bluuebunny/crossref_metadata_2025 huggingface dataset:
Discrimination of lexical tone contrasts in disyllabic nonsense pairs by adult speakers of Mandarin and of English
Authors: Berkowitz Shari , Strange Winifred
Published: September 2008

Most previous research on cross-language perception of lexical tone has used monosyllabic stimuli. In running speech, coarticulation affects the surface forms of the tones, creating contours that may be much more difficult to discriminate. Two tokens per type of disyllabic Mandarin nonsense words served as stimuli in a categorial same/different task. Pairings were chosen to be challenging, with some easy control pairs. Native Mandarin listeners (N=5) and American English listeners (N=15) with no background in tone languages completed 224 trials with feedback on 14 contrasts, then completed two additional blocks of 224 trials without feedback. Mandarin speakers were at ceiling on all contrasts; for Americans, accuracy did not improve across training and test blocks. Overall accuracy for different pairs averaged 78% (range 54–99%), and overall accuracy for (categorial) same pairs averaged 84% (range 75–97%). The mean A′ score for Americans was 0.88 (range 0.75-0.96). Performance across the 14 contrasts varied from A′=0.74 to 0.96, and showed significant context effects. Error analysis showed a pattern of errors across tone pairs that supported the hypothesis that American speakers relied on the overall pitch contour across the two syllables, rather than a syllable-by-syllable analysis.
Robust and accurate F0 estimation for reverberant speech by utilizing complex cepstrum analysis
Authors: Unoki Masashi , Hosorogiya Toshihiro
Published: September 2008

This paper presents comparative evaluations of 12 typical methods of estimating the fundamental frequency (F0) over huge speech-sound datasets in reverberant environments. They involve several classical algorithms such as cepstrum, AMDF, LPC, and autocorrelation methods. Other methods involve a few modern algorithms, i.e., instantaneous amplitude and/or frequency-based algorithms, such as TEMPO, IFHC, and PHIA. The comparative results revealed that the percentage of correct rates and SNRs of the estimated F0s were reduced drastically as reverberation time increased. This paper, thus, proposes a method of robustly and accurately estimating F0 in reverberant environments by utilizing the MTF concept and the source-filter model in complex cepstrum analysis. The MTF concept is used in this method to eliminate dominant reverberant characteristics from observed reverberant speech. The source-filter model is used to extract source information from the processed cepstrum. Finally, F0s are estimated from them by using the comb-filtering method. Additive-comparative evaluation was carried out on the proposed method and other typical methods. The results demonstrated that it was better than the previously reported methods in terms of robustness and in providing accurate F0 estimates in reverberant environments. [Work supported by a Grant-in-Aid for Science Research from the Japanese Ministry of Education No. 18680017.]
Bistatic target detection using a time-reversal operator
Authors: Edelmann Geoffrey F. , Fromm David M. , Gaumond Charles F.
Published: September 2008

Previous work has shown that time-reversal operator (TRO) methods can detect ocean targets using a time-reversal mirror [C. Gaumond, J. Acoust. Soc. Am 119, 976–990 (2006)]. To increase the signal-to-noise ratio, the TRO was measured by transmitting a set of orthogonal beams. The method is extended to a bistatic case where the receiver and transmitter are no longer collocated. Simulated deep ocean convergence zone and up-slope detection at 200 Hz will be discussed. The consequences of array geometry (horizontal and vertical) to beam orthogonality will be addressed. Specifically, under what conditions will a horizontal-receive array have enough aperture to measure a TRO of sufficient rank? It will be shown that the system must be able to both transmit and receive beams that are orthogonal with respect to the waveguide (and not simple free space arguments). [Work supported by the Office of Naval Research.]
Effect on surface ship interference on adaptive beamforming for the shallow water array performance (SWAP) project
Authors: Campbell Richard L. , Zurk Lisa M.
Published: September 2008

The shallow water array performance (SWAP) project is designed to explore the limits of large-aperture passive sonar array processing capability in a shallow-water environment with moving surface ship interference. The array, off the eastern coast of Florida near Ft. Lauderdale, has 500 elements with a total length of approximately 900 m. One challenge for the processing is the length of observation time needed by traditional adaptive beamforming formulations—with an array of this size and element quantity, the resolution of range and bearing cells is such that a ship may move across many cells during the snapshot time, spreading the resulting eigenvector structure and decreasing effective signal gain. A central question is the trade-off between array gain and this eigenvector spreading loss. To explore this question, tracks from actual ships in the vicinity of the array site, combined with sound speed and bathymetry data from the site, are used in an adiabatic normal mode simulation to predict the acoustic response across the array. The resulting simulated snapshots are used in adaptive and non-adaptive formulations to predict target detection performance as a function of the interference environment and processing parameters, for both full and sub-aperture processing schemes.
Target scattering waves emitted from a time reversal array in shallow water
Authors: Tsurugaya Yoshiaki , Kikuchi Toshiaki , Mizutani Koich
Published: September 2008

