow identification of signal portions where the target-to-background ratio is high. This hypothesis was tested. Speech reception thresholds were measured with steady and an 8-Hz amplitude-modulated noise background for signals processed to contain variable amounts of TFS information. Signals were filtered into channels and channel signals for channel numbers above a 'cut off channel' (CO) were tone-vocoded to remove TFS information, while channel signals with channel numbers of CO and below were left unprocessed. Signals from all channels were combined. Five values of CO were tested for each noise type, with ten normal-hearing subjects. Subjects benefited more from TFS information when listening in the modulated masker than the steady masker. For steady noise, addition of TFS information above 548 Hz did not improve performance, whereas for modulated noise, addition of TFS at high frequencies did improve performance. The greater benefit from TFS information when listening in modulated noise is consistent with the idea that TFS information is important for listening in the dips of a fluctuating masker.
Fine-structure processing, frequency selectivity and speech perception in hearing-impaired listeners
Authors: Strelcyk Olaf , Dau Torsten
Published: May 2008

Hearing-impaired people often experience great difficulty with speech communication when background noise is present, even if reduced audibility has been compensated for. Other impairment factors must be involved. In order to minimize confounding effects, the subjects participating in this study consisted of groups with homogeneous, symmetric audiograms. The perceptual listening experiments assessed the intelligibility of full-spectrum as well as low-pass filtered speech in the presence of stationary and fluctuating interferers, the individual's frequency selectivity and the integrity of temporal fine-structure processing. The latter was addressed in a binaural and a monaural experiment. In the binaural experiment, the lateralization threshold was measured for low-frequency tones with ongoing interaural phase delays. In the monaural experiment, detection thresholds for low-rate frequency modulation were obtained. In addition, these binaural and monaural thresholds were measured in a stationary background noise in order to assess the persistence of the fine-structure processing to interfering noise. Apart from elevated speech reception thresholds, the hearing impaired listeners showed poorer performance than the normally hearing in terms of frequency selectivity and fine-structure processing, despite normal audiometric thresholds at the test frequencies. However, the binaural fine-structure processing was not found to be particularly vulnerable to interfering noise in these listeners.
Intervention for restricted dynamic range and reduced sound tolerance
Authors: Formby Charles , Hawley Monica , Sherlock Laguinn , Gold Susan , Segar Allyson , Gmitter Christine , Cannavo Justine
Published: May 2008

Hyperacusis is an abnormal condition of sound intolerance that may cause some persons to reject amplified sound from their hearing aids. A significant secondary benefit reported for many patients receiving Tinnitus Retraining Therapy (TRT) is increased Loudness Discomfort Levels (LDLs). TRT involves both counseling and sound therapy (i.e., daily exposure to soft sound from bilateral noise generators (NGs)). We implemented a randomized, double-blind, placebo-controlled clinical trial to assess the efficacy of TRT as an intervention to improve sound tolerance in hearing-aid eligible persons with hyperacusis and/or restricted dynamic ranges. Subjects were assigned to one of four treatment groups: 1) full treatment, both counseling and NGs, 2) counseling and placebo NGs, 3) NGs without counseling, and 4) placebo NGs without counseling. They were evaluated at least monthly, typically for five months or more, on a variety of audiometric tests, including LDLs, the Contour Test for Loudness, and word recognition measured at comfortable and loud levels. Over 80% of the subjects assigned to full treatment achieved significant benefit (defined as shifts of greater than 10 dB in LDLs or the Contour Test uncomfortable level); whereas, most subjects assigned to a partial treatment group did not benefit from their treatment. [Supported by NIH].
Critical-band compression method of speech enhancement for elderly people: Investigation of syllable and word intelligibility
Authors: Yasu Keiichi , Ishida Hideki , Takahashi Ryosuke , Arai Takayuki , Kobayashi Kei , Shindo Mitsuko
Published: May 2008

