bluuebunny/crossref_metadata_2025 huggingface dataset:
A computational model for auditory scene analysis
Authors: Niessen Maria E. , Van Elburg Ronald A. , Krijnders Dirkjan J. , Andringa Tjeerd C.
Published: May 2008

Primitive auditory scene analysis (ASA) is based on intrinsic properties of the auditory environment. Acoustic features such as continuity and proximity in time or frequency cause perceptual grouping of acoustic elements. Various grouping attributes have been translated into successful signal processing techniques that may be used in source separation. A next step beyond primitive ASA is source identification through schema-based ASA. We present a computational model for ASA that is inspired by models from cognitive research. It dynamically builds a hierarchical network of hypotheses, which is based on (learned) knowledge of the sources. Each hypothesis in the network, initiated by bottom-up evidence, represents a possible sound event. The network is updated for each new input event, which may be any sound in an unconstrained environment. The analysis of new input events is guided by knowledge of the environment and previous events. As a result of this adaptive behavior, information about the environment increases and the set of possible hypotheses decreases. With this method of continuously improving sound event identification we make a promising advance in computational ASA of complex real-world environments.
Speech- synchronized visual cues release speech from informational masking
Authors: Wang Mengyuan Y. , Li Jingyu Y. , Huang Ying , Wu Yanhong , Wu Xihong H. , Li Liang
Published: May 2008

Visual speech information, such as lipreading cues, can assist listeners to segregate a target voice from competing voices (Helfer and Freyman, 2005). However, because signals contained in lipreading are multidimensional, it is not clear whether a simple visual cue, such as the light flash that is synchronous to the onset of each syllable in target speech, is sufficient to release target speech from noise or speech masking. In this study, when target speech was of a constant rate, the speech-synchronized light flash had no effects on speech recognition under either speech or noise masking condition. However, when the rate of target speech was artificially manipulated unstable or an intense noise burst occurred in the middle of the target sentence, the speech-synchronized light flash improved speech recognition when the two-talker speech masker but not the speech-spectrum noise masker was co-presented. These data suggest that only when the rate of target speech cannot be predicted and the masker is speech, speech-synchronized visual cues play a role in helping listeners attend to the target voice and follow the stream of target speech, leading to a release of target speech from informational masking. Supported by the National Natural Science Foundation of China.
Effect of lip movement cues on auditory streaming of concurrent speech
Authors: Devergie Aymeric , Grimault Nicolas , Berthommier Frédéric , Gaudrain Etienne , Healy Eric W.
Published: May 2008

Lip reading is known to induce audio-visual integration processes that enhance speech-in-speech intelligibility. Recent work in auditory scene analysis [see Gaudrain et al., Acoustic'08] found correlations between sequential segregation performance and intelligibility of speech-in-speech. Our working hypothesis is that an audiovisual binding process is involved in speech understanding in multi-talker situations. However, the potential effect of audio-visual integration on auditory stream segregation remains unclear. A single previous study [Gaudrain et al., J. Acoust. Soc. Am. 122, 3065 (2007)] found that lip movements congruent with the presentation of vowels may enhance segregation. In this paradigm, audio-visual sequences of vowels with alternating low and high fundamental frequencies have been generated, in which the lip movements are paired with the second ones. The current experiment is built to test further the potential effect of visual cues on segregation. As in the previous study, lips movements were either steady (control condition), open-closed (rhythm condition) or congruent with each particular vowel. Moreover, a congruent but asynchronous audio-visual condition has been introduced as a new control. This is expected to disrupt the audiovisual binding process. Results are interpreted as an evaluation of the contribution of visual cues in Cocktail Party situations.
Modelling the temporal response of the auditory nerve to the pitch of complex tones in reverberation
Authors: O'Mard Lowel P. , Sayles Mark , Winter Ian M.
Published: May 2008

A computational model of the auditory nerve has been used to investigate the effect of reverberation on the response to the fundamental frequency (F0) of frequency-swept harmonic complexes. The model used a computationally efficient dual resonance non-linear (DRNL) model of basilar membrane frequency selectivity. The DRNL architecture comprises two parallel filter paths providing linear and non-linear responses. The model successfully reproduced the results of a physiological investigation using the same stimuli in a population of single units from the ventral cochlear nucleus (VCN). In particular the effect of reverberation was dependent upon unit best frequency, F0 and source-to-receiver distance. The model results suggest that the effects of reverberation observed at the level of the VCN are already present in the auditory periphery. The use of the model has also enabled us to examine the effect of hearing impairment by the removal of the non-linear path response. This simulation of a sensorineural hearing loss accentuated the loss of the temporal representation of F0 in the presence of reverberation.
Intelligibility of interrupted speech in normal-hearing listeners and cochlear implantees
Authors: Gnansia Dan , Pean Vincent , Lorenzi Christian
Published: May 2008

