bluuebunny/crossref_metadata_2025 huggingface dataset:
Uncertainty of noise mapping software
Authors: Wierzbicki Jacek , Batko Wojciech
Published: May 2008

Noise map is a tool to come to noise policy and especially to an action plan. The most important question connected with noise mapping realized by computational methods is about reliability of results. It is closely connected with quality of input data and calculation algorithms and their parameters. An influence of elements location, buildings height and number of vehicles accuracy on calculated results was considered in first stage. Before implementation of Harmonoise/Imagine P2P algorithms there are differences between various software. The paper presents first results from investigations with two commercial prediction programs. The absolute accuracy (uncertainty) of predicted by software sound levels is very difficult to quantify because an uncertainty in the measurements. The main result of investigations is a table of differences in sound level calculations related to variation of input data (geometry of objects and acoustic parameters) and calculation algorithm.
Formation of ripples on sand surface as result of nonlinear interaction of sound waves and wind drift particles
Authors: Vilchinska Nora A.
Published: May 2008

The phenomenon speaks within the limits of nonlinear acoustics, considering it as amplification of surface acoustic waves (SAWs) by drift of carriers. Interaction and amplification of SAWs, borne from falling sand particles on dry sand surface and drift sand particles under wind action, is responsible for the first ripple forming on a dry sand flat surface. SAWs interact with drift sand particles in the moment, when the velocity of drift is equal to velocity of a sound propagation. The sound wave acts on drifting particles so that sand grains, a little bit advancing wave, will be braked, and lagging behind be accelerated. Particles will gather in a phase of a wave - take place grouping of particles and ripple forming. The distance between groups is a ripples wave length λ. Experimental in-situ are measured weak sound field on frequencies 25 Hz in just forming sand massif, and near surface velocity of sand drug under wind action. Measured ripples wave lengths λ are in good agreement with measured diapason of wind velocitys. Measurements were carried out on the sand beach of the gulf of Riga, Latvia. Theoretically the phenomenon speaks within the limits of nonlinear acoustics or within the limits of nonlinear mechanic.
Theoretical proof of acoustic source property single definition in liquid and solid
Authors: Zagorskiy Leo
Published: May 2008

The problem of acoustic waves property sourse single definition consists in unknown density and elastic modules of the medium. The Green's function definition is difficult as well as the real sourse isn't point and has the time dependence. In the early author's papers Green's function had been defined by using Levitan's polynomials. In the liquid or solid medium it is necessary for that to know only the first eigen values and shear or P-wave velocity at the first step from the free surface. It provides to calculate amplitudes of first mode of SH waves, α and β - the limits of lacunas in the spectrum. The time function of not moving sourse F is defined from the equation F* G≡U, where U is displacement, G- Green's function, * denominates rolling up. That equation is solved by direct and inverse fast Fourie transform. The further definition of source's location from the direct task for P or S body waves with defined velocity isn't hard by using modern finite element method. The rounding of the fronts gives the location of the sourse.
The effect of spectral overlap on sensitivity to interaural time differences
Authors: Brown Christopher A. , Yost William A.
Published: May 2008

Cochlear implant users who have some residual low-frequency hearing in both their implanted and unimplanted ears may have access to interaural time differences (ITDs), which could provide a cue for sound source segregation. Speech intelligibility has been shown in simulation to improve under these conditions. However, implant patients who stand to benefit from ITDs may have a significant asymmetry in the configuration of loss in the low-frequency region. It is unknown how a difference in corner frequency of loss across ears will affect ITD sensitivity. As a first step toward characterizing this relationship, the present study measured ITD just-noticeable differences (JNDs) of bands of noise that varied in the amount of spectral overlap across ears in five normal-hearing listeners. In one ear, the center frequency was 250 Hz, and the bandwidth was either 1/3 or 2/3 octaves. In the other ear, the center frequency was 250, 281, 315, 354, 397, or 794 Hz, and the bandwidth was 1/3 or 2/3 octaves. Results showed that although ITD JNDs increased as spectral overlap decreased, ITDs were only detectable so long as some spectral overlap was present.
Perception of speaker's facing angle
Authors: Kato Hiroaki , Takemoto Hironori , Nishimura Ryouichi
Published: May 2008

In pursuit of an ultimately realistic human-to-human telecommunication technology, the ability to auditorily perceive the facing direction of a human speaker was explored. A male speaker sat on a pivot chair in an anechoic chamber and spoke a short sentence (about 5 seconds) while facing either of eight azimuth angles (0=listener's direction, 45, 90, 135, 180, 225, 270, or 315 degrees) and either of three elevation angles (0=horizontal direction, -45, or 45 degrees). The azimuth angles were set solely by turning the pivot chair. Twelve blindfolded listeners heard the spoken sentence at a distance of either 1.2 or 2.4 meters from the speaker and were asked to indicate the speaker's facing angle. In separate sessions, the speaker changed facing angles while speaking and the listeners indicated the perceived direction of horizontal movement (clockwise or counter-clockwise) or vertical movement (up or down). Overall results showed that the listeners were more accurate in indicating the movement of the speaker than in simply indicating the facing angles. Effective acoustic cues were then discussed on the basis of the transfer characteristics from the speaker's mouth to the listener's ears measured by the cross-spectral method using the speaker's own voice.
Human cortical responses to change in sound lateralization suggest non-topographical coding of interaural time differences
Authors: Magezi David A. , Krumbholz Katrin
Published: May 2008