Auditory filters for the hearing impaired tend to be wider than those of normal hearing people. Thus, the frequency selectivity decreases because of increased masking effects [Glasberg and Moore, J. Acoust. Soc. Am., 79(4), 1020-1033, 1986]. We have developed a method, called "critical-band compression," in which the critical band is compressed along the frequency axis [Yasu et al., Handbook of the International Hearing Aid Research Conference (IHCON), 55, Lake Tahoe, 2004]. We investigated whether our method improves syllable and word intelligibility. Thirty one elderly people participated in experiments. First, we measured the auditory filter bandwidth using a notched noise method [Patterson, J. Acoust. Soc. Am., 59(3), 640-654, 1976]. Second, we conducted syllable and word intelligibility tests. The compression rates of critical-band compression were set to 0% for the original, and 25%, 50%, and 75%. The results were that the percentages of correct responses were almost the same at 0%, 25% and 50% compression rates for syllable and word intelligibility. A significant correlation was not obtained between the compression rate of processing and the auditory filter bandwidth. [Work supported by JSPS.KAKENHI (16203041) and Sophia University Open Research Center from MEXT.]
Perceived plausibility of a multi-modal musical performance with introduced auditory-visual spatial and temporal mismatch
Authors: Valente Daniel , Braasch Jonas
Published: May 2008

One of the biggest problems for multi-modal virtual musical performance is that of auditory-visual mismatch. Often such performances place musicians in a dry, studio-like environment and rely on acoustic room modeling techniques to place the musician's sound in an environment more conducive to musical performance. The problem, for example occurs when the musician sounds as if he/she is in a large performance space but from a visual standpoint appears to be located in a very small studio environment. This experiment aimed to reveal the impact of such mismatches on the perceived plausibility of a multi-modal musical performance. Listeners were confronted with three contrasting musical excerpts presented in a virtual environment with varying degrees of auditory visual mismatch. In the first phase, listeners were able to adjust the acoustic modeling algorithm using the salient parameters of direct to reverberant ratio and reverberation time. As the visual space volume that the performance was presented in increased, the listeners repeatedly increased reverberation time and decreased direct to reverberant ratio. During the second phase, where varying levels of spatial and temporal mismatch were purposefully introduced, the level of repeatability and accuracy of the listening group decreased despite subjective responses indicating accurate perceived performance realism.
Audio-visual quality model for internet protocol television services
Authors: Garcia Marieneige , Raake Alexander
Published: May 2008

This paper presents a model for predicting the perceived audio-visual quality of IPTV services. Our model follows a modular approach and audio-visual quality is deduced from the perceived audio quality, the perceived video quality, the interaction between the audio quality and the video quality and the quality of the interaction between audio and video (lip-synchronization). In its current form, the model covers H.264 video codec, Standard Definition and High Definition video resolutions, MP2 audio codec and wav audio format. Addressed degradations, which generate different visual and auditory perceptual dimensions, are compression artifacts, packet losses, reduced bandwidth and delay between audio and video. Results demonstrate a mutual influence of the perceived audio and video qualities and the predominance of the video quality on the overall audio-visual quality. We analyze the interaction between visual perceptual dimensions, like blockiness, blurriness, slicing and freezing, and auditory perceptual dimensions, like frequency content (brightness), interruptedness, and freezing. We also study the influence of the type of the degradation on the interaction between perceived audio and video qualities. At last, we examine the influence of the audio-visual content (music video, news, etc.) on the perceived audio-visual quality. An outlook highlights future model extensions.
The effect of masker type and word-position on word recall and sentence understanding
Authors: Ezzatian Payam , Li Liang , Pichora-Fuller Kathy , Schneider Bruce
Published: May 2008

Speech understanding is influenced by not only the presence, but also the specific nature of maskers. Noise maskers primarily result in energetic masking, whereas speech maskers create additional interference due to linguistic and acoustic similarities to target speech. The present study examined the influence of different types of maskers and target word position on the immediate recall of words in sentences by normal-hearing younger adults. In Experiment 1, nonsense sentences with 3 keywords (e.g., A "house" should "dash" to the "bowl".) were presented against a background of speech-spectrum noise or two-talker nonsense speech. With the speech masker, accuracy increased with word position in a linear fashion. With the speech-spectrum noise masker, performance was highest for the first word but the same for the second keyword and third keyword. In Experiment 2, when the speech-masker was noise-vocoded to preserve envelope information while disrupting fine structure cues and minimizing semantic content, performance was similar to that found with the speech-spectrum noise masker. The results suggest that the ability to track a target sentence in conditions of informational masking improves as the target utterance unfolds over time.
Empirical comparisons of pitch patterns in music, speech, and birdsong
Authors: Tierney Adam T. , Russo Frank A. , Patel Aniruddh D.
Published: May 2008