The intelligibility of a target speech signal is substantially degraded in the presence of a competing talker in cochlear implantees. The current study aimed to test whether this degradation is caused by an inability to i) make use of the partial target speech information glimpsed into the competing talker "valleys", and ii) fuse perceptually those successive glimpses into coherent speech streams. This hypothesis was tested by assessing the intelligibility of periodically interrupted vowel-consonant-vowel (VCV) signals pronounced by a male (F0=113 Hz) and female (F0=216 Hz) speaker in normal-hearing listeners and implantees. A 4-Hz square-wave modulator with random phase was used to interrupt periodically each signal. The interrupted VCVs from each speaker were either presented alone (Experiment 1) or interleaved (Experiment 2: the two speakers were alternated). In experiment 1, the mean identification score calculated for each voice was about 90% in normal-hearing listeners, and 40% in implantees. In experiment 2, the mean identification score corresponding to correct identification of both voices was about 50% in normal-hearing listeners, but at chance level in implantees. Taken together, these data suggest that implantees can make use of partial speech information, but cannot organize this partial speech information into coherent streams.
Cartilage conduction hearing aid for the patient with atresia auris
Authors: Sakaguchi Takefumi , Saito Osamu , Hosoi Hiroshi
Published: May 2008

Although bone conduction hearing aid is considered to be better to compensate the hearing loss of the patients who have atresia auris, some of these patients tend to wear air conduction hearing aid because of the feeling of tightness when wearing bone conduction haring aid. We expected cartilage conduction would reduce feeling of tightness compared to bone conduction, and improve compensation compared to air conduction. In this study, we report the result of the basic studies performed to evaluate the usefulness of cartilage conduction hearing aid. Patient who has atresia auris participated in the hearing test. Stimuli were presented to the patient by means of two types of transmitter. One was piezoelectric transducer placed on antilobium, another one was insertion earphone. We found that the hearing threshold of the patient improved about 25 dB or greater when the transducer was located on antilobium compared to the result when insertion earphone was used. These results suggest that the cartilage conduction hearing aid can be an option of the hearing aid for the patient who has atresia auris which is better than air conduction hearing aid in compensating their hearing, and more comfortable to wear compared to the bone conduction hearing aid.
Vector intensity reconstructions in a volume surrounding a rigid spherical measurement array
Authors: Williams Earl G. , Takashima Kazuhiro
Published: May 2008

An approach is presented that provides a prediction of the vector intensity field throughout a volume exterior to a rigid spherical measurement array consisting of 31 flush mounted microphones. The theory is based on spherical harmonic expansions of the measured field with the radial variation of the near-field pressure obtained using the Greens function with vanishing normal derivative at the rigid sphere surface. Experimental results with rigid spherical arrays of differing radii are presented using multiple incoherent sources. Successful intensity reconstructions are obtained over a volume three times the sphere radius up to a frequency of 1.5 kHz that clearly reveal the locations and levels of the two sources. This volumetric intensity probe is very similar mathematically to one described recently by the author (EGW) that used 50 microphones in an open array. The latter was used successfully inside an aircraft cabin in flight to uncover sources of noise. This work was supported by the US Office of Naval Research and Nittobo Acoustic Engineering Co. Ltd.
Acoustically invisible cylinder
Authors: Bobrovnitskii Yuri
Published: May 2008

Coatings of new type recently proposed by the author (Acoustical Physics, 2007, vol. 53, N5, pp. 535-545) are applied to bodies of cylindrical geometry to reduce reflection or scattering of sound and thus to make them undetectable by imaging systems. Such a coating, called as a coating with extended reaction, represents a periodic set of small elements with coupling between the neighboring elements. Appropriate choice of the coupling parameters makes its efficiency much higher than that of commonly used coatings. In the present paper, it is shown by computer simulation that a rather simple coating of this type can reduce the back-scattered pressure amplitude more than 40 dB (with respect to the rigid cylinder) practically at all frequencies. Considerable reduction of the scattered power can also be achieved in a low frequency range. The width of this range and the reduction index depend on the number of couplings introduced into the coating.
3D localization of acoustic sources with a spherical array
Authors: Haddad Karim , Hald Jo/rgen
Published: May 2008

This paper describes a technique dedicated for the localization of acoustic sources in all directions and in the far-field. Classical beamforming techniques based on planar arrays provides an acoustic map restricted to a limited aperture, but a spherical array does not have such a limitation since there is no preferential direction. In this processing called Spherical Harmonics Beamforming, the sound field on the sphere is decomposed with spherical harmonics functions, and then a corrected summation gives the acoustic contribution from a given direction. We have used a rigid spherical array, which has the advantage that cabling of microphones and integrated cameras can be hidden inside the sphere. A rigid surface also provides better numerical stability in connection with Spherical Harmonics Beamforming. In this study, this technique is evaluated in connection with the resolution and the dynamic range. Simulated and experimental results are presented.
Measurement of force and moment mobilities using a finite difference technique
Authors: Elliott Andrew , Pavić G , Moorhouse Andy T.
Published: May 2008