It is generally assumed that interaural time differences (ITDs) are coded by a parametric array of finely tuned ITD channels (topographic code). An alternative theory is that ITD is coded by a rate code, which involves only one channel in each hemisphere, broadly tuned to the contralateral hemispace. The aim of the current study was to investigate ITD coding in humans by using electroencephalography (EEG) with a continuous stimulation paradigm (CSP). In this paradigm, a test sound, with a given ITD, is immediately preceded by an appropriately matched control sound with a different ITD. At the transition from the test to the control sound, only neural elements that respond to the test but not the control ITD would be expected to be activated. Thus, the size of the transition response would be assumed to be determined by the degree of overlap between the neural responses to the control and test ITDs. The pattern of results was found to be inconsistent with the topographic model, and thus lend support to the hemispheric channel model.
Modeling the precedence effect in inferior colliculus neurons using converging excitatory and inhibitory inputs
Authors: Xia Jing , Brughera Andrew , Colburn H Steven , Shinn-Cunningham Barbara
Published: May 2008

Responses of a population model of low-frequency inferior colliculus (IC) neurons to a pair of broadband clicks were evaluated. IC inputs came from Hodgkin-Huxley models of medial superior olive (MSO) and cochlear nucleus bushy cells driven by a stochastic auditory nerve model. Each IC cell received converging inputs that were excitatory from ipsilateral MSO and inhibitory from contralateral MSO. Convergence of multiple excitatory, ipsilateral MSO inputs with similar interaural time difference (ITD) tuning improved ITD sensitivity and reduced the influence of MSO onset responses caused by monaural, rather than binaural, coincidences. Inhibition that lasted over several milliseconds suppressed IC responses to the lagging click. The effect of the inhibition depended on the ITD tuning of the excitation and inhibition converging on a given IC neuron. With appropriate convergence, the IC population showed no response to the lagging click at short delays, and responded to the lagging click but did not encode its ITD at intermediate delays. Only at longer inter-click delays did IC responses show sensitivity to the lagging click ITD, consistent with the perceptual phenomenon of the precedence effect. [Work supported by NSF and NIH].
Perceptual pitch compensation for low frequency bandwidth extension
Authors: Seung Kim , Sung Koeng-Mo
Published: May 2008

As the size of speakers decreases with miniaturization of audio devices and thickness of TV, there is increased demand to ensure low-frequency sound quality. Psychoacoustics bandwidth extension for low-frequency has been applied to recent audio devices. A perceptual pitch does not correspond with fundamental frequency of missing fundamental, because the perceptual pitch is heard highly than desirable fundamental frequency. Consequently, the algorithm compensating perceptual pitch with difference is needed. This paper describes the algorithm compensating the difference using mel scale curve, and the listening test result of sound source applying the algorithm in comparison with sound source applying conventional algorithm.
The noise induced harmful effects assessment using psychoacoustical noise dosimeter
Authors: Kotus Jozef , Kostek Bozena , Czyzewski Andrzej , Kochanek Krzysztof , Skarzynski Henryk
Published: May 2008

A new way of assessment of noise-induced harmful effects on human hearing system was presented in the paper. Employing the developed psychoacoustical noise dosimeter the new indicators of noise harmfulness were verified on the basis of hearing examinations and noise measurement results. The indicators were based on some psychoacoustical properties of the human hearing system and, at the same time, on evaluation of the time and frequency characteristics of noise. Additionally, time properties of the Temporary Threshold Shift are calculated during the noise exposure. The evaluation of the proposed indicators were conducted on the basis of hearing examinations in the real noise exposure situations and also on the basis of simulation results employing standard test signals (such as: white, pink and brown noise). The standard noise dose analysis results were also presented for the purpose of comparison. The performed analysis and obtained results confirmed correctness and practical usefulness of the proposed indicators.
Subjective and objective evaluation of localisation accuracy in wave field sysnthesis
Authors: Sanson Joseph , Corteel Etienne , Warusfel Olivier
Published: May 2008

