bluuebunny/crossref_metadata_2025 huggingface dataset:
Roughness detection in fricative-like noise and tone stimuli
Authors: Pincas Jonathan , Jackson Philip J.
Published: May 2008

Audio (spectral) and modulation (envelope) frequencies both carry information in a speech signal. While low modulation frequencies (2-20Hz) convey syllable information, higher modulation frequencies (80-400Hz) allow for assimilation of perceptual cues, e.g., the roughness of amplitude-modulated noise in voiced fricatives, considered here. Psychoacoustic 3-interval forced-choice experiments measured AM detection thresholds for modulated noise accompanied by a tone with matching fundamental frequency at 125Hz: (1) tone-to-noise ratio (TNR) and phase between tone and noise envelope were varied, with silence between intervals; (2) as (1) with continuous tone throughout each trial; (3) duration and noise spectral shape were varied. Results from (1) showed increased threshold (worse detection) for louder tones (40-50dB TNR). In (2), a similar effect was observed for the in-phase condition, but out-of-phase AM detection appeared immune to the tone. As expected, (3) showed increased thresholds for shorter tokens, although still detectable at 60ms, and no effect for spectral shape. The phase effect of (2) held for the short stimuli, with implications for fricative speech tokens (40ms-100ms). Further work will evaluate the strength of this surprisingly robust cue in speech.
Assessment of exposure to impulsive and continuous noise by auditory brainstem response
Authors: Zera Jan , Kochanek Krzysztof , Pilka Adam , Mlynski Rafal
Published: May 2008

The aim of the work was to measure the wave V latency or thresholds in the auditory brainstem responses (ABRs) produced by impulsive and continuous noise. A forward-masking paradigm was used in which the ABR was evoked by a 4-kHz tone pip. The tone pip was masked by a preceding 201-ms or 501-ms interval of click trains or band-pass noise presented at various levels. The inter-click interval ranged from 20 ms (50 clicks/s) to 100 ms (10 clicks/s). The center frequencies of the noise bands ranged from 250 to 4000 Hz. Results show that changes in wave V latency may be used as an indicator of the equivalence of the effect of various kinds of noise on the human auditory system. [Work supported by the Polish Ministry of Science and Higher Education, grant 4T07B00428].
Training of English vowel perception by Finnish speakers to focus on spectral rather than durational cues
Authors: Uther Maria , Ylinen Sari , Latvala Antti , Akahane-Yamada Reiko , Iverson Paul
Published: May 2008

This study used the High-Variability Phonetic Training (HVPT) technique to train Finnish speakers to distinguish English vowels. It was found that Finnish speakers tend to use durational cues (which are phonemically relevant in their own language) to make a vowel category distinction rather than the relevant spectral cues. We used duration-modified stimuli with a HVPT program to 'force' the use of spectral cues. We focused on the /i/ (as in 'feet') vs /I/ (as in 'fit') vowel contrast and tested behavioural performance using a perceptual identification task. We also measured the Mismatch negativity (MMN) component of auditory event-related potential (ERP) before and after the training to look at changes in brain responses. The worst pre-test performance was for the 'modified duration' condition (i.e. where the learner had no choice but to rely on spectral cues). There were also asymmetries in vowel perception in both behavioural and MMN tasks, with the detection of /i/ being more difficult compared to the detection of /I/. Nevertheless, training did result in marked improvement of the most problematic contrasts for Finnish speakers.
Detection threshold and prominence of background music with interfering speech
Authors: Houben Mark M.
Published: May 2008

Spoken radio and television messages are often accompanied by background music. When automating identification of broadcast music for handling copyright exploitation, it may be questioned to what extent music at very low sound levels should be taken into account. Should it just be perceivable by listeners who attend to the speech, or should the music be present more prominently? We studied detection and level of prominence of music with interfering speech. For this, Dutch radio and television broadcasts were recorded and speech fragments without background music were selected. To the fragments, music was added with various level differences between speech and music. Subjects were presented with these stimuli and answered a random generic question about the speech after each stimulus. As a second task, the prominence of the background music had to be scored on a scale (including 'not present'). Results show that, on average, 50% of the music is detected if the level of the music is 45 dB lower than the speech. A threshold based on a criterion of 'moderate prominence' results in a level difference of 26 dB. This level happens to coincide with a detection threshold of 95%.
Fast Detection for Natural Animal Sounds
Authors: Suied Clara , Magnin Marie , Langlois Sabine , Susini Patrick , McAdams Stephen
Published: May 2008