Measuring only force mobilities it is possible to derive a moment mobility without the need for an externally applied moment. The method uses finite differences of both forces and velocities about a central point. In this way pure forces and moments can be extracted as can pure velocities and angular velocities. There are, however, errors associated with finite difference techniques. In this case the purity of the extracted quantities depends on the area over which they are measured and its significance in terms of wavelength. Meanwhile experimental errors such as noise dictate that the area is sufficiently large to provide differences which are incorruptible. The paper aims to quantify the finite difference error in a general way free of situation specifics.
Plate Sound Radiation Prediction in Machines Using Multiple Input Techniques
Authors: Nejade Armand
Published: May 2008

In systems with various types of noise sources, the characterization and ranking of each source may require a different approach. Also, one of the main challenges in such investigations, consists of isolating each source from the influence of the others. Often in the machines, the constituting plates are efficient radiators whose specific contributions need to be properly determined. In this presentation, we describe a method of plate radiation characterization using multiple input techniques. These techniques have frequently been used in case of discrete sources such as, for instance, vehicle engine cylinders where the pressure variations are modelled as the inputs. This application differs from the traditional ones by the fact that the inputs are now, the elements of a continuous medium. Here the surface of a plate is meshed, each element is taken as a sub-source and consequently, the plate is modelled as an assembly of sub-sources. The plate contributions at the operational modes, are then obtained through vibration measurements and therefore, are not affected by the noise from the other sources. Investigations on structures of various geometries and materials have so far, demonstrated the efficiency of this type of application.
Refracto-vibrometry - a novel method for visualizing sound waves in transparent media
Authors: Zipser Lothar , Franke Heinz H.
Published: May 2008

For optimization of new sonic techniques, for development of acoustic sensors or for understanding of complex sonic phenomena it is often desirable to visualize the generation and distribution of the invisible sound waves. A novel method for measuring and visualizing the sound waves in transparent media was developed and tested by the authors. This non-contact method is based on a scanning laser-Doppler vibrometer. It can be used for measuring and visualizing sound fields in gases, fluids and even pellucid solid objects. The variation of the optical refractive index n of transparent media, caused by the sound waves is used as measuring effect. This justifies the denomination "refracto-vibrometry". The paper describes the theoretical fundamentals and the technical equipment required for the novel method. As examples, the generation and radiation of sound in air, water and transparent solid objects are presented. The animation of the results as videos is very impressive and instructive. In research, education and system design refracto-vibrometry has a large potential to make acoustics clearer and more feasible for application.
Noise Prediction for a Pumping Loop using the Noise Synthesis Technology Approach
Authors: Pepin Henri , Hass Axel
Published: May 2008

A collective project involving the french manufacturers of industrial pumps and valves is dedicated to the noise prediction in pumping plants. This paper presents a methodology based on virtual acoustic prototyping, and specifically on the noise synthesis technology (NST). Prediction of radiated noise in particular locations, such as operator stations or contractual specifications, allows us to control from the conception stage if the industrial plant will meet the prescribed noise criteria (regulations or user specifications). In addition, related noise control recommendations can be achieved by comparison of data obtained from the individual response of the components and their interaction. This methodology does not work as an absolute predictive process, and thus requires experimental validation. Components are taken into account by establishing their detailed vibroacoustic characteristics (technical data or measurements); in the present work on a simplified pumping loop, selected components are a centrifugal pumping unit, an open pipe circuit and several type of valves. Vibroacoustic and hydroacoustic measurements have to be adjusted according to the NST requirements. Comparison of the NST results with experimental data show a wide validity domain, covering the 100-10kHz third-octave range.
An analysis of the modes of wave propagation in cylindrical shells
Authors: Liu Yu-Cheng , Hwang Yun-Fun , Huang Jin-Huang
Published: May 2008

Similar to a recent paper by Karczub [J. Acoust. Soc. Am., Vol. 119(6), 3553-3557, 2006], this work utilizes a symbolic math solution to obtain analytical solutions for the dispersion relation in a thin circular cylindrical shell. Analyses on the propagating wave mode shapes are particularly emphasized. The analytical solutions provide a straightforward calculation for all four branches of the dispersion curves, and the cumbersome numerical root searching technique is avoided. The dispersion relations for the breathing and torsional modes (n=0 circular harmonic), which were neglected by Karczub, are included. Mode shapes that characterize the types of wave that are predominantly transversal, longitudinal, circumferential, or in any combination of the three are analyzed in considerable detail. Effects by thickness and Poisson ratio variations are considered and examined. The cutoff frequencies for the higher order propagating modes are compared among the various orders of circular harmonics, from n = 0 to 10. Finally, comparison among several shell theories in the dispersion relation prospective, which is directly related to their accuracy in predicting the vibratory resonant frequencies of a finite shell, is also included in the discussion.
The Investigation on Measuring the Reflection Coefficient of Materials in Semi-space by Spatial Fourier Transforms
Authors: Liu Yongwei , Shang Dejiang , Li Qi
Published: May 2008