In music, large intervals ("pitch skips") are often followed by reversals, and phrases often have an arch-like shape and final durational lengthening. These regularities could reflect motor constraints on pitch production or could reflect the melodic characteristics of speech. To distinguish between these possibilities we compared pitch patterns in instrumental musical themes, sentences, and birdsongs. Patterns due to production-related constraints should be common to all three domains, whereas patterns due to statistical learning from speech should be present in speech but not birdsong. Sequences were taken from English and French instrumental classical music, sentences from 4 languages, and songs of 56 songbird families. For sentences and birdsongs each syllable/note was assigned one pitch. For each sequence, we quantified patterns of post-skip reversals, the direction of the initial and final interval, the relative duration of the final syllable/note, and the pitch contour shape. Post-skip reversals predominated in all domains, likely reflecting a shared constraint: skips frequently take melodies toward the edges of the pitch range, forcing a subsequent reversal (as suggested by Von Hippel & Huron, 2000). Arch-like contours and final lengthening were found in music and speech but not birdsong, possibly reflecting an influence of speech patterns on musical structure.
Vibration control of flexible 3D robot arm with join and distributed actuators
Authors: Leniowski Ryszard
Published: May 2008

An active vibration control system is proposed for suppressing amplitude vibration of flexible 3D robot arm. This system integrates control algorithms, intelligent materials and software technologies. The mathematical model of physical system is based upon the geometry and properties of an experimental set-up consisting of a Flex3D robot with a flexible joints and flexible arm. The tip of the arm is loaded by eccentric mass. The vibrations of the plate are measured by the application of a grid of strain sensors and pair of coupled gyroscope-accelerometer. Two kinds of actuators are used. The first is a grid of PZT elements which form a local segments of compensators. Second is a standard BLDC motor located in the join. For the considered system the linear and non-linear (Neural Network of Runge-Kutta type models) of discrete-time model identification for real-time active vibration control have been applied. The mathematical model obtained by this method identification is then employed for the two class of controllers: linear pole placement controller for local segments compensators and non-linear reduced model reference for servo-controller. Virtual simulations are included and discussed.
Influence of temperature on sound transmission through viscoelastic sandwich plates
Authors: Assaf Samir , Guerich Mohamed
Published: May 2008

A numerical study to investigate the effects of temperature on the diffuse sound transmission loss (TL) of sandwich plates is presented. The numerical prediction tool used is based on a finite element formulation for the sandwich plate coupled to a boundary element method for the acoustic medium. The plate formulation is derived from Kirchhoff's theory for the elastic faces and Mindlin's theory for the core. The frequency-temperature dependence of the viscoelastic material properties are taken into account using an experimentally derived viscoelastic constitutive law. The results presented deal with a laminated glass subjected to a diffuse sound field. It is found that the dip of the TL curve at the coincidence frequency of the plate is totally removed for temperatures where this frequency is in the transition region of the used viscoelastic material. Indeed, the relative low value of the storage modulus and the high value of the loss factor in this region induce high transverse shear deformations of the viscoelastic interlayer and thus high energy losses.
Issues in the acoustic modeling of the vocal tract - a progress report on APEX
Authors: Lindblom Björn , Sundberg Johan
Published: May 2008