When a target exists between a sound source and a TRA in shallow water, the TRA receives the sound waves from the sound source and the waves scattering by the target. If the time reversal processing to them is carried out and they are re-emitted from the TRA, it will be thought that they are converged on the position of the sound source and the target. However, since the waves converging on the sound source have a high level, the waves converging on the target position are usually masked by the high level sounds. We cannot observe the waves converged on the target. We then eliminate only the high level sounds from the sound fields. In each array element, the signals in the case of nontarget are subtracted from the signals including the target. As a result of subtraction, the components of the scattering wave by the target are left on the array elements. The time reversal fields of the scattering wave are constructed by radiating the components of the scattered wave from each element again.
Identification of a resonant target buried in sediment using iterative, single channel, time reversal
Authors: Waters Zachary J. , Holt R. Glynn , Roy Ronald A. , Dzikowicz Benjamin R.
Published: September 2008

The presence of noise and clutter makes identification of targets buried beneath the seafloor a challenging problem. Iterative time reversal using a single-channel transducer is shown to enhance the signal-to-noise ratio of backscattered echo returns from a buried resonant target. Each iteration consists of: (1) Insonifying the target with a broadband pulse, (2) windowing a portion of the backscattered echo return, (3) reversing it in time, and (4) using this waveform as the source signal for the subsequent interrogation. Scaled laboratory experiments are performed with a broadband (rmQ∼2) transducer, operating between 500 kHz and 2 MHz, and a 6.35 mm diam hollow aluminum spherical shell target buried beneath a layer of simulated sediment. When the target is located within the time reversal window, there is a rapid convergence to a narrowband signal characteristic of a dominant mode of the target’s scattering response. Images are generated by scanning the transducer laterally in two dimensions above the buried target. The images reveal enhancement of different resonant modes of the target, depending on the transducer’s position. Results will also be reported from larger-scale experiments performed in a test pond at lower frequencies. [Work supported by Office of Naval Research Award No. N000140610044.]
Acoustics of university rehearsal spaces: Not too live, not too dead
Authors: Cooper Russell
Published: September 2008

The design of music rehearsal spaces for university schools of music must balance many factors. Most important of these is loudness and decay time. Rooms must have sufficient floor area and volume so that the rooms are not overly loud for the teachers and students. The size and volume of the room depend on the type of ensemble rehearsing in the room (marching band, concert band, wind ensemble, choral, orchestra, percussion, jazz/rock, etc.). Decay times are often desired to match that of the performance stage. We have found this not to be ideal, that in fact rooms should be less reverberant than the stage platform, to provide students and teachers the ability to hear their instrument’s articulation and syncopation. A certain amount of adjustable acoustics are necessary to account for multi-use of the room for different ensembles, and finishes should be selected for the proper diffusion of sound and to achieve the desired decay time. This paper will discuss many of our firm’s designs and some of the successful common design themes we use.
School music rehearsal spaces—How dry is dry enough?
Authors: Brooks Bennett M.
Published: September 2008

Handbook values for optimum reverberation time (RT60) in a music studio, such as a school band or chorus rehearsal space, fall into the 0.8 to 1.0 second range, depending on volume. One music room furniture manufacturer specifies RT60 values as high as 1.91 seconds. We submit that these RT60 values are excessive. The Classroom Acoustical Standard (ANSI S12.60-2002) specifies a maximum allowable RT60 of 0.7 seconds to achieve good speech intelligibility. Obviously, good speech communication is necessary for effective education of any kind. In special function rooms designed for music instruction, experience has shown that the design target RT60 should be as low as 0.2 seconds, for frequencies of 500 Hz and above. The requirement for such low reverberation is dictated by the need to reduce the extremely high sound levels that can be generated by a large group; and to give the instructor a dry environment that provides the ability to isolate, identify, and evaluate the musical performance of an individual student while rehearsing in a larger group. This allows the instructor greater opportunity to reinforce good student musicianship and to correct individual student mistakes, ultimately resulting in a higher level of concert performance. Sonic examples are given.
Improvement of reverberation and sound isolation in a high school music rehearsal facility
Authors: Abercrombie Clemeth L.
Published: September 2008

Music rehearsal facilities are used by some of the most acoustically demanding individuals; extremely discerning even at an early age in their musical path. This paper examines acoustical issues in a high school marching and concert band rehearsal suite that is home to approximately 70 students. Even though the two-year-old building met acoustical design goals provided by an acoustical consultant, the music program director using the facility was concerned enough with acoustical performance that he successfully lobbied the school district for a renovation. Outstanding issues included clarity and low-frequency response in the main rehearsal room, sound isolation and loudness in individual practice rooms directly adjacent to the main room, and noise experienced in the nearby choir rehearsal suite. This presentation will review the importance of acoustical parameters specific to rehearsal, as well as the implementation of custom designed low-frequency absorption and cost-effective sound-isolation improvements.
Concert hall canopy height comparison and computer model resolution efficacy
Authors: Ermann Michael , Na Ayudhya Indhava Kunjara
Published: September 2008