A new method using SFT (Spatial Fourier Transforms) has been investigated to measure reflection coefficient at arbitrary angles of incidence by NAH(Near-field Acoustical holography ) in semi-space. The paper gives an experimental verification of this method. The experimental frequency is from 3 kHz to 10 kHz. The size of test steel plate is 1.6m×1.5m,the size of hologram surfaces is 1.32m×1.14m, the distance between two hologram surface and the test material is about 2cm, the distance between sound source and water surface is about 4cm. The measured results are generally in good agreement with the theoretical value; the mean-square error of data is below 0.06. The experimental results show that this method eliminates the effect of the scattered sound field of the water surface completely. Because the sound source and the test material are set near the water surface during the experiment, it is very easy to set the measuring system, and this is very useful in practice.
Characterising elastic layers as non-lightly damped SDOF systems for the reduction of impact transmission noise
Authors: Simon Francisco , Anthony David K.
Published: May 2008

There is a wealth of literature dealing with modal determination of lightly damped viscous systems, however there is relatively little reported for medium and heavily damped systems or with non-viscous damping. The most commonly reported and applied methods have significant errors in the determination of both modal frequency and damping ratio for systems with high damping. For the lightly damped case simplification can be often conveniently made to allow the performance of each method to be studied theoretically, however for the non-lightly damped case this is not possible and the analysis must proceed on a heuristic basis. For the application modal parameter determination of elastic floor layers designed to reduce building transmission noise by impact testing, traditional and new methods of modal parameter determination for non-lightly are presented and heuristically evaluated for both viscous and non-viscous SDOF systems. It is also shown the performance often depends on the type of damping. This either must be known a priori or can be gleaned by comparing results from different methods.
The role of generalization in learning perceptual contrasts
Authors: Sumner Meghan
Published: May 2008

Understanding spoken language is a difficult task. Variation caused by accents make this task even more daunting. This project examines the generalization of novel phonetic cues by listeners of non-native speech. Native speakers of English were trained on Korean-English pronunciations of minimal pairs with word-final contrasts (e.g., bed - bet). While native English speakers rely on vowel duration to cue a final voicing contrast, native Korean speakers of English do not produce a vowel duration contrast, but use a different distinguishing phonetic cue (e.g., voicing into closure duration). Listeners were trained on a subset of minimal pairs ending in final stops (bet - bed). Following successful training, listeners participated in an immediate or long-term form priming task. Four conditions included: Trained Identity (bet - bet), Trained Non-identity (bed - bet), Novel Non-identity, Trained Rime (wet - wed), and Novel Non-identity, Novel Rime (beat - bead). Preliminary results show episodic treatment of words in the immediate priming task, but generalization to novel forms in the long-term priming task. These data suggest that early learning is mediated by episodic representations - learning proceeds episodically until sufficient information is available to support a more abstract generalization.
Arabic Diagnostic Rhyme Test using minimal pairs
Authors: Boudraa Bachir , Boudraa Malika , Guerin Bernard
Published: May 2008

In this paper, we try to adapt to Arabic the diagnostic rhyme test (DRT) developed by Voiers for English and adapted to French by Peckels & Rossi. DRT is specific to each language and is essential to subjective assessments of coders and synthesizers. Six Arabic pertinent dimensions are used which are acuteness, compactness, tenseness, stridence, nasality and flatness. 72 monosyllabic meaningful pairs of words called minimal pairs have been developed. The apprehensibility of every attribute is tested in each of six vowel contexts. Every feature has been repeated six times and has been paired twice with every vowel.
A fascinating personage in the history of phonetics: The phonetics and the politics of Elise Richter
Authors: Warner Natasha
Published: May 2008

Elise Richter (1865-1943) was the first woman lecturer or professor in Austria or Germany, and she was also an experimental phonetician and even an experimental phonologist. Richter's career was delayed by the roles allowed for women at the time, and the end of her career, and her life, came about through the Nazis' policies toward Jews. Richter began her scientific work in Romance philology, but decided to seek explanations for historical sound change in phonetics. This led her to a productive career in experimental phonetics, and she even applied phonetic methods to study questions of phonological structure. Richter was a leader in exploring new methods and bringing together questions across widely differing fields. Although her specific experimental findings were superseded with time, Richter's refusal to be constrained by the limits of a field led her to ideas that have become current in phonetics just recently. In addition to her scientific career, Richter was politically active, both in working to expand women's opportunities and in general politics. This poster presents a new perspective on Richter's contributions to phonetics and on her view of feminism. Furthermore, it makes new observations of a parallel between Richter's political activities and her scientific work.
The role of tongue articulation for /s/ and /z/ production in whispered speech
Authors: Yoshioka Hirohide
Published: May 2008