Faced with an acoustic record of the human voice an investigator is often led to wonder: How was this sample produced? Direct articulatory measurements lacking, the researcher may nonetheless find information of value by resorting to numerical models that relate cavity shapes to acoustic parameters. The classical 'three-parameter models' exemplify this type of tool. The APEX model is another example, developed to help answering questions about the acoustic consequences of articulatory movements. Based on X-ray measurements from a single speaker, APEX converts input specifications for articulatory parameters such as jaw, lips, larynx, tongue tip and tongue body into formant frequencies. The introduction of independent control of the jaw and the tongue and the fact that possible tongue shapes are specified relative to a neutral reference tongue have made significant insights into various topics possible (e.g., coordination jaw/tongue in singing, compensatory articulations). Recently we have been able to increase the physiological realism of APEX representations using MRI and X-ray data. We have also investigated physical models to improve the treatment of certain 3-D front cavity configurations such as raised tongue blade and spread lips. The goal of our paper is to present an overview of the most recent version of APEX.
Feature extraction for vowel recognition using mellin transform
Authors: Jamaati Mahdi , Lankarany Milad , Marvi Hossein
Published: May 2008

This paper proposed a new feature extraction algorithm for vowel recognition using Mellin transform and MFCC. The scale transform is a particular restriction of the Mellin transform. The key property of the scale transform is the scale invariance. This algorithm consists of 3 main stages which are as follow. First stage contains, Extraction spectral envelope from vowels, by using cepstrum method. In the second stage, Mellin transform is used for mapping same vowels with different pitch and different vocal tract to the same waveforms. In third stage, we used a new MFCC algorithm in order to extract main parameters from the output of second stage. The new MFCC algorithm contains third and fourth order cumulant of log-mel-amplitudes. This new MFCC coefficients are robust in presence of additive white noise. Moreover, The LPC method has been used instead of MFCC method in third stage, Experimental result indicates that the new MFCC algorithm produced better result than LPC and original MFCC.
Acoustic tubes with maximal and minimal resonance frequencies
Authors: De Boer Bart
Published: May 2008

This paper presents a theoretical derivation of acoustic tract shapes that minimize and maximize resonance frequencies. The derivation is based on a symmetry of Webster's horn equation and on Ehrenfest's adiabatic invariance hypothesis. It is shown that for minimizing formant frequencies, abrupt transitions are necessary, while for maximizing resonance frequencies, gradual transitions are needed. It is argued that this has implications for modeling human, animal and prehistoric vocal tracts. Such models should represent the anatomic (in)ability to produce abrupt and gradual transitions correctly, otherwise they would have biases towards different sets of formant frequencies than real vocal tracts.
Rhythmic characteristics of prose and verse in varieties of Portuguese
Authors: Bond Zinny , Stockmal Verna , Marks Emilia A. , Woods Audra
Published: May 2008

Whether varieties of Portuguese differ in their rhythmic classification is not entirely clear. European Portuguese is generally considered to employ stress-based rhythm whereas the rhythmic classification of Brazilian Portuguese is disputed. Acoustically-based measures of rhythm have usually employed spoken prose passages for language samples. Possibly, the apparent rhythmic characteristics of language varieties may be clarified by employing different types of language materials. We selected spoken prose and traditional verse for investigation. Five native speakers of European Portuguese and five native speakers of Brazilian Portuguese recorded a sonnet consisting of 154 syllables and a short prose passage consisting of 270 syllables. From these recordings, acoustically based rhythm metrics, as suggested by Ramus, et al. [Cognition 73, 1999, 265-292] and Ling, et al. [Language and Speech 43, 2000, 377-401] were calculated. Measures of consonantal duration variability increased from spoken prose to spoken verse, suggesting that the talkers employed slower, more clearly articulated speech in reading the poem than in reading the prose passage. However, the two tasks did not clearly distinguish between the two varieties of Portuguese.
Experimental study of turbulent flow sound production in presence of a simplified vocal tract constriction
Authors: Estienne Olivier , Van Hirtum Annemie , Pelorson Xavier , Bailliet Helène
Published: May 2008