Human listeners seem remarkably good at identifying complex and natural sounds. Relatively little attention, however, has been paid to the simple detection of such sounds. The present experiments measured the time course of sound detection in human listeners. Two categories of sounds were used: a train of 1-kHz pure tone pulses varying along an Inter Onset Interval dimension (termed here "IOI" sounds) and animal call sound samples. All sounds were equalized in loudness in a preliminary experiment. The task consisted of a simple reaction time, in which listeners were presented with interleaved IOI and animal sounds and had to manually press a button as fast as possible when they heard any stimulus. Two main results were observed. First, reaction times were significantly shorter for IOIs of 50, 33 and 25 ms than for an IOI of 100 ms, with no significant differences between IOIs of 33 and 25 ms, highlighting a threshold at 33 ms. Second, average reaction times were significantly shorter for animal sounds than for IOI sounds. Differences in terms of spectral content and temporal modulation characteristics might explain part of these effects. These results also suggest a fast detection advantage for natural animal sounds.
The effect of postural information on the perceived velocity of moving sound sources
Authors: Sakamoto Shuichi , Martens William L. , Suzuki Yôiti
Published: May 2008

In this study, the effect of postural information on the perceived velocity of moving sound sources was investigated. Auditory information was presented via five surround speakers and five subwoofers. Two sounds were presented first in front of participants, and then these moved past them toward the rear. At the same time, rearward pitch motion was presented via a motion platform on which participants were situated. Duration of stimulus was 8 s. Initial distance of sound sources was set to 2.0, 4.0, 6.0, 8.0 and 10.0 m. Amplitude of pitch variation was 0, 0.2, 0.4, 1.0 and 2.0 degrees. Participants made magnitude estimates for speed of moving sound images, and judgment of goodness of movement matching between both auditory and postural information. The resulting speed magnitude estimates showed that pivot magnitude significantly affected the estimated velocity of sound sources except when velocity of sound sources was extremely high. Moreover, participants judged the multimodal match to be poor when the velocity of the moving sound sources was extremely high or low. These results suggest that the strongest multimodal interaction occurs when auditory information and postural variation are well matched, and are consistent with self motion.
Auralization of an orchestra with phase-shifted string sections
Authors: Vigeant Michelle C. , Wang Lily M.
Published: May 2008

Orchestra auralizations have been created in ODEON using multi-channel individual instrument anechoic recordings of two symphonies; however, only one or two string instruments were recorded to represent each string section. To simulate the chorus effect of an entire string section more accurately, the anechoic tracks of the single string instruments have been mixed with other versions of the same signal, each with some phase shift in time. Two groups of phase shifts were used: one with shorter delays of up to 23 ms, and one with longer delays of up to 47 ms. A maximum of seven differently phase-shifted signals were combined with the original to create a final anechoic recording for use in the auralizations, depending on the number of players each source represented. Using paired comparisons, test subjects were asked to identify the auralization that sounded most similar to the experience of listening to an orchestra in an actual concert hall: one having none, short or long phase-shifts. Results show that subjects have difficulty differentiating between these three types of auralizations, indicating that phase shifting may not be required for such multi-source multi-channel orchestra auralizations. [Work supported by the National Science Foundation.]
Why is sharp-limited low-frequency noise extremely annoying
Authors: Krahé Detlef
Published: May 2008

Sharp-limited low-frequency noise having only weak components of higher frequencies, such as noise of an air condition or traffic noise attenuated by a thick window, has a very annoying effect on persons also at low levels. The strong fluctuation, which is specific to this kind of sound, is a frequently used explanation for this effect possibly caused by adaptation in the inner ear. Another or additional explanation could be a strong synchronism in the activities on the nerve fibers. Computer models of the auditory system show this synchronism. If some components at higher frequencies are added, the synchronism disappears and the noise is judged less uncomfortable. This raises the question, if noise protection resulting in a sound as described can not be even counterproductive. Differently sharp-limited sounds are investigated by an auditory model. The results are compared with results of judgment by hearing.
An Equivalent-Acoustic Finite Element Method for Modeling Sound Absorbing Materials
Authors: Nefske Donald J. , Sung Shung H.
Published: May 2008

An equivalent-acoustic finite element method is developed for modeling sound absorbing materials, such as seats and interior trim in the automobile passenger compartment. The equivalent-acoustic method represents the sound absorbing material using acoustic finite elements with frequency-dependent material properties determined from the measured acoustic impedance of sound absorbing material samples. Solution of the equivalent-acoustic model within the Nastran computer capability and coupling of the model with an acoustic finite element model of a surrounding enclosure, such as the passenger compartment, are developed. The accuracy of the equivalent-acoustic method is assessed for modeling a sound absorbing material in a one-dimensional impedance tube, a foam layer in a rectangular box enclosure, and an automotive seat in a semi-reverberant enclosure.
Finite element modelling of transient elastic wave propagation in an inhomogeneous anisotropic fluid/solid multilayer medium: A time-domain method
Authors: Haiat Guillaume , Naili Salah , Grimal Quentin , Talmant Maryline , Desceliers Christophe , Soize Christian
Published: May 2008