This line of inquiry is twofold: (1) It uses computer modeling to compare the height of an adjustable concert hall ceiling canopy with established room acoustics metrics, and (2) It does so simultaneously with two computer models, one at a rough resolution, and one at a resolved resolution. The height comparison looks to establish best practices in operation of such a canopy in a hall; how high should it be set? The model resolution comparison looks to establish best practices in room acoustics computer modeling; how detailed must the model be?
Modulated ultrasound and multifrequency radiation force
Authors: Urban Matthew W. , Fatemi Mostafa , Greenleaf James F.
Published: September 2008

Modulating ultrasound can create dynamic ultrasound radiation force to induce local tissue vibration. Using different types of modulating signals produces radiation force with weighted multifrequency components. Modulation signals with frequency fr produce force with components at multiples of fr. Both amplitude modulation (AM) and double sideband suppressed carrier (DSB-SC) amplitude modulation were explored. Different waveforms were investigated such as sine, square, triangle, and sawtooth for modulating continuous wave ultrasound. A 3.0 MHz transducer produced ultrasound which was measured with a needle hydrophone and used to derive and analyze the spectrum of the radiation force function. The modulated ultrasound was used to vibrate a steel sphere in a gelatin phantom, and the motion was measured using a laser vibrometer. The measured spectra were used to find the frequency response of the sphere. Different modulating signals were used to produce multifrequency radiation force. The sphere frequency response was recovered using different modulating signals. Weighted multifrequency radiation force was produced by modulating ultrasound with different signals. Application of multifrequency radiation force can be used to measure the frequency response of objects or tissue. [This work was supported in part by grants EB002167, EB002640, and CA091956 from NIH.]
Estimation of tissue stiffness with the surface wave generated by ultrasound
Authors: Zhang Xiaoming , Urban Matthew W. , Kinnick Randall R. , Greenleaf James F.
Published: September 2008

The mechanical response of tissues to external forces has gained considerable interest in medical diagnosis. Typically, a force is needed to produce displacement in the tissue for estimation of the tissue’s elasticity. One approach is to apply an external force on the body’s surface. Another emerging technique is to generate a localized force inside the tissue with ultrasound. Shear waves can be used in bulky tissues for estimating the tissue’s elasticity. The bending wave can be generated and measured in vessels for noninvasive estimation of a vessel’s elasticity. We have found that the Rayleigh surface wave can be generated on the tissue’s surface, [X. Zhang et al., Proc. International Congress on Ultrasonics, Vienna, April 9–13, 2007], [X. Zhang et al., IEEE Trans. Medical Imaging, 26, 843–852 (2007)]. In our method, a localized radiation force of ultrasound is remotely and noninvasively applied inside the tissue. This force can generate the shear wave as well as compression wave inside the tissue. However, only the surface wave can exist on the body’ surface. The surface wave can be used to estimate and image the stiffness of tissue. Some results of using the surface wave are presented.
Speech-in-noise perception and recognition
Authors: Bradlow Ann R. , Espy-Wilson Carol
Published: September 2008

Recent speech research has established that, for humans, speech-in-noise and speech-in-quiet perception differ along numerous dimensions that span levels of signal encoding and linguistic representation. Listeners place more or less weight on specific acoustic cues, draw more or less on signal-independent, contextual information, and are more or less distracted by lexical neighbors depending on masker type and level. Moreover, noise has different effects at different levels for different listener populations. Machine recognition of noisy speech is well below (at least by an order of magnitude) human recognition of noisy speech. This difference is true for both additive noise and convolutive noise. Researchers have focused on the problem of machine recognition of noisy speech in several ways: Developing robust features, training systems on noisy speech, or developing speech enhancement algorithms to clean up the noisy speech signal before performing recognition. A remaining challenge for understanding how humans do and how computers should handle speech-in-noise is to develop a conceptual framework that goes beyond the division of masking effects into peripheral, energetic versus central, informational.
Speech privacy in healthcare facilities
Authors: Tocci Gregory C. , Sykes David
Published: September 2008

In response to the growing recognition in the U.S. that personal privacy requires more protection, two federal regulations were enacted. These are the Health Insurance Portability and Accountability Act of 1996 (HIPAA) and the Gramm-Leach-Bliley Act of 1999 (GLBA). The first includes provisions for protecting patient healthcare information and the second for protecting personal financial records information. These focus on data privacy and only obliquely refer to speech privacy; however, the enforcement guidelines for HIPAA specifically require acceptable speech privacy consistent with the intention of HIPAA. The definition of speech privacy, how to evaluate it, and what constitutes acceptable speech privacy are not addressed in these Acts nor in enforcement guidelines. To provide much needed guidance on this, ANSI S12 Working Group 44 Speech Privacy was formed in 2006 to develop standards for speech privacy. The efforts of the working group are to be discussed. In networking with the healthcare building design profession, it became evident that the American Institute of Architects (AIA) Hospital Design Manual was in great need of general acoustical, as well as speech privacy guidelines. This presentation will discuss this and the many on-going efforts in the area of healthcare acoustics.
Noise diagnosis using nearfield acoustical holography
Authors: Wu Sean
Published: September 2008