Sound production due to turbulence is widely shown to be an important phenomenon involved in a.o. fricatives, singing, whispering and speech pathologies. In spite of it relevance, only a few recent models are dealing with turbulence consequences during voiced sound production. The current study presents preliminary results of an experimental characterisation of the aeroacoustical effects of a turbulent flow in case of a constricted channel flow, by means of measuring the velocity fluctuations and the acoustic field downstream of the constriction. Aiming a future application in speech production, the influence of typical vocal tract shape parameters on the velocity distribution and sound field is explored: the tube shape and length as well as the degree, geometry and position of the constriction. The influence of the Reynolds number of the upstream flow is also observed. Results are discussed with respect to the upper airways and human sound production and will be exploited with respect to simplified models for velocity fluctuations and sound production.
Amplitude modulation shape and speech intelligibility
Authors: Prendergast Garreth , Johnson Sam R. , Green Gary G.
Published: May 2008

Research has demonstrated that low frequency amplitude modulations in speech signals are crucially important to maintaining intelligibility. The current work demonstrates a flexible way of characterising the pulsatile bursts of energy found in the temporal envelopes of sub-band filtered speech. Speech was passed through a 128-filter Gammatone filterbank and the temporal envelope of each filter extracted using the Hilbert transform. Thirty-five raised cosine pulses were fitted to model the envelope of each filter and each pulse was defined by its amplitude, half-duration and centre position. The distribution of these pulses demonstrates that the most commonly found pulse half-duration in speech is around 10 ms and few pulses have half-durations longer than 25 ms. Highly intelligible vocoded speech is generated using the extracted pulses and these measures suggest that the auditory system may signal the position in time of the amplitude modulations rather than representing low-frequency information. This method creates a flexible framework within which to further probe the mechanisms involved and allows the ability to focus on cross-channel information in the time domain.
Computerized assessment and training of the perception of American English (AE) speech sounds by adult learners of English
Authors: Miller James , Sillings Roy , Watson Charles S. , Darcy Isabelle , Bardovi-Harlig Kathleen
Published: May 2008

Fourteen volunteer students, with first languages other than English and enrolled in English pronunciation classes, used a specialized software program, the "Speech Perception Assessment and Training System (SPATS)." The software was made available in a language laboratory for seven weeks. Students used the program between 3 and 19 hours, the mean being 13 hours. The goal of SPATS is to train the perception of English words as they are produced in naturally spoken, fluent sentences. The average student correctly identified 1723 words in 313 different sentences as spoken by 10 different talkers and also attempted to identify a total of 7530 presentations of syllable onsets, nuclei, or codas as spoken by 8 different talkers in several phonetic contexts. These syllable constituents were presented in quiet, the sentences in moderate amounts of 12-talker babble. All showed significant progress in perception of AE speech sounds, a few approaching the levels of native speakers. By extrapolation, with 20 to 30 hours of program use nearly all might approach the perceptual performance of native speakers. Those that returned post-training questionnaires viewed the program favorably and recommended its use in regular classes. Speech samples were collected. Accent ratings and spectrographic analyses will be presented.
An MRI Study of the Effect of Vowel Context on English Fricatives
Authors: Shadle Christine , Proctor Michael I. , Iskarous Khalil
Published: May 2008

To gain a better understanding of the long-observed effects of vocalic context, the articulation of fricatives was investigated using Magnetic Resonance Imaging. Five speakers of American English were imaged while producing eight fricatives in the contexts [i-a-u-schwa]. Sagittal, axial and oblique-coronal volumes were acquired for each vowel-fricative combination. Acoustic recordings were made during scans and separately in an anechoic chamber. Vocal tract models were generated by aligning and superimposing all three stack orientations. The models reveal that a variety of articulatory strategies are employed in the production of English fricatives, and that vocalic context is significant. For some subjects, tongue shape differs little with vowel context; other subjects show highly varied tongue shape differences but little difference in lip rounding. Two subjects show significant variation with vowel context for every fricative, including [θ]; two show very little difference, even for [f]. The sublingual cavity in [sh] is extensive for two subjects, but only slightly bigger than in [s] for two other subjects. Tongue grooving in and behind the constriction also varies, and is especially evident in the oblique scans. Overall, more variation and effect of vowel context was observed than expected from the literature. [Funded by NIH-NIDCD-RO1-DC006705]
Increasing speech alignment through crossmodal speaker familiarity
Authors: Miller Rachel M. , Sanchez Kauyumari , Rosenblum Lawrence D.
Published: May 2008