Although the timing of the initiation and cessation of vocal fold vibrations is crucial to characterize the voiced and voiceless cognates, other cues, such as the duration of preceding vowels, the patterns of the formant transitions in the following vowels, and the period of stop closure, may also play important roles in performing these distinctions. The present study is to further investigate the role of tongue articulatory movements during voiced and unvoiced consonant production, specifically when the vocal folds do not vibrate during the production of fricative consonants, /s/ and /z/ in whispered speech. A normal Japanese speaker served as the subject. The palato-lingual contact patterns during the intervocalic consonant /s/ and /z/ in whispered speech were recoded using dynamic electro-palatography. The results show that the area of palato-lingual contact is clearly wider during /z/ production than /s/ production; the opening for fricative turbulent noise production is narrower and longer for /z/ than for /s/. In addition, the contact pattern is unstable for /s/ production.
Duration of Japanese singleton and geminate obstruents in two- to four-mora words
Authors: Hirata Yukari , Kato Hiroaki , Tajima Keiichi
Published: May 2008

Hirata and Whiton [J. Acoust. Soc. Am. 118, 1647-1660 (2005)] found an invariant durational structure for Japanese stop quantity distinction in two- and three-mora words across different speaking rates. The present study examined whether their finding extends to include fricative and affricate quantity distinctions and three- and four-mora words. Stimuli were stop, fricative, and affricate contrasts in four types of words, including (1) a long vowel (e.g., ka.so.o vs. ka.s.so.o), (2) a moraic nasal (e.g., ji.se.n vs. ji.s.se.n), (3) CV sequence (e.g., ho.so.ku vs. ho.s.so.ku), and (4) shorter words (e.g., i.shi vs. i.s.shi), spoken in isolation at three speaking rates by four native Japanese speakers. Duration of contrasting obstruents, words, and the interval between the onsets of the first and the second vowels (VOI) was measured. Results indicated durational patterns similar to those found previously for stop contrasts. In addition, duration of words, regardless of their segmental composition, reflected well the number of moras they contained. Finally, the ratio of the contrasting obstruent to the word and the ratio of the VOI to the mean mora duration were useful in classifying the singleton and geminate categories across rates. [Supported by Grant-in-Aid for Scientific Research, JSPS]
English /r/-/l/ pronunciation training for Japanese speakers
Authors: Hattori Kota , Iverson Paul
Published: May 2008

Previous studies have demonstrated that perceptual training improves both perception and production by adult second-language (L2) learners. The present study examined whether production training likewise improves both perception and production. Japanese speakers underwent ten sessions of production training for English /r/ and /l/. The training combined three methods: (1) subjects received audio-visual examples and pronunciation instructions for /r/ and /l/; (2) they produced mono syllables (i.e., /ra/ and /la/) and minimal-pair words, with online feedback about their formant frequencies using a real-time spectrogram as well as pronunciation feedback from an instructor; and (3) they made recordings of minimal-pair words and compared them to "enhanced" versions of these recordings (i.e., signal-processed versions with the correct F3 frequencies and transition durations). All participants completed a battery of pre/post training tests (i.e., English /r/-/l/ identification, discrimination, perceptual mapping of best exemplars, and production). The preliminary results indicate that this training approach improves production; the results will be further discussed in terms of its effect on perception and the underlying representation of these categories.
Differential effects of the phonemes on identification of previously unknown speakers
Authors: Amino Kanae , Arai Takayuki
Published: May 2008

In perceptual speaker identification, it is known that the identification accuracy depends on the contents of the stimuli presented to the listeners. In our previous experiments, we found that the stimuli containing a nasal sound are effective for human speaker identification, and that coronal sounds are more effective than labial ones. This tendency was observed in the identifications of both familiar and previously unknown speakers. In this present study, we investigated the effects of the speech contents again, by focusing on the vowels in CV monosyllabic stimuli. Through the experiment we obtained several findings: 1) stimuli containing a nasal gained significantly higher accuracy compared to stimuli with only oral sounds; 2) coronal sounds were more effective than labial sounds; 3) palatalisation caused an improvement in performance; and 4) back vowels were more effective than front vowels significantly. These results can be explained by speaker-specific morphologies of the nasal cavity and the paranasal sinuses, and also of the pharyngeal cavity. We will also show analyses and discussions on the acoustical properties of the stimuli and the performance differences among the subjects. [Work supported by Sophia University Open Research Centre from MEXT.]
Acquisition of rhythm: evidence from spontaneous L2 speech
Authors: Shport Irina
Published: May 2008