The axial transmission technique is used clinically for cortical bone assessment. However, ultrasonic propagation in this multiscale medium remains unclear, in particular because of the heterogeneous nature of cortical bone. The aim of this work is to evaluate the effect of spatial gradients of elastic moduli on the ultrasonic response of the bone structure. Therefore, a 2D finite element time-domain method is developed to simulate transient wave propagation in a three-layer medium constituted of an inhomogeneous transverse isotropic solid layer sandwiched between two acoustic fluid layers and excited by an acoustic linear source located in one fluid layer delivering broadband ultrasonic pulses. The model couples the acoustic propagation in both fluid media with the elastodynamic response of the solid. The conditions of continuity are used to model the fluid-structure interaction. A constant spatial gradient of material properties in the direction perpendicular to the layer is considered in the solid structure. In the presence of a gradient, the first arriving signal (FAS) velocity depends on the average material properties when the thickness is smaller than the wavelength (guided wave modes) whereas the FAS velocity depends on the velocity at the surface when the thickness is larger than the wavelength (lateral wave).
Acoustic frequency wave propagation and fluid-loading in composite sandwich panels
Authors: Crossley Jennifer F. , Sienz Johann
Published: May 2008

Composite sandwich panels comprising glass fibre reinforced epoxy skins with nomex honeycomb cores have found application as aircraft interior fixtures and partitions. This is due to their low density combined with static strength and stiffness. In this complex acoustic and vibration environment however, the same properties can lead to unwanted sound transmission through the structure. We analyse, in this paper, fluid-loading of such a panel by an acoustic fluid using an elementary theory which has been developed in the study of structural dynamics and vibration, based on a generalisation of Timoshenko theory for homogeneous beams. The dispersion relations derived from this theory, with and without fluid-loading terms, are used to quantify the effect of fluid-loading by air on shear and flexural waves at acoustic frequencies in such stiff, lightweight structures. The most appropriate method of fluid-structure coupling to be applied in modelling an internal acoustic field enclosed by such a structure is discussed.
A formulation based on modal optimization for predicting sound radiation from fluid-loaded aircraft structures
Authors: Collery Olivier , Guyader Jean-Louis
Published: May 2008

This study is led in the context of understanding sound transmission through fluid-loaded aircraft structures with non-uniform damping. The fluid/structure coupling is here highlighted. Classic approaches using modal radiation impedances to formulate the fluid/structure coupling do lead to complexity and are computationally time-consuming. A method defining rigorously this coupling and avoiding the calculation of the modal radiation impedances is proposed. This method aims at developing a simple formulation taking advantage of current technical progress in optimization algorithms. Sound radiation from a simply supported, baffled, fluid-loaded plate excited mechanically or acoustically is here solved in optimizing the modal amplitudes so that they fit the governing equation with fluid loading. To perform this optimization, a sampling of the plate into observation points is first done, then a modal decomposition into in vacuo modes is led. Comparison with results from the literature over [10-1000Hz] for 3 reference cases of steel plate immersed in air (critical frequencies equal to fC1=1.2kHz and fC2=12kHz) show excellent agreement within 1dB. The simplicity and computation time allow an extension to non-uniform damped aircraft structures and a prediction over a large frequency band. As perspectives, results from multi-layered plates with a local damping patch are presented.
Use of head mounted microphone arrays for active control
Authors: Johnson Marty
Published: May 2008

There has been an effort recently to create head mounted or user worn microphone arrays. These arrays pose challenges in their design and characterization but can be used for a number of different purposes. Initially these arrays were intended for sound source localization and natural hearing restoration (where the hearing of a user whose ears are obstructed by an encapsulating helmet or headset is restored). Once in place, and assuming the user is wearing headphones, these arrays can be used for a number of different applications including voice isolation for communication, focused listening and noise cancellation. Specifically this paper investigates the use of these arrays for the active control of noise both at the user's ears and in the communications/voice signals sent from the user. A numerical study, using both data generated numerically and experimentally, demonstrates that large reductions in noise can be achieved using adaptive active control methods.
New prospects in implementing distributed control strategies for mechanical structures optimization
Authors: Collet Manuel
Published: May 2008

The research activities in the fields of smart materials and structures today are very important and represent a large potential for the technological innovations. New methods are now available which allow active transducers and their driving electronics to be directly integrated in otherwise passive structures. Today the main research challenge deals with the development of new multi-functional structures integrating their own electro-mechanical controlling systems. In the past few years, a technological revolution has occurred in the fields of integrated MEMS that offers new opportunities for smart structures design and optimization. By using such as integrated active or hybrid distributed set of electromechanical transducers, we could also control the material's intrinsic mechanical behavior for building new desired functionalities. We can also speak of "integrated distributed smart structures". Through two examples, this paper aims at showing what can be the main advantages in developing such of integrated distributed control strategies in comparison with passive and "classical" active systems. The adopted point of view takes also into consideration energy balance assessments, absolute efficiency and of course robustness. The first introduced example deals with the vibroacoustic impedance control of tube for wave power flow cancellation. The second one treats of the optimization of hybrid shunted piezoelectric distributed patches for mechanical power flow control.
Damping of a device with piezoelectric materials and negative capacitance electrical circuits
Authors: Granger Christian S. , Hladky Anne-Christine , Debus Jean-Claude
Published: May 2008