In speech alignment phenomena, individuals inadvertently imitate aspects of another talker's utterances. Recent research has shown that when asked to shadow words, subjects not only align to the speech they hear, they also align to the speech they see when shadowing words by lipreading [Miller, et al., J. Acoust. Soc. Am., 120, 5, Pt. 2 (2006)]. This research also showed that some of the dimensions to which subjects align are the same whether based on shadowing of auditory or visual speech stimuli. This might mean that subjects can align to a speaker's idiolectic dimensions available in both modalities. To examine this possibility, an experiment was conducted to see if alignment increased with exposure to the same or a different speaker, across two blocks of presentations that were: a) both auditory; b) both visual; or c) one auditory and one visual. If subjects align to amodal, idiolectic speaker style, then alignment should be comparable across presentation types in the same speaker condition. Results revealed that alignment increased when the speaker was the same over the course of the two blocks regardless of presentation type. These results suggest that alignment can be based on amodal, idiolectic dimensions which are available across modalities.
Influence of music education on second language acquisition
Authors: Pastuszek-Lipinska Barbara
Published: May 2008

To explore the extent to which music education influences second-language acquisition, two groups of native Polish speakers, musicians and non-musicians, were asked to reproduce sentences in six languages: English, French, Italian, Spanish, Japanese, and Belgian Dutch. The speech stimuli were developed with a text-to-speech application and differed phonemically, phonostylistically, and in length. The paper includes results of a general auditory analysis of subjects' productions as well as the results of a web-based listening test with a panel of native speakers of the involved languages. All collected data were also analyzed with statistical tools. The results revealed that music education exerted a measurable impact on speech perception and production. Musicians outperformed non-musicians in the study. From the results, it appears that the influence of musical expertise extends beyond music processing to speech processing, and the strength of this influence is connected not only to auditory training. Therefore, the superior performance of the musicians in the task may be interpreted as evidence that music education is an enabling factor in the successful acquisition of a second language. It also indicates that the impact of musical training is not a myth, but has a scientific basis.
On the influence of vocal fold collision on phonation
Authors: Neubauer Juergen , Zhang Zhaoyan
Published: May 2008

For laryngoscopic observations show that the closed time of the glottis is significantly long, especially for low-frequency vocal fold vibrations in the vocal fry and chest register. Vocal fold contact appears to be a major part of the phonatory cycle and may play an important role for self-sustained vocal fold oscillations. Using a 2D, finite element, self-oscillating model of the coupled vocal fold-glottal flow system, we studied the influence of the mechanical impact on phonation onset mechanisms and vocal fold vibratory behavior. The air flow was assumed to be laminar and the compressible Navier-Stokes equations were solved for the flow domain. For fixed values of the Young's modulus of the vocal fold we found that vocal fold contact significantly increased the vibration frequency as compared to the case of no contact. The changed total pressure forces on the vocal fold in medial-lateral (lift force) and inferior-superior (drag force) directions resulted in different phonation threshold pressure values and vocal fold vibration patterns. The increase in phonation frequency due to contact will be discussed based on the theory of impact oscillators.
ERPs to words correlate with behavioral measures in children with Autism Spectrum Disorder
Authors: Coffey-Corina Sharon , Padden Denise , Kuhl Patricia K.
Published: May 2008

Children with Autism Spectrum Disorder (ASD) participated in a research study that involved both electrophysiological and behavioral measures. Event related brain potentials (ERPs) were recorded during auditory presentation of known and unknown words. Behavioral measures of language/cognitive function and severity of autism symptoms were also collected at the time of ERP testing and again one year later. In general, higher functioning children with ASD exhibited more localized brain effects for differences between known and unknown words. Lower functioning children with ASD had more diffuse patterns of response to the different word classes and also exhibited a stronger right hemisphere lateralization. That is, they showed differences between known and unknown words at many electrode sites and larger differences in the right hemisphere. In addition, significant correlations were obtained between specific brain wave measurements for both known and unknown words and the various behavioral measures. Patterns of ERPs effectively predicted later behavioral scores.
Magnetoencephalography as a tool to study speech perception in awake infants
Authors: Imada Toshiaki , Bosseler Alexis N. , Taulu Samu , Pihko Elina , Mäkelä Jyrki , Ahonen Antti , Kuhl Patricia K.
Published: May 2008