This presentation focuses on diagnosis of air- and structure-borne sounds by using nearfield acoustical holography (NAH) with an emphasis on applications of Helmholtz equation least-squares method for various industrial projects. The presentation is structured with strong consideration towards the end result, which is the visualization and quantification of the acoustic field and ranking of relative contributions from individual noise sources. It is emphasized that, in general, this approach does not deal with actual noise control but rather with identifying the areas where such effort may yield the most cost-effective results. Comparison of NAH-based diagnosis with other techniques such as transfer path analysis and an intensity probe is presented, and their relative advantages and disadvantages are discussed. Specific examples of identifications of sound transmission paths into a passenger vehicle compartment, fuselage of an aircraft, and disk brake squeals are presented. The reconstructed acoustic quantities, such as the acoustic pressures and normal surface velocities, are compared with the benchmark results.
Effect of level of presentation on scaled speech intelligibility of speakers with dysarthria
Authors: Kim Yunjung , Weismer Gary , Kent Raymond D. , Duffy Joseph R.
Published: September 2008

A considerable amount of effort has been made to seek the acoustic correlates of reduced speech intelligibility in dysarthria, although some acoustic variables are under debate: For example, articulation rate, F2 slope, and vowel space have been frequently discussed in previous studies as acoustic predictors of speech intelligibility of dysarthria [Kim (2007)]. One variable that is not well understood, with respect to its effect on speech intelligibility in dysarthria, is level of presentation. This includes level generated by a speaker as well as level of presentation after the speaker has recorded utterances. Clearly, variations in level of presentation of recorded utterances will have some effect when an intelligibility test requires some segmental analysis (as in a minimal pairs test), but the potential effect when speech intelligibility is scaled is not so clear. This is an important question because of the frequent use of scaled intelligibility in dysarthria research. In this presentation, we will report data on how scaled speech intelligibility is affected by level of presentation of speech samples produced by speakers with various types and severity of dysarthria. [Work supported by NIH DC00319.]
Effects of clear speech on duration and fundamental frequency of vowels produced by monolingual and bilingual talkers
Authors: Rogers Catherine L. , Bianchi Michelle , Frisch Stefan A. , Krause Jean C.
Published: September 2008

Prosodic characteristics of vowels produced by monolingual and bilingual talkers were investigated. Ten monolingual, 15 early Spanish-English bilingual (age of onset of immersion of age 12 or earlier), and ten late Spanish-English bilingual (age of onset of immersion of age 15 or later) talkers produced the target words ‘‘bead, bid, bayed, bed, bad,’’ and ‘‘bod’’ in conversational and clear speech styles. Vowel duration was computed, and F0 measurements were made at 20%, 50%, and 80% of the vowel duration. A significant group by style by vowel interaction showed that monolingual and early bilingual talkers enhanced inherent duration differences between target vowels by lengthening long vowels significantly more than short vowels in clear speech. The vowels of the late bilingual talkers, by contrast, became more alike in duration in clear than in conversational speech. The monolingual talkers showed a falling F0 pattern from 20% to 80% of the vowel duration in both styles; the late bilingual talkers showed a flat or rising F0 pattern in both styles; and the early bilingual talkers showed a flat or rising pattern in conversational speech, but a falling pattern in clear speech. [Work supported by NIH-NIDCD ♯5R03DC005561.]
Tonal coarticulation in Yoruba: Locus equation analysis
Authors: Agwuele Augustine
Published: September 2008

This study examines the acoustic coarticulatory effects of tone on VCV sequences of Yoruba, when each vowel of the VCV sequences bears a different tone. The first aim is to understand how the presence of tone affects CV coarticulation in the traditional locus equations parlance as used by Krull (1987). This marks the first application of the locus equation metric to a tonal language in the study of CV coarticulation. The second aim is to determine how the stop consonant in VCV sequences is affected by prosodic overlay, independently of the vowels altered location in articulatory/acoustic space. In order to achieve the second objective, the modified locus equation regression metric that was used by [Lindblom et al. J. Acoust. Soc. Am. (2007)] to dissociate vowel context effects from rate-induced effects on consonantal F2 onsets are applied. Similar to the findings of Lindblom et al. (2007), the analyses document separate effect for F2 locus relative to F2 nucleus.
Speeded discrimination of American vowels by experienced Russian learners of English
Authors: Gilichinskaya Yana D. , Law Franzo , Strange Winifred
Published: September 2008