In this paper, the damping of a device is obtained by a transfer of the vibratory energy into electrical energy and then into thermal energy (dissipation in an electrical resistance). The transfer is carried out by using piezoelectric materials (PZT piezoelectric plates, macro fibre composite MFC) and it is improved by charging the piezoelectric materials by an electrical circuit having a negative capacitance impedance. Two devices are considered: a clamped plate, witch is a academic case, and a large aluminium plate (0.85m×0.78m×2mm). The optimal position and the geometry of the ceramics are determined using an analytical method and a numerical method with the help of the ATILA finite element code. The equivalent electrical circuit of the device is conceived. Damping of the device charged by the electrical circuits is measured using a laser vibrometer. Damping can be performed on a given frequency range, covering several bending modes, by using several ceramics on the plate and several external electrical circuits, connected to the active material. Finally, tests in an anechoic chamber show the efficiency of the device for the reduction of the noise transmitted through the plate.
Decentralized control of sound radiation using a high-authority/low-authority control strategy with anisotropic actuators
Authors: Schiller Noah H. , Cabell Randolph H. , Fuller Chris R.
Published: May 2008

This paper describes a combined control strategy designed to reduce sound radiation from stiffened aircraft-style panels. In particular, the control approach uses robust active damping in addition to high-authority LQG control. Active damping is achieved using direct velocity feedback with triangularly shaped strain actuators and point velocity sensors [P. Gardonio and S.J. Elliott, JASA 117(4), 2046-2064 (2005)]. However unlike previous work, anisotropic actuators are used since they outperform traditional isotropic actuators in this application. While active damping is simple and robust, stability is guaranteed at the expense of performance. Therefore, this approach is often referred to as low-authority control. In contrast, LQG control strategies can achieve substantial reductions in sound radiation. Unfortunately, the unmodeled interaction between neighboring control units can destabilize decentralized control systems. Numerical simulations show that combining active damping and decentralized LQG control can be mutually beneficial. In particular, augmenting the in-bandwidth damping supplements the performance of the LQG control strategy and reduces the destabilizing interaction between neighboring control units. Therefore, the performance of the combined system can be better than the sum of each individual strategy.
An impedance matching technique for active-passive vibration control
Authors: Svensson Jonas L. , Andersson Patrik B. , Kropp Wolfgang
Published: May 2008

Impedance matching techniques have shown potential for active vibration control of structures in bending. Such structures are commonly described by Euler-Bernoulli theory. Previous studies concerning impedance matching of these structures have only considered scalar quantities. However, for an Euler-Bernoulli beam four field variables are involved which implies that a scalar impedance is insufficient. The purpose of this study is therefore to expand the technique to include full 2x2 matrices. This is achieved by first deriving the reflection matrix as a function of the characteristic impedance matrices of an Euler-Bernoulli beam and an arbitrary termination impedance. An active impedance load is then introduced in order to manipulate the reflection matrix. A theoretical example is given where the approach is utilized to match the junction between an Euler-Bernoulli beam and a sandwich composite. This proposed active-passive damping configuration employs active control to enclose all incident power in the sandwich composite. Results show that the active impedance load is responsible for the main part of the power absorption over a broad frequency range.
Investigation of the mechanisms of voicing offset
Authors: Barney Anna , Jesus Luis , Santos Ricardo
Published: May 2008

During voiced speech the source of sound production arises from the vibrations of the vocal folds within the larynx. In order to terminate these vibrations it is necessary either to adduct the vocal folds to produce a forced closure or to reduce the pressure drop across them to below a threshold level. A reduction in pressure drop might be achieved by reducing the sub-glottal pressure, by increasing the supra-glottal pressure or by abduction of the folds. Two human subjects were asked to achieve phonation offset by each of these strategies in turn. Acoustic and EGG signals measured on the subjects were compared with the output of a simple theoretical vocal fold model. A further comparison was made with corresponding signals recorded during the production of selected short phrases by a European Portuguese speaker; a language where devoicing of both vowels and voiced consonants is particularly prevalent.
Theoretical and experimental results of phonation threshold pressure vs. oscillation frequency of the vocal folds
Authors: Lucero Jorge C. , Van Hirtum Annemie , Ruty Nicolas , Cisonni Julien , Pelorson Xavier
Published: May 2008