This study examined the acquisition of Japanese rhythm by comparing the rhythmic characteristics of elicited spontaneous speech of English learners in the setting of an oral proficiency test. Two metrics for the quantification of speech rhythm that have been shown to reliably discriminate rhythmic classes in both L1 and L2 speech (White and Mattys, 2007) were utilized for the analysis - 1) the proportion of vowel intervals within the sentence (%V), and 2) the standard deviation of vowel intervals divided by the mean vowel interval duration and multiplied by 100 (VarcoV). A difference in the rhythm scores of novice-mid and intermediate-high learners of Japanese was found. Furthermore, the learners' scores were significantly different from Japanese native speakers' scores, suggesting that speakers start acquiring speech rhythm relatively early, but they are still short of the target at the intermediate level of proficiency. By comparing the obtained scores with scores of stress-timed languages, such as English, the difference in the scores may be attributed to an L1 transfer into the rhythmic organization of the L2 speech.
Prosody in French theatrical declamation traced backwards in time
Authors: Brock-Nannestad George
Published: May 2008

Listening to early recordings of a French theatrical personality, such as Sarah Bernhardt, the emphatic emission of words is very noticeable. It is sometimes ascribed to Bernhardt's theatrical mannerisms, but possibly also caused by an erroneous speed of reproduction. But even at correct speed, huge variations in intenseness are very noticeable. Literature does not give a clear indication whether this manner of performance was considered deviating at the time (ca. 1900). New research into the original handwritten patent documents of Édouard-Léon Scott (filed 1857 and 1859) has analyzed his method of recording the performance of the human voice. Scott documented the performance of a complete stanza from a French play, and looking at his graphical output and interpretation it becomes clear that the prosody of Mme. Bernhardt was already in place in 1859. Having identified the French play, even earlier descriptions of the prosodic approach in French declamation were found. The paper also discusses the patents of Édouard-Léon Scott and his desire for the use of his invention, which was different from that of his manufacturer, Rudolph Koenig. Furthermore, the prosodic documentation will be demonstrated, and a historically informed reproduction of a recording by Sarah Bernhardt will be presented.
Articulatory coordination in German and English onset and coda clusters
Authors: Pouplier Marianne , Marin Stefania
Published: May 2008

Articulatorily, onset consonants differ from coda consonants in their spatial as well as their temporal characteristics in that onsets exhibit greater spatial displacement and are generally less variable than codas. As to their temporal organization, it has been claimed that onsets, but not codas exhibit a so-called c-center organization in which all consonant gestures of a cluster are organized as a single complex structure with respect to the vowel, while consonant clusters in codas have been hypothesized to be organized sequentially [Browman & Goldstein 1988 Phonetica]. While some studies have confirmed the c-center hypothesis for onsets but not codas, others found a c-center organization for both onset and coda consonants. The current study uses German and English onset and coda clusters to test systematically the c-center hypothesis on the basis of EMA data. Results for both languages indicate that a c-center organization may be observed in both onset and coda position, yet it does not appear consistently for all clusters. Also cross word-boundary clusters may exhibit c-center organization, possibly indicating resyllabification.
Embodiment, not imitation, leads to the replication of timing phenomena
Authors: Messum Piers
Published: May 2008

In many languages, a spoken vowel is shorter before phonologically voiceless consonants than before voiced ones. In West Germanic languages, including English, tense and lax vowels have different lengths and there is a characteristic 'stress-timed' rhythm. For these phenomena and some others, it is generally assumed that time is the controlled variable for production, and that children replicate these speech behaviors through the reproduction of timing patterns abstracted from the adult input (i.e. by imitation). The developmental data does not support these assumptions, and an imitative mechanism would present a young speaker with a highly complex challenge. Instead, these behaviors are more plausibly the result of the embodiment of speech. Embodiment goes beyond laryngeal and articulatory function. A child's speech breathing is not a scaled-down version of the adult model but a distinctly different skill, and one that must be learnt during speech production. Similarly, the aerodynamic setting of child speech differs significantly from that of adults. The constraints that these factors impose become manifest in speech as changes in timing, but these changes are epiphenomenal, not modelled directly. Phenomena particular to West Germanic languages reflect the style of speech breathing these languages require of a child.
Vowel place detection for a knowledge-based speech recognition system
Authors: Lee Sukmyung , Choi Jeung-Yoon
Published: May 2008

This work aims to detect vowel place as part of a knowledge-based speech recognition system. Vowel place was classified into 6 groups based on tongue advancement [Front/Back] and height [High/Mid/Low]. Experiments were performed using 300 /hVd/ utterance data from Hillenbrand [J. Acoust. Soc. Am. 97, 3099-3111] and 6600 TIMIT vowels. Features used include fundamental frequency (F0) and formant value (F1̃F3), where formant measurements were classified into separate groups using F0 measurements. The nearest class was found using a simple Mahalanobis distance measure, and yielded a 91.5% classification rate for the /hVd/ data. The results for the TIMIT data were 64.4%, and error analysis with regard to adjacent segment manner and place was carried out to observe the effects of coarticulation, which was not observed in the /hVd/ data.
Cross-modal perceptual learning of spectrally degraded speech: Behavioral and neuroimaging studies
Authors: Eisner Frank , McGettigan Carolyn , Rosen Stuart , Faulkner Andrew , Scott Sophie K.
Published: May 2008