This study is part of a project examining how L2 learners whose native languages have small vowel inventories (Japanese, Russian, Spanish) perceive American English (AE) vowels. Experienced Russian (RU) L2 learners’ discrimination was evaluated in a speeded ABX task that included nil Experimental contrasts among adjacent height pairs and front/back pairs; four nonadjacent height pairs served as Controls. An earlier study of Japanese L2 learners indicated that AE contrasts with spectrum and duration differences (S+D) [i:/■, æ:/[g\/], u:/■, ■:/■] were discriminated more rapidly, relative to Control contrasts [i:/[g\/], ■/æ, u:/■, ■/■:], than were contrasts that differed only in spectral structure (S-Only) [■/[g\/], ■/■, [g\/]/■, æ:/■:]. Since vowel duration is not contrastive in Russian, it was hypothesized that discrimination of S-Only and S+D contrasts might be equally slow, relative to Controls, if RU listeners did not attend to duration differences. Preliminary findings confirm that reaction time (RT) difference scores (relative to Controls) were not different for S-Only and S+D contrasts. Relative RTs on both contrast types were slower than for AE listeners, whose RTs were about the same for S-Only and S+D pairs. These findings suggest that speeded discrimination is a sensitive measure of continuing perceptual difficulties of L2 learners. [Work supported by NSF.]
Perceptual similarity of American English and Japanese vowels for native speakers of American English and Japanese
Authors: Nozawa Takeshi , Frieda Elaina M.
Published: September 2008

Native speakers of American English and Japanese chose American English vowels that best represented Japanese vowels uttered in two different consonantal contexts. The two groups of subjects’ responses differed noticeably, showing the effects of linguistic experience. The Japanese subjects tended to match Japanese long two-mora vowels with long, tense vowels and short one-mora vowels with short lax vowels, but the English subjects were relatively unaffected by durational differences. The Japanese subjects matched /a/ and /e/ to /æ/ and /[g\/]/, respectively, but the English subjects matched /a/ with /■/ or /■/, and /e/ with /■/ as well as /[g\/]/. The Japanese subjects matched /o/ with /■/, while the American subjects matched it with /o■/. The Japanese subjects took part in two other experiments. They equated American English vowels to Japanese vowel categories and they also identified American English vowels. They equated English /■/ to Japanese /a/, and /■/ was often misidentified as /æ/. The results of the three experiments show that there are discrepancies between how an American English vowel is mapped into Japanese vowel categories and how the Japanese subjects expect it to sound, which may result in the poor identification of American English vowels.
Vowel intrinsic structure involved in continuous speech in articulatory space
Authors: Lu Xugang , Dang Jianwu , Fujita Satoru
Published: September 2008

The relation of the speech production and perception was described using the similarity of acoustic (F1-F2) space and articulatory (vowel-triangle) space for isolated vowels. However, such a relation has not been found in continuous speech. This study attempts to find out the intrinsic structure of vowels involved in continuous speech by focusing on the whole vowel structure involved in the articulation, by keeping the neighboring topological relationship of the data set. A nonlinear dimension reduction method, Laplacian eigenmap, is adopted to realize this mission. Japanese vowels were extracted from an articulatory data set that was obtained using the electromagnetic articulography system for continuous utterances. When the articulatory data were compressed from 14 dimensions to three dimensions, a nonhomogeneous structure emerged from the vowels. The vowels were distinctively categorized based on their similarity relationship revealed by their distribution on the structure along the three dimensions. For the three dimensions, the first dimension corresponds to the height of the tongue, and the second one relates to the lip rounding. The third dimension is concerned with the articulatory place along the vocal tract. The same method was also used to extract the intrinsic structure of the corresponding acoustic data set.
Acoustic articulatory evidence for quantal vowel categories: The feature [low]
Authors: Jung Youngsook , Stevens Kenneth N.
Published: September 2008

The goal of this study is to determine whether acoustic coupling between the first subglottal resonance F1sub (about 600 Hz) and the F1 frequency for vowels creates a region near 600 Hz in which the F1 prominence shows an irregularity. Such a finding would provide evidence for a defining quantal articulatory-acoustic relation for the distinctive feature [low]. The time course of F1 in relation to F1sub was examined for certain diphthongs and several monophthongs produced by a number of speakers of English using Chi’s data [X. Chi and M. Sonderegger, J. Acoust. Soc. Am. 115, 2540–2550 (2004)]. For the diphthongs, a discontinuity in F1 or a dip in amplitude of the F1 prominence was observed as it passed through F1sub, while for the monophthongs, F1 was usually above F1sub for [+low] vowels and below F1sub for [-low] vowels. A preliminary further study of data from the literature on F1 for vowels from various languages showed that the boundary between F1 values of [+low] vowels and those of [-low] vowels agrees with the average value of F1sub obtained from the laboratory study with English. [Supported by NIH Grant No. DC00075.]
Incorporating vocalic segment memories in automatic dialect identification
Authors: Rojas David M.
Published: September 2008

Systematic pronunciation differences among speakers of regional varieties of U.S. English are recognizable to other native speakers to varying degrees. This has often been demonstrated through experiments wherein listeners were asked to match a talker to his or her dialect region. Machines have also been able to identify the regional origin of a speaker to some degree, although attempts to this end have typically not been as successful as efforts to automatically identify the language of a speaker. In order to refine the dialect discrimination ability of a machine, this paper draws methodological inspiration from the area of musical artist classification, and from linguistic notions that vowels contribute more heavily than consonants to regional differences. Using this insight, an automatic dialect identification system is developed that first recognizes the more vowel-like slices of the signal, and then updates a vocalic segment memory component with Mel-frequency cepstral coefficient, formant, and pitch information from the current frame. Besides providing a means to analyze MFCC and formant trajectories, the segment memory enriches the representation of vocalic events by allowing the system to explicitly model prosodic aspects such as duration and tilt.
The application of a psychophysical difference metric to perceptual similarity judgments in vowels
Authors: Harnsberger James , Shrivastav Rahul , Skowronski Mark
Published: September 2008