Wave Field Synthesis (WFS) is a holophonic technique that relies on the reproduction of physical properties of sound fields in an extended listening area. Limitations of this technique are due to approximations regarding the mathematical basis and have been studied in [Ber88] [dB04]. The synthesis remains correct up to a corner frequency referred to as the spatial aliasing frequency and inside an area limited by the length of the WFS array and the position of the virtual sources. A Multichannel Equalization filter (MEQ) is applied to the driving limentation signals of the loudspeakers, as described in [Cor06]. The aim of the present study is to evaluate localisation in WFS depending on both listening and source positions. An objective analysis of the binaural signals is compared to the results of a localisation task for which spatialised high-pass filtered white noises served as stimuli. The results of the psycho-acoustic test are compared to the binaural localisation cues, namely ITD (interaural time differences) and ILD (interaural level differences). Above the aliasing frequency, ITD and ILD are conflicting and ILD errors account in most cases for the perceptual errors.
The ability of temporally integrating acoustic waveforms is associated with release of speech from informational masking under reverberant conditions
Authors: Li Liang , Huang Ying , Huang Qiang , Chen Xun , Wu Xihong H.
Published: May 2008

Perceptual integration of the sound wave directly emanating from the source with reflections of the source needs both bridging temporal gaps and calculating correlations between sound waves. In this study, we examined whether the temporal integration of source/reflection signals is frequency dependent and associated with speech unmasking under simulated reverberant conditions. In Experiment 1, a break in correlation (BIC) between correlated wideband or narrowband noises at the two ears was detectable even when an interaural interval (IAI) was introduced. The longest IAI varied markedly across listeners. In wideband noise, it could be up to 21 ms; in narrowband noise, it decreased as the center frequency was increased. In Experiment 2, when the interval between target speech and its single-reflection simulation (inter-target interval, ITI) was reduced from 64 to 0 ms, intelligibility of target speech was markedly improved under the speech-masking condition but not the noise-masking condition. The longest effective ITI under the speech-masking condition significantly correlated with the longest IAI for detecting the BIC in low-frequency (< or = 400 Hz) narrowband noises. Thus the frequency-dependent temporal integration of sound-waveform signals is critical for releasing speech from informational masking in reverberant environments. Supported by the National Natural Science Foundation of China.
Perception of speech and environmental sounds in cochlear implant patients
Authors: Shafiro Valeriy , Gygi Brian , Cheng Min-Yu , Mulvey Megan , Holmes Blythe
Published: May 2008

Limited research has investigated cochlear implant (CI) patients' ability to perceive environmental sounds. In this study, environmental sound perception was assessed with a large-item test composed of 40 sound sources, each represented by four different tokens. The relationship between speech and environmental sound perception and the role of working memory and basic auditory abilities was examined based on patient performance on a battery of speech tests (HINT, CNC, and individual consonant and vowel tests), tests of basic auditory abilities (audiometric thresholds, gap detection, temporal pattern and temporal order tests), and a backward digit recall test. Twelve postlingually deaf adult CI patients, having 1-5 years of implant experience participated. The results indicate substantially reduced ability to identify common environmental sounds in CI patients. Speech test scores correlated strongly with the scores on the environmental sound test. Both speech and environmental sound tests moderately correlated with gap detection, temporal order test and backward digit recall test. However, the correlation between speech and environmental sounds changed little after partialling out the variance due to other variables. These results suggest that speech and environmental sounds may overlap considerably in their perceptual processing, being largely independent of peripheral limitations that may affect both sound classes.
The effects of induced loudness reduction on tone-burst otoacoustic emissions
Authors: Epstein Michael , Marozeau Jeremy
Published: May 2008

Induced loudness reduction (ILR) is a phenomenon by which the loudness of a sound is reduced when it is preceded by a higher-level sound (inducer). Because the effects of ILR can last for at least several minutes after exposure to an inducer, many clinical and laboratory test procedures may be unintentionally confounded. However, little is known about the physiologic manifestation of ILR or how it might affect clinical tests. The present study examined whether ILR could be observed in otoacoustic emission measurements, which have been shown to correlate well with loudness. Otoacoustic emissions were measured monaurally in 12 normal-hearing listeners in response to a 1 kHz 70 dB SPL tone-burst stimulus before and after a series of 1 kHz 90 dB SPL tone-burst inducers. Additionally, otoacoustic emissions were measured in two control conditions in which ILR was not expected to occur. For most listeners, the level of otoacoustic emissions decreased after inducer exposure and did not decrease in the control conditions. The results indicate that at least some component of ILR is likely to arise as a cochlear process and that although the observed effects are small, ILR may affect measures beyond just loudness judgments.
Modelling categorical loudness perception for arbitrary listeners and sounds
Authors: Kollmeier Birger , Appell Jens E. , Verhey Jesko , Hohmann Volker
Published: May 2008