The dynamical principles of the vocal fold oscillation at phonation were set forth by Titze (I. R. Titze, J. Acoust. Soc. Am. 83, 1536-1552 , 1988), by representing motion of the tissues as a surface wave propagating in the direction of the airflow. An important result of his work was an equation for the phonation threshold value of lung pressure, defined as the minimum value required to initiate the vocal fold oscillation. Titze's model assumed a small time delay for the mucosal wave to travel along the vocal folds, with the consequence that the phonation threshold pressure results independent of the oscillation frequency. Here, we consider an extension of his model for an arbitrary time delay. Our results show that the threshold pressure increases with oscillation frequency following a x/sin(x) law. We investigate the validity of the theoretical equation by comparing it with pressure measures from a mechanical replica of the vocal folds, under various configurations. In general, the equation shows good agreement with the experimental data, and may find applications for building empirical relations of glottal aerodynamics, and for clinical studies of phonation. [Work supported by CAPES-Brazil]
Physical modelling of Vowel-Stop-Vowel sequences
Authors: Brutel-Vuilmet Claire , Pelorson Xavier , Van Hirtum Annemie , Fuchs Susanne
Published: May 2008

The physical phenomena involved in the production of vowel sounds are now well described and several effective physical models have been developed. On the contrary, only a few studies concern the physical aspects of the production of consonant sounds (fricatives, stops...) or the production of vowel-consonant-vowel sequences. Indeed, more complex acoustical and aerodynamical mechanisms should be considered (generation of aeroacoustical sources, dynamical description...). In this contribution, we focus on the production of vowel-stop-vowel sequences. A stop sound is produced after a complete occlusion of the vocal tract. In an intervocalic context, the closure entails an increase of the intraoral pressure, which results in the cessation of the vocal folds vibration. The opening is quickly followed by the generation of an aeroacoustical source, and entails a decrease of the intraoral pressure, which results in the resumption of the vocal folds motion. A physical model able to describe these aerodynamical observations and to predict the evolution of the intraoral pressure during the closure gesture is first detailed. Different interesting results are discussed (influence of the closure time, influence of the closure position, coordination of the different sources...). And a confrontation between theoretical results and experimental data is presented.
Analysis of frication noise modulation from a physical model
Authors: Barney Anna , Jackson Philip J.
Published: May 2008

A physical model, built to investigate the aeroacoustic properties of voiced fricative speech, was used to study the amplitude modulation of theturbulence noise it generated. The amplitude and fundamental frequency of glottal vibration, relative positions of the constriction and obstacle, and the flow rate were varied. Measurements were made from pressure taps in the duct wall and the sound pressure at the open end. The high-pass filtered sound pressure was analyzed in terms of the magnitude and phase of the turbulence noise envelope. The magnitude and phase of the observed modulation was related to the upstream pressure. The effects of moving the obstacle with respect to the constriction are reported (representative of the teeth and the tongue in a sibilant fricative respectively). These results contribute to the development of a parametric model of the aeroacoustic interaction of voicing with turbulence noise generation in speech.
Motor regions contribute to speech perception: awareness, adaptation and categorisation
Authors: Davis Matthew H. , Johnsrude Ingrid S. , Hervais-Adelman Alexis G. , Rogers Jack C.
Published: May 2008

Functional imaging and TMS studies show that motor and premotor cortex responds to heard speech though the functional significance of this response is unclear. Three recent fMRI studies, showing modulation of motor responses to heard speech in the absence of overt spoken or manual responses, may shed light on how regions typically associated with speech production contribute to perception. (1) Motor activity remains robust during light anaesthetic sedation, but is obliterated for deeply sedated participants who are no longer aware of speech. (2) Motor cortex responds more to distorted yet intelligible noise-vocoded words than to clear speech or unintelligible noise. This neural correlate of listening effort is also observed for "clear-then-vocoded" presentations that enhance perceptual adaptation compared to a matched "vocoded-then-clear" condition that doesn't enhance adaptation. (3) During paired priming of audio-morphed syllables we see a greater response to acoustic changes that cross phonological category boundaries compared to acoustically-equivalent within-category changes. Additional responses to between-category pairs extend to motor regions, suggesting that neural correlates of categorical perception include regions involved in speech production. These findings illustrate how perceptual awareness, adaptation and categorisation of speech all engage motor regions. Implications for neurobiological accounts of speech perception will be discussed.
From Speech to Language: Mapping the Auditory Comprehension Network
Authors: Obleser Jonas
Published: May 2008