Models of cross-language speech perception have had limited success in predicting the discriminability or perceptual similarity of non-native contrasts. These failures may be attributed partly to an inability to quantify the phonetic differences between non-native speech sounds. This study attempted to quantify such gross psychophysical differences between speech sounds, specifically by utilizing dynamic time warping (DTW) on human factor cepstral coefficients to compare the spectrum of the entire length of the speech sounds in question. This technique has been successfully applied to account for the discriminability of different non-native consonant contrasts [Harnsberger, J. D., Shrivastav, R., and Skowronski, M.; J. Acous. Soc. Am. 117, 2460, 2005]. This study extends this work to perceptual similarity judgments of vowels. Specifically, twenty native speakers of English were presented with all possible pairings of ten vowels produced by two speakers of English. Subjects were asked to rate their similarity on a seven point scale. The resulting similarity scores were then compared with the output matrix of the DTW psychophysical difference metric for the same stimulus materials. The results showed a significant correlation (r=.60**) between the two measures, demonstrating the efficacy of the metric with a greater range of stimulus types and tasks.
Robust unsupervised extraction of vocal tract variables from midsagittal real-time magnetic resonance image sequences using region segmentation
Authors: Bresch Erik , Narayanan Shrikanth
Published: September 2008

The tracking of deformable objects in image sequences has been a topic of intensive research for many years, and many application-specific solutions have been proposed. In this work, we describe a method that was developed to robustly track the tissue structures of the human vocal tract in midsagittal, real-time magnetic resonance (MR) images. The goal of the algorithm is to fully automatically extract the vocal tract outline, the position of the articulators, and the tract variables to facilitate the study of the shaping of the vocal tract during speech production. The algorithm is unsupervised and requires only a one-time initialization step for a particular subject. Importantly, the tracking algorithm operates on the spatial frequency domain representation of the underlying images, and it is hence specifically fit to the data produced in the MR imaging process. The proposed method carries out a multiregion segmentation of the individual MR images using an anatomically informed model of the vocal tract whose fit to the observed image data is hierarchically optimized using an anatomically informed gradient descent procedure. The mathematical key components of the algorithm are the closed-form solution of the two-dimensional Fourier transform of a polygonal shape function and the design of alternative gradient descent flows for the iterative solution of an overdetermined nonlinear least squares optimization problem. Various examples of segmented real-time MR images and a summary of open challenges will be presented. [Work supported by NIH.]
Volumetric MRI acquisition and processing
Authors: Carbone Inês , Martins Paula , Silva Augusto , Teixeira António
Published: September 2008

To provide fast and potentially more accurate three-dimensional anatomic information on European Portuguese (EP) sounds, a direct acquisition of volumetric MRI information was attempted, through a 3-D spoiled fast gradient echo sequence. The viability of the used MRI acquisition protocol relies on the possibility of obtaining consistent information. For that, two different image processing approaches were attempted: First, a so-called 2.5-D method consisting in a first segmentation on the sagittal plane, multiplanar reslicing perpendicularly to the sound propagation in the vocal tract, and new segmentation in the 45 planes obtained; second, a direct 3-D level set segmentation. A method was applied to a comprehensive single speaker corpus contemplating: Oral and nasal vowels, nasal consonants, unvoiced fricatives, and laterals. Method advantages are: Reconstruction of slices with good detail in any direction, useful to obtain slices perfectly orthogonal to the vocal tract centerline; higher SNR when compared with 2-D imaging; easier 3-D visualization; and acquisition two times faster, with positive impact in the amount of material acquired in one session with less subject effort. 3-D information was especially important for the study of nasal [P. Martins et al., InterSpeech 2007, accepted] and lateral sounds, some of them with specific characteristics for EP. [Work supported by FCT, Portuguese Research Agency, by Project HERON POSC/PLP/57680/2004.]
Anatomic development of the vocal tract during the first two decades of life: Evidence on prepubertal sexual dimorphism from MRI and CT studies
Authors: Vorperian Houri K. , Schimek E. Michael , Wang Shubing , Chung Moo K. , Kent Ray D. , Ziegert Andrew J. , Gentry Lindell R.
Published: September 2008