While "classical" loudness models predict loudness in sone using the concepts of Stevens' compressive power law, (subdivided) categorical loudness perception after Heller follows the compressive logarithmic Weber-Fechner law. To bridge the gap between both approaches, this contribution reviews various steps towards a loudness model that predicts categorical loudness (in categorical units, CU) for normal and hearing-impaired listeners for arbitrary sounds. It uses a (modified) classical loudness model for stationary signals to derive the loudness in sone and a nonlinear transformation from sone to CU. This transformation is approximated by a cubic polynomial equation wich is derived from categorical loudness data of 84 normal-hearing subjects. The model parameters are further set to predict the standard isophones that are in good agreement with the equal loudness level contours derived from categorical loudness data. Also, the model predicts the loudness functions near threshold both for normal and hearing-impaired listeners and can be extended to predict duration-dependent loudness perception. Since categorical loudness can be measured more easily and directly than loudness in sone, the current modelling approach can be experimentally tested and can be used in various applications, such as, e.g. hearing aid processing and fitting procedures.
Analysis of enclosed sound fields using spherical microphone array processing
Authors: Rafaely Boaz
Published: May 2008

Enclosed sound fields in rooms and auditoria are characterized by multiple reflections in addition to the direct sound. The spatial and temporal manner in which sound propagates from the sound source to a listener determines the acoustics of the enclosed space. Recent studies showed that spherical microphone arrays and associated array processing can be used effectively in the experimental study of the acoustics of auditoria. The main advantages presented are the true three-dimensional analysis and the effectiveness of array processing in the spherical harmonics domain. Spherical microphone array design issues will first be reviewed in this work, in particular concerning the design of open-sphere arrays useful in room acoustics analysis with high spatial resolution. Then, beam-forming methods will be presented which can facilitate identification of individual room reflections, and computation of directional room impulse responses. Finally, the use of optimal array processing methods will be presented which facilitates improved sound field analysis.
Measurement of low-frequency sources in non-anechoic room using near-field acoustic holography
Authors: Melon Manuel , Langrenne Christophe , Garcia Alexandre , Herzog Philippe
Published: May 2008

Low frequency (20 Hz-200 Hz) measurements of sound sources are rather difficult to perform because free field conditions can not be easily achieved properly. Moreover, some industrial sources have to be measured in situ. In such a case, a Field Separation Method (FSM) can be used to subtract the pressure field reflected by walls of the testing room from the measured data. This approach required the knowledge of both acoustic pressure and velocity on a closed surface surrounding the source. In this paper, a spherical harmonic expansion of measured data is used to solve the problem. The proposed method is applied to the measurement of the frequency response of a closed box subwoofer tested under various conditions: in a room with variable reverberation time (6.4 s to 0.6 s). Theoretical frequency response of the subwoofer is also calculated using the Thiele and Small model. Results show a good agreement between separated data and simulations. The influences of the measurement distance and of the measurement point number required on the separation process are discussed.
An improved method of visualizing the energy flow: the Rayleigh wave
Authors: Dean Cleon E. , Braselton James P.
Published: May 2008

A variety of ways of visualizing the energy flux or elastodynamic Poynting vector field are demonstrated including gridded vector field, color coding, and other methods for the simple example of a Rayleigh wave. In particular an improved version of a technique shown at the 8th International Conference on Theoretical and Computational Acoustics, ICTCA 2007 is demonstrated [David M. F. Chapman, "Visualizing acoustic energy flow into the seabed using energy streamlines," Eighth International Conference on Theoretical and Computational Acoustics, ICTCA 2007, Heraklion, Crete, GREECE, 2-5 July 2007.]. The improved streamline method shows both the direction and the intensity of the energy flow.
Implementation of the high order asymptotic models of linings and coatings to the SAW spectra evaluation
Authors: Zakharov Dmitry , Shuvalov Alexander , Poncelet Olivier
Published: May 2008

High order asymptotic models of the relatively thin linings and coatings are used to reduce the problem dimension when dealing with a reasonable frequency range. The essential point consists in the use of asymptotic expansion for the internal stress and strain state of a laminate, subjected to the tight conditions at least on one of its faces, i.e., the conditions formulated for the displacements. The respective algorithm, based on it, is shorter than for the direct evaluation of the SAW spectra. It can be implemented to the calculation of Rayleigh waves or Lamb waves in layered media. Numerical examples are presented with both asymptotical and numerical evaluation of error.
Optimisation of building vibration isolation by means of numerical methods
Authors: Augusztinovicz Fülöp
Published: May 2008

Ground-borne vibrations pose a steadily increasing environmental problem in densely populated areas of large cities. Although not the cheapest, the application of resilient elements in the construction of new buildings is an effective and viable method. Obviously, experimentation with parameters and placement of the isolators is not possible in this case. Therefore, a careful and watchful optimization of the static and dynamic characteristics of the supporting elements is a must. The proposed paper gives an overview of the vibration isolation design and reports on the expected effect of some basic design versions of a typical multi-storey building. Standard structural finite element method is used as a basic design tool, but a number of vibro-acoustic aspects are also considered and treated. Experimental results as well as comparisons of measurements and predictions will be reported too.
The investigation on sound souce indentification in semi-space by NAH
Authors: Shang Dejiang , Liu Yongwei
Published: May 2008