We investigated normal-hearing listeners' ability to adapt to a speech signal that simulates aspects of the stimulation received from a cochlear implant. The training materials were spoken sentences which were spectrally degraded by noiseband-vocoding and shifted upwards in frequency. A control condition consisted of spectrally inverted, unintelligible versions of these stimuli. Participants listened passively to these sentences, and on each trial received visual feedback which could be either a written version of the sentence or a video of the talker who originally produced it. Learning under both feedback conditions was relatively fast: Subjects improved on average by 25% on keyword recognition scores after 100 trials. No learning effects were observed in the control condition. We further used functional magnetic resonance imaging to investigate which cortical areas may be recruited during learning. A comparison of degraded and learnable sentences with the unlearnable control stimuli showed activity in the left superior temporal sulcus both during passive listening and receiving feedback. In contrast, the left inferior frontal gyrus was activated only when subjects were receiving feedback in the learnable condition. These results suggest that the inferior frontal gyrus plays an important role in integrating acoustic-phonetic processing with externally provided feedback.
The '[+English]' effect in acquisition of English as a second language: Novel- category biases in second language perception and production
Authors: De Jong Kenneth , Hao Yen-Chen , Park Hanyong , Silbert Noah H.
Published: May 2008

Silbert & de Jong (JASA, 122: 3092) investigated correlations between response bias and segment frequency in a number of previously published segment identification data sets. In general, bias tends to be positively correlated with frequency. When applied to second language (L2) learning, this pattern predicts biases toward identifying novel L2 categories as L1 categories, since the number of instances of L1 categories far outweighs those of novel categories in learners' experience. Nagao, Lim, & de Jong (2003, ICPhS) confirm this: Japanese listeners tend to label ambiguous English coda consonants as onsets. However, younger and less experienced learners of English tend to label ambiguous tokens as codas; they tend toward using the novel category even though it is less frequent in their experience. This paper reports experiments in which Korean EFL learners identified English obstruents and in which they produced them. Cases of novel category response bias are observed in identification, and less so in production. Such novel category effects suggest a model in which novel categories serve, in part, as indicators of the L2 itself and stand out as response options in L2 identification tasks. These effects are likely counteracted by motor difficulties in L2 production tasks.
Study on voice quality parameters for anger degree estimation
Authors: Arimoto Yoshiko , Ohno Sumio , Iida Hitoshi
Published: May 2008

With great advance of automatic speech recognition (ASR) systems and a voice command system are demanded to be more sensitive to user's intention or emotion. These systems currently process linguistic information, but not process nonlinguistic information or paralinguistic information which users expressed during dialogs. For that reason, computers can obtain less information about a user through a dialog than human listeners can. If computers will recognize user's emotions conveyed by acoustic information, more appropriate response can be made toward users. For realization of emotion recognition, we have continued our study on anger degree estimation by both prosodic features and segmental features with anger utterances which were recorded during two kinds of pseudo-dialogs. This report focuses on only segmental features related to voice quality and examines them for capabilities to estimate anger degree. The first cepstral coefficient of anger utterances has been analyzed to obtain acoustic parameters related to spectral tilt in our previous works. In addition to the previous voice quality parameters, features related to amplitude of formants and harmonics were extracted from the same anger utterances as that in the previous works. Comparisons were made with the previous parameters to evaluate estimation accuracy of each voice quality parameters.
Acoustic vector-sensor array beamforming based on fourth-order cumulants
Authors: Li Tingting , Li Xiukun
Published: May 2008

Abstract: Some high-resolution direction-of-arrival (DOA) estimation methods, such as the MUSIC method, have been developed based on vector-sensor array. However, these high-resolution methods suffer from serious drawbacks. Indeed, they are not able to estimate coherent signals and sensitive to calibration errors. Mainly to overcome these limitations and in particular to satisfy practical engineering application, vector-sensor array beamforming based on fourth-order cumulants is adopted in this paper. The main interest in using fourth-order cumulants instead of second-order ones in vector-sensor array processing application is the aperture extension property of higher-order cumulants, which makes it possible to increase both the resolution and the number of sources to be processed from a given array, and suppress the Gaussian noise. Based on this method, smaller size of array and higher accuracy of DOA estimation can be realized, besides, this method is less sensitive to the coherence of the signals and calibration errors, which makes it possible to work well in engineering application. The results of computer simulation and analysis of data from outfield trials show that this method excels traditional direct estimation and can improve estimation capability of vector-sensor arrays.
Beam-forming with progressive focalisation
Authors: Caronna Gaetano , Testa Pierluigi
Published: May 2008