The growth of the vocal tract (VT) is known to be nonuniform insofar as there are regional variations in anatomic maturation. This study presents quantitative anatomic data on the developing VT from 604 imaging studies (277 male; 327 female) between birth and 20 years. Data analyses include detailed assessment on the growth of the oral (anterior or horizontal) and pharyngeal (posterior or vertical) regions of the VT for both sexes. The oral region of the VT was segmented into lip-thickness, anterior-cavity-length, and orophayrngeal-width; and the pharyngeal region of the VT into posterior-cavity-length, and nasopharyngeal-length. Findings from all variables/segments indicate differences in growth trend, rate, and type (somatic versus neural) between males and females. However, it appears that prepubertal sex differences at specific age ranges are masked by overall growth rate differences between males and females. Comparing males versus females, using a limited age range of 60 months, unveiled prepubertal sexual dimorphism in the oral/horizontal region where differences appear more pronounced in the oropharyngeal-width segment. Such novel anatomic findings indicate one possible source for noted sex differences in formant frequencies before age 10, where there is no sexual dimorphism in VT length. [Work supported by grants NIH-NIDCD R03-DC4362, R01-DC006282, and NIH-NICHHDP30-HK03352.]
Production facilitates lexical acquisition in young children
Authors: Zamuner Tania , Fikkert Paula , Gick Bryan
Published: September 2008

Recent research in language acquisition argues that production plays a unique role in the acquisition of phonological and lexical representations [Fikkert. In press, LabPhonX]. In this research, we investigate the effect of production in early lexical acquisition. Twenty Dutch-learning children participated in a word-learning task, where children were taught and asked to imitate non-words. Children were then tested on their recall of the non-words, and their receptive knowledge of the non-words. Children’s performance on the two tasks was compared to whether they imitated the non-words during training, the number of times they imitated the non-words, and the accuracy of their imitations. While no effect of production during training was found on children’s receptive knowledge of the non-words, production during training was associated with children’s ability to recall the non-words. Specifically, children who produced the non-words during training were more likely to recall the non-words at test. No relationship was found between the number of times children produced the non-words during training, or the accuracy of their imitations. These results indicate that production enhances early lexical acquisition, and suggests a unique role for articulation in the formation of phonological and lexical representations.
How acoustic cues in infant-directed speech facilitate 19-month-olds’ word recognition: Evidence from time course analysis
Authors: Song Jae Yung
Published: September 2008

This study investigated how the acoustic characteristics of infant-directed speech (IDS) facilitate infants’ recognition of familiar words. The individual roles of three typical characteristics of IDS were examined: Exaggerated pitch range, slow speech rate, and hyperarticulated vowels. Using the intermodal preferential looking procedure, 34 19-month-olds were presented with 12 stimulus sentences (Where is the [target]?); six were spoken in typical IDS style and the other six were digitally altered to remove one of the three IDS characteristics. Infants’ looks to target and distracter were coded frame by frame to examine the time course of word recognition. Results showed that infants’ ability to recognize words was affected by speech rate. Total looking time to the target was greater (t(11)=3.337, p=0.007), latency of the first look was shorter (t(71) =−1.919, p=0.059), and the accuracy score was higher overall when listening to typical IDS as compared to accelerated IDS. In contrast, infants showed no difference in performance when pitch range or the degree of hyperarticulation was modified. This suggests that slow speech rate enhances infants’ ability to recognize words more efficiently and reliably during the stage of rapid vocabulary development. [Work supported by NSF BCS-0544127, NIH R01MH60922.]
Investigation of the effect of low frequency components of sound to speech intelligibility
Authors: Harun Mokhtar , Dimon Mohamad Ngasri , Abdul Hamid Siti Zaleha , Nasoha Hamim
Published: September 2008

Sound attenuation depends on source-to-listener distance, the finishes, and the shape of the room. The amount of sound attenuation at different frequencies influences the overall speech intelligibility in a room. The sound at high frequencies, especially at 1 kHz and 2 kHz, contribute the most for the articulation of consonants, and thus affects speech intelligibility. However, it has been found previously that sound attenuation is lower and harder to achieve at lower frequencies. This would lead us to assume that reverberant sound is mostly of low frequency components. Even if the contribution of sound at low frequency is low for speech intelligibility, this paper attempts to gauge the effect of sound pollution at low frequencies to the overall speech intelligibility in the room. With fixed source-to-listener distances, the measurements were conducted in classrooms with different sizes and finishes.
Children’s perception of American English consonants in noise
Authors: Nishi Kanae , Lewis Dawna E. , Hoover Brenda M. , Choi Sangsook , Stelmachowicz Patricia G.
Published: September 2008

Following the seminal study [Miller and Nicely, J. Acoust. Soc. Am. 27(2), 338–352 (1955)], numerous studies have explored consonant confusion patterns for adult listeners. Although children are known to be less skilled than adults in many aspects of auditory abilities, little is known about how these immature abilities affect consonant perception. In the current study, 72 children (4-, 5-, 6-, 7-, 8-, and 9-year-olds) and 12 adults identified 15 American English consonants in /aCa/ nonsense syllables presented in speech-shaped noise at 0, +5, and +10 dB signal-to-noise ratios (SNRs). Overall percent-correct identification of 15 target consonants improved with age and SNR, but no SNR age interaction was observed. Further analysis of confusion patterns using indexes for transmitted information (TI) for linguistic features (manner, place, and voicing) indicated the following: (1) TI for all features increased with improvement in SNR for all age groups; (2) TI for voicing did not vary with age; (3) TI for place was significantly lower for younger children than adults. These results will be discussed in relation to the development of auditory skills. [Work supported by NIH DC004300.]
Voicing and devoicing in similar German and English word pairs by native speakers of German
Authors: Smith Bruce L. , Bruss Michael , Hayes-Harb Rachel , Hamilton Amy
Published: September 2008