The near field acoustic holography (NAH) theory model in semi-space has been constructed on the foundation of NAH theory in full-space. The technique of semi-space acoustic field reconstruction by NAH has been developed on Fast Fourier Transform Algorithm. This method has considered the water surface as soft boundary condition. The equivalent free sound field can be acquired if the measured sound field has been handled dissymmetrically by the principle of the mirror. This method can overcome the disadvantages of the boundary element method which can be used to resolve the problems of sound source identification in semi-space, such as the using of the semi-space Green function, complex transformation matrix calculation, more time consummation and singularity solution existence. This method has been verified by numerical calculation and by experiment respectively. The experimental frequency is between 3 and 10 kHz. The distance between the hologram plane and source plane is 3 centimeter. The results show that it's an efficiency way to deal with the problem of sound source identification in semi-space.
Eigenvalue imaging method for subsurface defects detection
Authors: Teramoto Kenbu
Published: May 2008

This paper proposes the eigenvalue imaging method to be independent of frequency and phase velocity and to characterize the Lamb-wave field. Lamb-waves play important roles in NDE fields. Being placed on the surface of a plate, a sound source excites several kinds of waves: (a) pressure and shear waves, (b) a Rayleigh wave, and (c) symmetric and anti-symmetric Lamb-waves traveling along the plate. The first two waves are utilized for traditional ultrasonic techniques such as pulse-echo methods. In contrast to these classical techniques, in which the wavelength is shorter than the plate thickness, the Lamb-wave has the advantage of propagating over large area. The variation of the Lamb-wave velocity, however, causes difficulties for interpretation of observed signals. It is important, therefore, to establish the crack detection criterion independent of local wave numbers. This paper proposes a novel imaging method which utilizes the 3rd and 4th eigenvalues of the covariance matrix, which is defined over the 4- dimensional vector space which is spanned by following components: (1) a vertical displacement, (2) its particle velocity, (3) the orthogonal pair of out-of-plane strains. This study shows that the 3rd eigenvalue detects a wave scattered by defects and the 4th one has an ability to localize defects.
fMRI evidence for central auditory processing of speech in deaf infants under sedation
Authors: Holland Scott K. , Karunanayaka Prasanna , Rajagopal Akila , Smith Kristen
Published: May 2008

Functional MRI (fMRI) performed in infants with congenital hearing loss provides evidence that auditory language stimulation produces activation of the central auditory system in the brain. Babies are normally sedated for clinical MRI scans. With IRB approval we have added a 10 minute fMRI scanning procedure to clinically indicated MRI scans in infants with severe to profound hearing impairment (n=18) or normal hearing (n=22). Sedation is performed according to clinical guidelines at our institution using either Propofol ((200-250 mcg/kg/min i.v.) or pentobarbital (5 mg/kg orally). fMRI scanning is performed using Echo-planar gradient echo acquisitions on a 3 Tesla clinical MRI system while a mothers voice reads stories to the babies at a sound level 10 dB above the measured hearing threshold. Brain activation is measured in primary auditory cortex and planum temporale speech recognition areas. Results in the normal hearing group of infants demonstrates that the central auditory system is stimulated by speech in infants, although sedation does attenuate brain activation in a dose dependent manner. Central auditory and language activation is also detected in hearing impaired infants with levels of activation correlating to measured hearing thresholds.
Muscle saturation effect in /i/ production: Counterevidence from a 3D biomechanical model of the tongue
Authors: Buchaillard Stéphanie , Perrier Pascal , Payan Yohan
Published: May 2008

Numerous works suppose the existence of a saturation effect to facilitate the control of the constriction area when producing vowel \i\. This study exploits a 3D biomechanical Finite Element model of the vocal tract to evaluate the effectiveness of this assumption. The model includes the tongue and its major muscles, represented as a hyperelastic body, inserted in the oral cavity including jaw, palate, pharyngeal walls and hyoid bone. Muscles activations are controlled with the lambda model (Equilibrium-Point Hypothesis). After determining a set of possible motor commands to generate vowel \i\, the impact of local variations of these commands on tongue shape was studied. The main tongue muscles motor commands were independently modified (by ±2%, 5%, 8% around their values at target) and the effect on the tongue shape and on the acoustic signal was studied. The simulations showed a global stabilization of the tongue body, thanks to the palatal contacts, but also variability in the alveolar groove due to the anterior genioglossus activation. Though extremely localized, this variation has an important impact on the constriction size and, then, on the acoustic signal. Consequently, a specific control of the articulatory variability could be necessary to explain experimental data for vowel \i\.
Contrastive Focus in Taiwan Mandarin
Authors: Pan Ho-Hsien , Tsou Hui-Ying , Huang Wan-Ting
Published: May 2008