Speech comprehension is a complex perceptual and cognitive task that is fulfilled in a surprisingly robust manner. I will present a series of studies that aim at disentangling the interaction of sensory/acoustic and cognitive/linguistic factors driving the speech comprehension system: (i) How does the system in its entirety deal with noise at the auditory entry level? (ii) Which are key contextual influences that aid speech comprehension when the signal quality drops, and (iii) what is the functional circuitry within and across auditory cortex that copes with comprehension difficulties? Main results include: 1. Comprehension of intermediate signal quality based on semantic predictability engages a left- hemispheric, widely distributed array of brain structures. Also, functional connectivity amongst these areas appears enhanced. 2. Signal intelligibility gates and enables the expected semantic effects (cloze probability; BA44 BOLD response and EEG N400), whereas semantic effects shape the intelligibility modulation in anterolateral temporal cortex. 3. The angular gyrus (BA 39) enhances difficult yet successful speech comprehension whereas left posterior STG reflects higher computational effort (either poor signal quality or low cloze probability) in speech comprehension. Behavioural, EEG and fMRI data will be presented, and themes of (bi-)laterality and bottom-up/top-down directionality will be re-visited along the way.
An fMRI study of subliminal priming of spoken words
Authors: Pallier Christophe , Kouider Sid , De Gardelle Vincent
Published: May 2008

Repetition priming has been widely used to study spoken and written word recognition. Its physiological counterpart is repetition suppression, a reduction in neural activity resulting in a measurable decrease of the fMRI signal. By varying the representational level at which the repetition occurs, one can determine which properties are encoded in a given brain area, and which are not. We will report on an fMRI experiment using subliminal auditory priming of spoken words. Subliminal priming has been used, for example, by Dehaene et al. to study visual word recognition. Our experiment employs a technique developped by Kouider & Dupoux that allows subliminal presentation of auditory stimuli using temporal compression and forward masking. The participants perform a lexical decision task on the target item, which is preceded by a subliminal prime that can be phonetically identical or different from the target, and spoken or not by the same speaker. A fast-event related paradigm is used where each prime-target pair is presented during silent gaps between the acquisitions. The planned analyses will seek to identify the brain regions showing subliminal repetition suppression to word repetition regardless of speaker change, as well as other areas sensitive to speaker change regardless of linguistic content.
The neural bases of normalising for accented speech: A repetition suppression functional magnetic resonance imaging study
Authors: Adank Patti , Hagoort Peter
Published: May 2008

A repetition suppression fMRI paradigm was employed to explore the neuroanatomical substrates of normalisation for accented speech in spoken sentence processing. Sentences were produced in two accents: in Standard Dutch and an artificial accent of Dutch. In the experiment, participants listened to two sentences presented in quick succession. The second sentence was either spoken by the same speaker in the same accent, by the same speaker in a different accent, by a different speaker in the same accent, or by a different speaker in a different accent. This design allowed us to study neural responses to a change in speaker only, a change in accent only and a change in accent and speaker. Results showed small effects for a change of speaker only in right Superior Temporal Gyrus (STG). A change of accent only showed extensive activations in left and right STG and Superior Temporal Sulcus (STS). Finally, a change of speaker and accent showed extensive activations in left and right STG and STS, and increased activity in left Inferior Frontal Gyrus (IFG). The results indicate that normalisation processes for accented speech recruit a wide neural network. The role of left IFG in normalisation processes will be discussed.
Streams of processing and hemispheric asymmetries in speech perception
Authors: Scott Sophie K.
Published: May 2008

Studies in non-human primates have indicated that, as in the visual system, there are (at least) two streams of processing in the auditory system. These pathways are associated with different types of auditory processes - an anterior 'what' pathway and posterior 'how/where' pathway(s). In this talk I will use these neurophysiological theories as a framework for interpreting findings from a range of PET and fMRI studies of human speech perception and production, and present evidence that the anterior 'what' pathway in humans shows hierarchical processing of the speech signal, reflecting a move from acoustic/phonetic processing to a more abstract representation in the anterior superior temporal sulcus. In contrast, posterior auditory areas in humans are associated with sensory/motor interactions in speech, and with aspects of working memory processing. I will address how these systems are differentially recruited when speech perception is made difficult, due to different types of masking noise. Finally, I will outline differences in the processing of speech in left and right auditory areas.
The temporal analysis of spoken language
Authors: Poeppel David
Published: May 2008

The (concurrent) construction of syllabic and phonemic representations forms the basis for creating interpretable representations of speech; therefore we look here to temporal attributes commensurate with their acoustic implementation. Based on a distributed model of the functional anatomy of speech perception (Hickok & Poeppel 2007) and on the assumption that the perception of speech requires multi-time resolution analysis (Poeppel 2003), electrophysiological data are shown that illustrate how auditory cortex makes use of one specific temporal mechanism, the processing of phase (Luo & Poeppel 2007). We hypothesized that the phase pattern of cortical rhythms associated with modulation rates mediating intelligible speech provide an encoding mechanism. We observed that the phase of the theta band response generated in auditory cortex tracks sentence-level acoustics with the sensitivity and specificity necessary for neuronal encoding. The data are consistent with the view that a ̃200 ms temporal window (period of theta oscillation) segments the incoming signal, resetting and sliding to track speech dynamics. This hypothesized mechanism for cortical speech analysis is based on the stimulus-induced modulation of inherent cortical rhythms and provides supporting evidence implicating the syllable as a computational primitive for the representation of spoken language.
ERPs and speech sound perception - possibilities and restrictions
Authors: Peltola Maija S.
Published: May 2008