A number of languages manifest a pattern of word-final obstruent devoicing, i.e., words ending with underlying voiced obstruents are pronounced so as to be (largely) indistinguishable from words ending with final voiceless obstruents. One question regarding this tendency to devoice voiced obstruent targets is what occurs when native speakers of such a language learn a second language that has a word-final obstruent voicing contrast. For example, do German speakers neutralize voicing contrasts in English as they tend to do in German, or can they learn to produce voiced obstruents in Englis, despite devoicing them in German? To examine this issue, ten native speakers of German produced various phonologically-similar minimal pairs in German and English (e.g., English: Lied/light; German: Leid/leit). Acoustic measurements were made of their productions in both languages, viz., vowel duration preceding final consonants, final consonant duration, duration of voicing during final consonants, and final release burst duration; their English productions were also compared to those of native English speakers. The native German speakers tended to neutralize the voicing distinction to a greater extent when producing German words versus phonologically-similar English words, but they typically did not produce as much of a contrast in English as native English speakers.
At-sea measurements and modeling of acoustic scattering from solid-filled elastic targets on the sea-bed
Authors: Zampolli Mario , Tesei Alessandra , Canepa Gaetano
Published: September 2008

Monostatic and bistatic measurements of acoustic scattering from solid-filled fiberglass objects (a sphere and a cylinder with hemispherical endcaps) deployed proud on a sandy seabed and insonified by a rail-mounted parametric source at low frequency (roughly ka=540), are presented. The targets are thin-walled, approximately isotropic random-fiberglass shells, filled partially with an isotropic epoxy resin and partially with sea water. The targets are simple enough in shape to be treated by currently available modeling techniques, but realistic enough to give a first insight into the physics of the elastic waves supported by such material combinations. Preliminary data analysis indicates that the scattering signatures are dominated by the solid filling. The experimentally measured target responses are compared to numerical simulations obtained with a finite-element tool based on the decomposition of the field variables in a series of azimuthal Fourier modes. The experimental data were acquired in October 2006 during the EVA-06 trial off the Island of Elba, Italy.
Tank measurements and modeling of elastic scattering by resin-filled fiberglass spherical shells
Authors: Tesei Alessandra , Zampolli Mario , Guerrini Piero
Published: September 2008

Acoustic elastic scattering measurements were conducted in a tank on 6 cm-radius fiberglass spherical shells filled partially or completely with a low-shear-speed epoxy resin. Preliminary measurements were conducted also on the void shell before filling and on a solid sphere of the same material of the filler, in order to estimate the constituent material parameters via acoustic inversion. The objects were measured in the backscatter direction, suspended at midwater, and insonified by a broadband directional transducer. From the inspection of the response of the solid-filled shell, it was possible to detect and characterize inhomogeneities of the interior (air inter-layers), the presence of which were later confirmed by CT scan and ultrasound measurements. Elastic wave analysis and modeling tools supported the physical interpretation of the measured responses. A spherical shell partially filled with epoxy resin and water was found to exhibit a strong focusing effect in the backscattering direction, for near-perpendicular incidence with respect to the planar internal water-resin interface. Spheres of this kind have the potential of being employed as sonar calibration targets.
Scattering analysis accelerated by a 3-D multilevel non-uniform grid field evaluation algorithm
Authors: Brick Yaniv , Boag Amir
Published: September 2008

A fast algorithm for computing the scattering cross section of arbitrary shaped large rigid bodies using an iterative method of moments solver has been presented. The main computational bottleneck of such iterative solvers stems from the need to perform at each iteration at least one matrix-vector product. If performed directly, matrix-vector multiplication, which is equivalent to field evaluation for a given source distribution, is characterized by O(N2) complexity (N being the number of unknowns). To that end, a multilevel non-uniform grid (MLNG) algorithm for 3-D fast field evaluation has been proposed, developed, and tested on representative examples of elongated, quasi-planar, and full 3-D scatterers. The algorithm relies on hierarchical domain decomposition, field evaluation on highly sparse non-uniform grids, and multilevel field aggregation through phase-compensated interpolations. Computational complexity and memory requirements of O(Nlog10N) have been achieved by the MLNG without affecting the convergence of the iterative solver. Complexity of the MLNG is similar to that of the multilevel fast multipole algorithm [S. Ko and W. C. Chew, J. Acoust. Soc. Am. 103, 721–734 (1998)]. The MLNG approach is inherently geometrically adaptive, provides seamless transition from the high frequency to quasi-static regime, and is quite easy to implement.