This study investigated the production and perception of given, new, and contrastive information in Taiwan Mandarin. Spontaneous dialogues elicited through a game were analyzed to access the acoustical parameters marking different information structures. In perceptual studies, two perceptual experiments, (1) a dialogue history experiment, of which listeners identified the questions preceding the sentences, and (2) a pair comparison experiment, of which subject compared sentence with same wording but different information structures were used. The stimuli were either natural utterances form elicited spontaneous dialogues or resynthesized utterances with conflicting duration and f0 cues swapped from utterances of different information structures. Preliminary results of production experiments showed that duration elongation is a more consistent cue for marking narrow and contrastive foci than f0 range expansion. Listeners performed better in the pair comparison experiments than in the dialogue history experiment. For nature utterances, listeners were able to identify sentences with given information, given versus new information, and given versus contrastive information, but were not able to identify utterances with more than two contrastive foci. Listeners' performances declined when resynthesized utterances with conflicting duration and f0 cues were presented. When presented with conflicting cues, listeners relied more on durational than f0 cues.
Consonants, but not vowels, prime lexical decision following masked priming
Authors: New Boris , Araujo Veronica , Bour Nathalie , Nazzi Thierry
Published: May 2008

Recently, Nespor et al (2003) have argued that consonants play a more important role at the lexical level than vowels. This proposal has received direct support from studies showing that 16/20/30-month-old infants are better at processing specific consonantal than vocalic information while learning new words (Nazzi, 2005; Nazzi & New, 2007). This proposal is also supported, although indirectly, by studies on adults. Adults have been found to rely more on consonants than on vowels when asked to transform a nonword into a real word (Cutler et al., 2000). Moreover, adults presented with a continuous stream of syllables are able to segment it into "lexical units" by relying on consonantal but not vocalic information (Bonatti et al., 2005). In our study, we directly investigate this issue using a visual masked priming lexical decision task (an oral version being currently implemented). The test items are bisyllabic (CVCV or VCVC) and trisyllabic (CVCVCV or VCVCVC) words, which are presented following four different primes: identity (e.g., joli-joli), unrelated (vabu-joli), same consonants (jalu-joli) and same vowels (vobi-joli). Priming was found for the identity and same consonants conditions, not for the same vowels condition, supporting the privileged role of consonants at the lexical level.
Frequency importance function of Mandarin Chinese speech
Authors: Chen Jing , Qu Tianshu S. , Wu Xihong H. , Huang Qiang , Huang Ying , Li Liang , Chi Huisheng S.
Published: May 2008

The speech intelligibility index (SII) theory objectively assesses speech intelligibility, and the frequency-importance function (FIF), which reflects the relative importance of various frequency bands to speech intelligibility for various languages, occupies the central part of the theory. However, the FIF has not been examined for tonal Chinese Mandarin speech. In this study, considering the characteristics of Mandarin speech, 50 phonemically-balanced one-syllable words were selected as the speech stimuli from the speech corpus "A Method for Calculating Articulation Index (GB/T 15508-1995)", which meets the National Standards of China, and the 1/3 octave FIF was measured and computed with the Fletcher's method. The results show that spectrum regions with frequencies 2000-4000 Hz are more important in Chinese speech than those in English speech. The results predicted by SII model with the new FIF fit the result of human psychophysical studies well, which indicate that the new FIF model is effective and appropriate for assessing the intelligibility of Mandarin speech even when a steady-state noise masker is co-presented. [Supported by the NSFC 60435010; 60535030; 60605016; 30670704]
Effects of hand gesture and lip movements on auditory learning of second language speech sounds
Authors: Kelly Spencer , Hirata Yukari , Simester Jen , Burch Jackie , Cullings Emily , Demakakos Jason
Published: May 2008

Previous research found that auditory training helps native English speakers to perceive phonemic vowel length distinction in Japanese, but that their performance has never reached native levels (Hirata et al., 2007). Given that multimodal information, such as hand gesture and lip movements, influences semantic aspects of language processing and development (Kelly et al., 2002), we examined whether this multimodal information helps to improve native English speakers' ability to perceive Japanese vowel length distinction. Forty-five native English speakers participated in one of three types of training: (1) audio alone; (2) audio with hand gestures; and (3) audio with lip movements and hand gestures. Before and after training, participants were given phoneme perception tests that measured their ability to distinguish between short and long vowels in Japanese, e.g., /kato/ versus /kato:/. Our original prediction was that more modalities in training would result in greater learning. Although all three groups improved from pre- to post-test, there were no significant differences among the three training groups. Unlike the original prediction, hand gestures and lip movements did not seem to augment learning of difficult sound distinctions. We will discuss possible benefits and limitations of using multimodal information in second language speech learning.
The acoustics of yer and non-yer vowels [e] and [o] in Slovak
Authors: Benus Stefan
Published: May 2008