Speech sound perception is a complex combination of attention independent and attention dependent processes which both contribute to the final goal of understanding the spoken message. The preattentive level has become more accessible to research and event related potentials (ERPs) can easily be used to study the automatic processing of the speech signal. In particular, the mismatch negativity (MMN) response offers a tool for investigating the manner in which speech sounds are encoded as neural representations. Cross-linguistic studies revealing different kinds of representations in native speakers of different languages form the core for further studies, which have shown the plasticity of the brain in forming new representations for non-native sounds in various types of learning environments. However, despite all these promising advances there are still some restrictions connected both with the methodology available and the conclusions that can be reached on the basis of the occasionally contradicting results. Also, since results obtained by using attention independent and dependent methods are not always compatible, some potentially significant results may never reach the attention that they deserve, even if this incompatibility may be one of the keys into the understanding of the complicated mechanisms underlying speech sound perception.
Water column tomographic inversion with a network of drifting buoys
Authors: Jesus Sergio , Soares Cristiano , Martins Nelson
Published: May 2008

The estimation of ocean environmental properties by means of the inversion of acoustic signals has in several occasions been performed using a single vertical array of acoustic receivers, with a towed acoustic source as an attempt to ensure a rapid spatial coverage of the area of interest, as only a single ocean transect is "seen" at each time. Ideally, one would like to obtain an instantaneous picture of the complete area (volume) under observation. However, the resulting acoustic observations, hence environmental estimates, are not simultaneous in time. Using multiple acoustic receiving arrays appears to be a natural step towards both increasing the spatial coverage, and obtaining simultaneous environmental estimates of different ocean transects. It also gives a higher chance to capture spatial transient features, as for example solitons. Using multiple receiver arrays represents the addition of a new spatial dimension at the receiving end and opens up the number of possibilities to a Nx2D or full 3D view of the ocean. Taking support on the data set of the RADAR'07 experiment (July 9 - 16, 2007) where data was simultaneously collected on three vertical arrays, this paper explores space coherent processing of the several receiving arrays and Nx2D or 3D environmental constrained water coloumn matched-field inversion.
Desoloing for musical accompaniment systems
Authors: Raphael Lawrence J.
Published: May 2008

I discuss ongoing work for musical accompaniment systems in which we remove the soloist from a full recording for soloist and accompanying ensemble (e.g. a concerto), leaving just the accompaniment. I will discuss briefly the score matching problem, which generates a correspondence between a symbolic music representation and the audio. Using this score match, straightforward masking leads to usable source separation results, since the nature of the accompaniment problem partially compensates for the damage done by masking. However, I will discuss methods of improving this separation process involving imputing unobserved audio. The presentation will include a live demonstration of the accompaniment system.
Geoacoustic Environment Tracking Using Kalman and Particle Filters
Authors: Gerstoft Peter , Yardim Caglar , Hodgkiss William
Published: May 2008

This paper addresses the problem of tracking the geoacoustic environmental parameters such as the sound speeds in water, sediment and the bottom, sediment attenuation, density and thickness. The tracking is based on using acoustic measurements within an extended Kalman (EKF), unscented Kalman (UKF), and particle filter (PF) framework. The acoustic field is computed using a normal mode code which introduced a varying degree of nonlinearity depending on the environmental parameter of interest. Posterior Cramer-Rao lower bounds (PCRLB) are used to compute the tracking performances of the filters, including the filter efficiencies, divergence statistics, and computational complexities. The results showed that some of the parameters such as the water column parameters can be tracked by Kalman filters, however, the tracking performance of the Kalman filters was limited by the highly nonlinear relation between the sediment/bottom parameters and the acoustic field and non-Gaussian densities of these parameters. Particle filters proved to be very promising in tracking sediment layer parameters, even in the abruptly changing environments.
Efficient Bayesian inference for multiple pitch estimation of music audio
Authors: Vincent Emmanuel
Published: May 2008

Multiple pitch estimation consists of estimating the number of active notes and their fundamental frequencies on each time frame of a music signal. This is a core problem for several applications, including score transcription and source separation. Bayesian harmonic models are a promising approach, since they allow the joint exploitation of various priors on the model parameters. However existing Bayesian inference methods often rely on specific prior distributions and remain computationally demanding for realistic data. We propose a generic inference method based on adaptive factorization of the joint posterior that allows the application of such models to real-world data. We evaluate the results for the task of multiple pitch estimation using different levels of factorization.
The effects of non-linear internal wave curvature on acoustic propagation
Authors: Duda Timothy , Lynch James F. , Lin Ying-Tsong , Newhall Arthur , Graber Hans , Caruso Micheal
Published: May 2008