Magnetoencephalography (MEG) provides a safe, noninvasive method for studying the developing brain by offering reliable localization of the brain regions activated during speech processing. However technical challenges make recording awake infants difficult. The small size of the infant head in the adult-sized helmet results in a low signal-to-noise ratio. Head and limb movement, which is typical of young infants, produces signal artifact that is difficult to overcome during signal processing. This study used MEG to study phonetic processing in awake, non-sedated typically developing infants from 5 to 16 months. The recordings were made using the Elekta Neuromag® 306-channel instrument at BioMag Laboratory, Helsinki University Central Hospital, Finland. Infants listened to speech syllables produced by a loudspeaker inside the magnetically shielded room. Newly developed signal processing methods and behavioral entertainment greatly improved the quality of the data, producing 29 successful infant recordings out of 35 attempts. We describe the methods, as well as removal of movement-modulated artifacts, efficient interference suppression, and movement compensation during data analysis. Whole-head MEG recordings in awake babies a few months old are now feasible.
Detection and classification using the Estimated Ocean Detector
Authors: Culver Richard Lee , Jemmott Colin W. , Bissinger Brett E. , Bose Nirmal K.
Published: May 2008

We have developed the Estimated Ocean Detector, a likelihood ratio receiver with an estimator-correlator structure, and applied it to detection and classification of underwater acoustic signals. The receiver requires that the noise probability density function (pdf) to belong to the exponential class but need not be Gaussian. A composite hypothesis is employed in order to incorporate knowledge (or predictions) of the signal parameter statistics. Previously, receiver performance was demonstrated for Gaussian noise and sinusoidal signals with known frequency and phase and whose amplitude pdfs were predicted using knowledge of the ocean environment and an acoustic propagation program. In order for the receiver to be useful operationally, it must be able to accommodate unknown signal phase and incorporate numerical estimates of noise and signal parameter pdfs. Progress toward satisfying these requirements is reported in this talk. Work supported by the Office of Naval Research Undersea Signal Processing.
An experiment for signal identification of the MIMO communication by sonic waves
Authors: Hayashi Daisuke , Maeda Yasukazu , Itakura Toru , Shibayama Hideo
Published: May 2008

Recently, the communication method by MIMO works in a field of the wireless communication. We do research on application to the MIMO system in an acoustic field by sonic wave. As the transmission method, we use the space division multiplexing (SDM) that is the method for the purpose of the improvement of the transmission rate that accepted the number of the transmission elements by sending plural signals at the same time. It is important to identify each signal from several different signals that are transmitted at the same frequency band. And, we study method for detecting each signal. This paper describes influence of the signal detection for the different conditions of the multi-path propagation.
Transverse vectorization of fast Fourier transforms on multicore architectures
Authors: Humble Travis , Barhen Jacob , Traweek Michael
Published: May 2008

Single-instruction, multiple-data (SIMD) multicore computing architectures, such as the IBM Cell Broadband Engine Architecture, offer new opportunities for quickly and efficiently calculating the 1D-FFT of acoustic signals, as time-sampled data arrays can be naturally partitioned across the multiple cores on which vectorized implementations of the FFT operate. Building on this parallel pipeline model, we consider the case that M data arrays of length N reside within each core. Whereas the cost of sequentially executing these M FFT's conventionally scales as αMNlog2N, we demonstrate a transverse vectorization solution whose cost scales as αβNlog2N, where α and β are constant scaling factors. Our approach makes use of the SIMD instruction set and large vector register file inherent to each core of the IBM Cell in order to calculate the FFT of M data arrays simultaneously. By efficiently using all the available vector registers in performing the FFT, this transverse SIMD vectorization solution further reduces the computational complexity of the conventional parallel pipeline model.
Physics-based signal processing methodologies for separating target echoes into their constituent elastic and geometric components
Authors: Kirsteins Ivars , Tesei Alessandra
Published: May 2008