Yers of Slavic languages are vowels that alternate with zero and historically developed from high short lax vowels. In Slovak, both front and back yers were preserved and surface as [e] and [o] respectively. For example, the second [o] in kotol 'cauldron-Nom-Sg' is a yer because it disappears with adding a suffix vowel: kotla 'cauldron-Gen-Sg'. Compare with a non-yer [o] in kostol 'church-Nom-Sg', kostola 'church-Gen-Sg'. Traditional phonological accounts of this difference (e.g. Rubach 1993) assume that yer vowels are underlyingly different from non-yer vowels and merge with [e] and [o] through a phonological process. Therefore, these accounts predict that yer and non-yer vowels should be phonetically identical since they enter the phonetic component already merged as [e] or [o]. The results of our acoustic experiments show that yer vowels are phonetically different from non-yer vowels. The most salient differences were observed in the first formant and duration: yers have lower F1 than non-yers, and for some subjects they are also shorter. This finding supports the view that the phonetic component has access to deep phonological alternations and that phonetics-phonology is a single cognitive system in which the components have different granularities and interact bidirectionally.
Perception of sinewave vowels
Authors: Hillenbrand James M. , Clark Michael J.
Published: May 2008

There is a significant body of research examining the intelligibility of sinusoidal replicas of naturally spoken utterances. Discussion has followed about what the sinewave speech phenomenon might imply about the auditory and perceptual mechanisms that are involved in phonetic recognition. However, this work has typically been conducted using meaningful and syntactically well formed sentences, making it unclear what the relative contributions are of higher level, top-down processes as opposed to lower level auditory and phonetic mechanisms. The simple purpose of this study was to measure vowel intelligibility using sinusoidal replicas of naturally spoken /hVd/ syllables. The sinusoidal signals were modeled after 300 utterances selected from a database of 1,668 syllables spoken by men, women, and children [Hillenbrand, Getty, Clark, and Wheeler, J. Acoust. Soc. Am., 97, 3099-3111 (1995)]. Listeners consisted of 74 students enrolled in an introductory phonetics course. Recognition rates for the sinusoidal vowels averaged 55%, with considerable variability across listeners. This figure is substantially lower than the ̃95% intelligibility (measured in several earlier studies) of the naturally spoken signals upon which the sinusoidal replicas were modeled. Attempts to improve performance using three different training methods met with modest success, with post-training recognition rates rising by ∼5-12 percentage points, depending on the training method.
Letter sound and letter name recognition for automated literacy assessment of young children
Authors: Narayanan Shrikanth , Kazemzadeh A , Black Matthew , Tepperman Joseph , Lee Sungbok , Alwan Abeer
Published: May 2008

Evaluations of letter naming and letter sounding are commonly used to measure a young child's growing reading ability, since performance in them is well-correlated with future reading development. Assessing a child's oral reading skills requires teachers, as well as technologies that attempt to automate such assessment, to form an item-level accept/reject decision based on speech cues and prior knowledge of the child's literacy level and linguistic background. With data collected from 171 K-2 children, both learners and native speakers of American English, we designed and evaluated an automated letter naming assessment method using a simple word-loop HMM decoding for the word-level letter names. The automated accept/reject evaluation performance, 81.9%, approached the agreement of human raters, 83.2% (0.62 kappa). However, the task where children must produce the sound that the letter represents was more difficult: English orthography allows one-to many letter-to-sound mapping, teachers showed less agreement in their assessment (80.9%, 0.55 kappa), and the brief durations of some of the letter sounds made it difficult to distinguish them from each other and from background noises. Phone-level HMM based evaluation accuracy was 58.2%. Preprocessing the recordings into speech, silence, and noise improved these results, especially for plosive sounds. [Supported by NSF]
Production of American English [eI] and [ε] by Mandarin speakers: mono- vs. disyllabic words
Authors: Yang Chung-Lin
Published: May 2008

Mandarin productions of English tense vs. lax vowels are difficult for English speakers to differentiate (Chen et al. 2006). In this study the production of American English [eI] and [ε] by Mandarin and American participants was investigated. The target vowels were embedded in monosyllabic and disyllabic words with a stop-V-voiceless stop context in carrier sentences of variable length. There were two formant measurements for each vowel (20% after the onset and before the end of the vowel). In the monosyllabic condition, Mandarin speakers' performance was similar to American speakers in duration and formant values except that [eI] started too low. However, in the disyllabic condition, Mandarin speakers' formants for [eI] and [ε] were overlapped and there was no significant durational distinction between [eI] and [ε]. One possible account for this difference is that when producing a disyllabic word, Mandarin speakers tend to shorten the vowel duration of the first syllable to compensate for the production of the following syllable. The difference in syllable conditions may reflect a tendency that Chinese regularizes word durations regardless of the number of syllables. Reference: Chen, Y. et al. (2006). Vowel production by Mandarin speakers of English. Clinical linguistics and phonetics, 15, 427-440
A numerical analysis of fluctuations in pressure wave within the larynx using two-dimensional asymmetrical vocal folds model
Authors: Nomura Hideyuki , Kamakura Tomoo , Funada Tetsuo
Published: May 2008