This investigation is aiming to detect and track a moving target using its acoustic emissions and applying a new beam-forming approach. An original algorithm of medium computational complexity is reported that is capable of locating, with good accuracy, the direction of an acoustic source with only a few microphones. This algorithm circumvents the known localisation ambiguities due to the unwanted "grating lobes", by using all spectral content of the acoustic source and progressively and narrowly focusing its beam in the direction of the target. The algorithm starts by locating the direction of the target with low angular resolution, using only low frequency components of the acoustic emissions; in this way, no grating lobes occur even with only a few microphones. Then, the angular resolution is increased by processing higher frequencies of the spectrum, but profiting of the information on the angular position achieved in the previous stages. Some algorithmic means are introduced to increase the performance of the algorithmic against environmental noise. This scheme is described in detail. Some experimental tests are included and discussed.
Detection of Ultrasonic closer flaws using Nonlinear signal processing
Authors: Benammar Abdessalem , Drai Redouane , Kechida Ahmed , Guessoum Abderrezak
Published: May 2008

The ultrasonic flaw detection is an important problem in the nondestructive evaluation (NDE) of materials. In order to successfully detect and classify flaw echoes from high scattering grain echoes, an efficient and robust method is required. In this paper, a method using split-spectrum processing (SSP) combined with a neural network (NN) has been developed and applied on the ultrasonic signals to perform the detection of closer echoes. SSP can display signal diversity and is therefore able to provide the signal feature vectors for signal classification. The neural network (NN) performs highly complex nonlinear mapping by which signals can be classified according to their feature vectors. Therefore, the combination of SSP and NN (SSP-NN) presents a powerful technique for ultrasonic NDE. The SSP is achieved by using Gaussian bandpass filters. Then, an adaptive three layer neural network using a backpropagation learning process is applied to perform the classification processing of frequency diverse data. The SSP-NN method has been tested using both simulated and experimental ultrasonic signals, and the results show that SSP-NN has good sensitivity in the detection of ultrasonic closer flaws echoes drowned in the noise.
Time reversal elastography in soft solids
Authors: Gallot Thomas , Catheline Stefan , Benech Nicolas , Brum Javier , Negreira Carlos , Roux Philippe
Published: May 2008

One channel time-reversal (TR) experiments allow focalization of waves in reverberant cavities. According to Rayleigh criteria, the focal spot width is directly related to wavelength and therefore depends on the mechanical properties of the medium. Thus the general idea of this work is to extract quantitative estimations of these mechanical properties from TR experiments. They were performed inside 3D soft solid cavities (gelatin based phantoms). An external source creates mechanical waves in the audible frequency range and one component of the vectorial field is measured as function of time by ultrasonic techniques developed in elastography. The wavelength information of shear waves, that dominates the vectorial field, is deduced from TR experiments. The advantage is that the technique works with sources of any shape and with any time dependence. This robustness as regard to shear wave source allows to think about many applications in the medical field, including deep or moving organs. Examples of elasticity image will be shown in inhomogeneous soft tissues.
Continuous estimation of VTL from vowels using a linearly VTL-covariant speech feature
Authors: Feldbauer Christian , Monaghan Jessica J. , Patterson Roy D.
Published: May 2008

Vowel sounds encode information about the speaker's vocal tract length (VTL) as well as vowel type. This paper shows how VTL can be estimated on a frame-by-frame basis from vowels to facilitate speaker tracking and make automatic speech recognition (ASR) more robust in multi-speaker environments. The algorithm is based on a new, VTL-covariant speech feature that encodes the size information linearly. The paper shows that this new speech feature is better suited for VTL estimation than the more traditional cepstral coefficients. The VTL estimation is based on a Gaussian mixture model that was trained on the speech material of speakers with known body height but unknown VTL. Research supported by the Austrian Science Fund (FWF) [J2541-N15], EOARD [FA8655-05-1-3043], and the UK-MRC [G0500221].
Source localization and tracking in a waveguide via sequential bayesian estimation
Authors: Cockrell Kevin , Schmidt Henrik
Published: May 2008

This talk addresses the problem of sequentially estimating the location of a moving acoustic source in a waveguide, given a series of acoustic field measurements from an array. The approach taken, in it's most general form, is known as sequential Bayesian estimation. A well known special case of sequential Baysian estimation is the Kalman filter. However, the Kalman filter is only optimal when the relationship between the measured parameters and the ones to be estimated is linear with Gaussian noise. Given the highly non-linear relationship between the acoustic source location and the measured acoustic field, other approaches must be taken. The performance and computational feasibility of various approaches, such as grid-based methods, will be explored using simulated data.