It is well known that coastal nonlinear internal waves can have curved wavefronts, due to either being generated at a "point source" such as a submarine canyon or valley or due to the wave being refracted by large scale water column or bathymetric features. We will explore the first case here, with the emphasis on circular wave fronts. Three cases of acoustic propagation in the vicinity of a circular wavefront will be considered: 1) the source within a soliton train, 2) the source in front of the soliton train, and 3) the source behind the soliton train. Simple geometric forms are presented showing how the curved wavefronts can lead to mode number and frequency dependent dispersion and shadowing effects. Work sponsored by Office of Naval Research.
Observed acoustic arrival structure and intensity variability induced by transbasin nonlinear internal waves in the South China Sea basin
Authors: Chiu Chingsang , Reeder D. Benjamin , Miller Christopher , Reeves Justin , Ramp Steve , Yang Yiing-Jang , Wei Ruey-Chang , Chen Chifang
Published: May 2008

During the 2005-2006 Windy Island Nonlinear Soliton Experiment (WISE), large-amplitude internal waves were observed by in situ oceanographic sensors to propagate eastward from the Luzon Strait, through the deep basin, onto the shelf and Dongsha Plateau of the South China Sea year round except winter. In a parallel effort to measure the effects of these transbasin internal waves on long-range, low-frequency sound propagation, an acoustic source with a center frequency of 400-Hz and bandwidth of 100-Hz was moored on the west side of the deep basin transmitting a phase-modulated m-sequence signal every 15 min from February to October 2006. These periodic transmissions were recorded by a receiver moored ̃166 km to the east of the source. The recording was processed to give the arrival structure of a pulse and its temporal changes over the eight-month period. The observed temporal variability in the arrival structure and in the acoustic intensity were analyzed using time-series techniques and models with emphasis to elucidate the connection to the observed sound speed variability induced by the nonlinear transbasin internal waves. Results from the analysis are presented and discussed. [The research is sponsored by the US ONR and the Taiwan NSC.]
Temporal correlation of MFP with the presence of internal waves
Authors: Li Zhenglin , Ji Guihua , Dai Qiongxing
Published: May 2008

Internal wave is a dominant source of ocean uncertainties in shallow waters. The ability of passive source localization may be degraded due to mismatch between model predictions and measurements caused by the activities of internal waves. Using ocean environment measurements from an experiment, the effects of Garrett-Munk and solitary internal waves on the temporal correlation of matched-field processing (MFP) in shallow water for sources with different frequencies and different depths are numerically investigated. It is shown that the temporal correlation of MFP decreases as the amplitude of solitons or the average energy density of linear internal waves increased. For a source with lower frequency or located below the thermocline, the temporal correlation of MFP is less affected by internal waves, and the length of which is longer. Moreover, the effects of the range between solitons and source on the temporal correlation of MFP are relatively small.
Modeling probability density functions for acoustic propagation through internal waves in shallow water environments
Authors: Lepage Kevin D.
Published: May 2008

Acoustic propagation through internal waves is shallow water introduces randomness to the acoustic field in that internal waves are generally incompletely measured and are therefore best understood as random realizations of a stochastic process. Due to the physics of internal waves these environmental perturbations are expressed in a finite number of modes, each of which affects acoustic propagation differently. As acoustic propagation in shallow water is itself confined into modes, a matrix of interaction strengths for both the accumulated phase and the mode coupling may be derived which can be integrated forward along the acoustic path to account for the accumulation of uncertainty. A powerful ansatz for solving this problem is a Polynomial Chaos (PC) expansion of the complex modal amplitudes in the random variables which are the internal wave amplitudes. In this work the PC technique is used to derive the probability density functions of the complex modal amplitudes in the presence of a homogeneous internal wave field, showing good agreement with Monte-Carlo results. The pdfs of the complex modal amplitudes can then be combined to obtain the pressure field pdf as well as various statistics of interest such as the scintillation index. [Work supported by ONR]
The underwater acoustic testing of modest sized panel materials using a multi-element array technique in a laboratory test vessel
Authors: Martin Michael J. , Robinson Stephen P. , Smith John , Humphrey Victor F.
Published: May 2008

The underwater acoustic properties of materials can be assessed in panel form via a simple single hydrophone measurement technique which is ideally conducted on large test panels in open water. Alternatively measurements may be conducted in a laboratory test vessel which is capable of simulating ocean conditions. However, physical limitations imposed by the constraints of the vessel, including a reduction in test panel dimensions, require modification of the traditional technique to allow measurements to be made down to low kilohertz frequencies. This can be achieved by, in addition to a directional source, the use of a directional receiver in the form of a planar multi-hydrophone array. The technique is described and illustrated with measurements of reflection loss and insertion loss between 1 kHz and 7 kHz performed on a test panel in a pressure vessel through changing hydrostatic pressure from ambient to 2.8 MPa. The efficacy of this technique is considered with simulated multi-hydrophone array measurements on a scaled panel in an open tank.