Numerical simulations of pathological voice production and estimations of pressure wave fluctuations are performed based on a two-dimensional asymmetrical vocal folds (VFs) model. The asymmetrical VFs model takes into account of geometrical asymmetries (the thickness, effective depth of vibration region, and lateral rest position) and mechanical asymmetries (the Young's modulus, density, and viscosity of VF tissues). Simulation results based on the asymmetrical VFs model show that the left and right VFs vibrate with a phase difference. Obtained pressure waves within the larynx and vocal tract indicate fluctuations of fundamental frequency, amplitude, and waveform. In order to quantitatively evaluate the fluctuations, the coefficient of variation of the fundamental frequency, the coefficient of variation of the amplitude, and the harmonic-to-noise ratio are estimated. With increasing the VF asymmetries, especially on the effective depth and the density of VF elements, remarkable fluctuations are observed not only at the glottis but also in the regions downstream of the glottis and faraway from the glottis. This suggests that the estimation of fluctuation in pressure wave is useful for the diagnosis of pathological VFs.
Effects of training, style, and rate of speaking on speech perception of young people in reverberation
Authors: Hodoshima Nao , Arai Takayuki , Kurisu Kiyohiro
Published: May 2008

Because of the difficulty of listening to speech in reverberation (e.g. train stations or airports), we need to find characteristics of intelligible speech sounds that are appropriate for announcements over loudspeakers in public spaces. This study investigated the effects of training (seven talkers who have received speech training or not), style (conversational/clear) and rate (normal/slow) of speaking on speech perception of young people in simulated reverberant environments. The talkers were instructed to speak nonsense words embedded within a carrier sentence clearly or normally in an anechoic room, and listening tests were carried out with young people in simulated reverberant environments. Results showed that correct rates significantly differed among the talkers, but no difference in correct rates was found between the two speaking rates, and conversational speech had significantly higher correct rates than clear speech. Casual inspections of the stimuli indicate that vowels are enhanced as well as consonants in clear speech so that clear speech had lower correct rates than conversational speech due to increased reverberant masking in clear speech compared to conversational speech. [Work supported by Sophia University Open Research Center from MEXT.]
Acoustic-phonetic features for stop consonant place detection in clean and telephone speech
Authors: Lee Jung-Won , Choi Jeung-Yoon
Published: May 2008

This work classifies voiceless stop consonant place in CV tokens of English using burst release cues for clean (TIMIT) and telephone speech (NTIMIT). We compared the performance of cepstral coefficients to acoustic phonetics-motivated features such as center of gravity, burst amplitude and relative difference of formant amplitudes. In clean speech, cepstral coefficients resulted in better classification. However, for test data from NTIMIT, acoustic phonetic-based features outperformed cepstral coefficients, particularly if models were trained on clean speech. In addition, augmenting cepstral coefficients with acoustic phonetic-based measurements resulted in the best performance. These findings suggest that cepstral coefficients are able to model speech in a given environment in finer detail, whereas acoustic phonetic-based features are more robust to changes in environment, so that combining both types of measurements leads to the best performance.
Hierarchical approach to voice activity detection in realistic environmental noise
Authors: Bach Jörg-Hendrik , Anemüller Jörn
Published: May 2008

The automatic identification of acoustic objects, particularly speech, in adverse backgrounds can pose a difficult problem. In this work, the problem of speech detection in different backgrounds is tackled using a hierarchical decision tree: The root node provides a coarse classification of the noise background, while the next nodes are optimised to detect speech in one particular background each. The results of this approach are compared to the ITU G729B standard VAD and to a binary classifier trained on speech vs a mixture of the backgrounds. The features employed are amplitude modulation spectra of the signals (Kollmeier & Koch, 1993) that represent the modulation content of a signal as a two-dimensional (modulation-frequency vs centre frequency) spectral pattern. The best features for classification at each node are selected independently (node-specific) using a sequential floating forward search algorithm. The statistical learner is an open source support vector machine (Chang & Lin, 2001).
Neural correlates of speech and non-speech processing: Role of language experience in brain activation
Authors: Bosseler Alexis N. , Imada Toshiaki , Pihko Elina , Mäkelä Jyrki , Taulu Samu , Ahonen Antti , Kuhl Patricia K.
Published: May 2008

In adults, neural responses to the acoustic properties of native and non-native speech sounds differ. Recent magnetoencephalography (MEG) studies indicate that two regions of the brain, the superior temporal area and inferior parietal area (Broca's), in both the left and right hemispheres, may contribute to how the adult brain responds to the acoustic properties of speech. The aim of the present study was to investigate neural activation in adults when listening to native and non-native speech, and to non-speech analogs of speech. We used MEG to examine activation of the superior temporal and inferior parietal areas. Native speakers of Finnish were presented with Finnish (/pa/ and /ta/) and Chinese (/t "looped c" "superscript h" i/ and /"looped c" i/) syllables, and sine-wave replicas of /pa/ and /ta/, in a passive oddball paradigm. Our analysis examines the properties of localized neural activities to determine how activation of sensory and motor speech areas differs for speech and non-speech, and the role of language experience in brain activation.