Echoes from even simple canonical symmetric shapes such as solid elastic spheres are highly complex, consisting of a superposition of specular, elastic, and geometric diffractive components closely spaced in time and frequency. Although measurement of the individual elastic and geometric components is important for analysis of scattering physics and object identification, in practice individual components are often hard to measure because of mutual interference between components. When an object interacts with a boundary, such as the sea floor, additional interfering echoes make the object signature even more complicated to interpret. Motivated by this problem, a frequency-domain technique is developed for separating an echo into its constituent elastic and geometric components, and for isolating specular reflections from a possibly interacting boundary. It is based on modelling the echo components in the frequency domain as amplitude-modulated piecewise-linear chirps, utilizing physically-inspired group delay models and constraints, and then using a modified Costas' residual signal analysis (RSA) algorithm in combination with chirp transform analysis to decompose the echo. The RSA scheme is demonstrated on actual echoes collected in the NURC EVA-06 sea trial from spherical and cylindrical target shapes and is shown to work well with the separated components verified against modelling predictions.
Computer Vision Techniques Applied for Reconstruction of Seafloor 3D Images from Side Scan and Synthetic Aperture Sonars Data
Authors: Bikonis Krzysztof , Stepnowski Andrzej , Moszynski Marek
Published: May 2008

The Side Scan Sonar and Synthetic Aperture Sonar are well known echo signal processing technologies that produce 2D images of the seafloor. Both systems combines a number of acoustic pings to form a high resolution image of seafloor. It was shown in numerous papers that 2D images acquired by such systems can be transformed into 3D models of seafloor surface by algorithmic approach using intensity information, contained in a grayscaled images. The paper presents the concept of processing the Side Scan Sonar and Synthetic Aperture Sonar records for detailed reconstruction of 3D seafloor using Shape from Shading techniques. Shape from Shading is one of the basic techniques used in computer vision for the objects reconstruction. The algorithms proposed in the paper use the assumed Lambert model of backscattering strength dependence on incident angle and utilize additionally the information from shadow areas for solving obtained set of equations. The idea was verified by simulation study. The obtained results of 3D shape reconstruction are presented and the performance of the algorithms is discussed.
Gaussian beam tracing for high-frequency acoustics
Authors: Porter Michael B. , Siderius Martin , Hursky Paul
Published: May 2008

Gaussian beam tracing is an approach that constructs full-wave beams around the skeleton of conventional ray theory. This Gaussian beam approach leads to a very simple algorithm and provides remarkable accuracy and speed. For high-frequency, broadband applications, Gaussian beams are often the only practical approach, as the standard full-wave modeling alternatives are often thousands of times slower. Gaussian beams are derived using high-frequency asymptotics, and therefore fit naturally to certain current areas of interest in HF acoustics, such as acoustic communications. However, what often surprises people who are not familiar with the technique is that it works quite well at lower frequencies, depending on the water depth. That fact, is really a consequence of 20 years of continued advances in the Gaussian beam method. We will review those developments in the context of HF variability, considering effects of boundary, volume, and source/receiver dynamics.
Field-calibration: exploiting high-frequency mobile platform transmissions for source localization at lower frequency with arrays
Authors: Hursky Paul , Porter Michael B.
Published: May 2008

Multipath arrivals in an ocean waveguide with a reflective enough bottom can be used as a fingerprint for source range and depth and this has been demonstrated by many researchers using matched field processing. However, MFP relies upon acoustic propagation models to produce the Green's functions to be used as steering vectors for this processing. We have proposed to instead directly measure these Green's functions using wideband acoustic comms signals from mobile platforms such as AUVs that are increasingly part of naval applications. Thus we measure the multipath arrival pattern at 8-16 kHz from the many locations AUVs are visiting, and then apply such measurements for locating other sources at low frequency. A key factor in this process is being able to capture the essential features of the impulse response function at high frequency, where fluctuations are much more severe than at low frequency. We have previously presented experimental results of applying this technique using a single hydrophone receiver. Here we will report on a continuation of that work, using vertical arrays and data from the RADAR '07 experiment.