bluuebunny/crossref_metadata_2025 huggingface dataset:
Neuroimaging correlates of spatial cues in speech and noise stimuli
Authors: Saberi Kourosh , Smith Kevin R. , Hsieh I-Hui , Hickok Gregory
Published: September 2008

Three fMRI experiments are described that investigated neural activity patterns in the human auditory cortex in response to motion and multi-source cues in speech and noise stimuli. In the first study, ten subjects were scanned while listening to either moving or stationary AM-modulated noise. We found that motion stimuli activated STG/planum temporale (bilaterally) and right parietal lobe, previously implicated in motion processing. Stationary stimuli, whose perceived lateral position varied across presentations, activated these regions equally well. In the second study, we used an event-related fMRI-adaptation paradigm in which pairs of stimuli, consisting of permutations of sequentially presented motion/stationary sounds (m/m, m/s, s/m, s/s), were used during scanning of ten subjects. Consistent with study 1, we observed no motion-selective activation in the planum or parietal regions. In the third study, we examined cortical activity patterns in response to multiple speech streams filtered through individualized HRTFs originating from either a single or multiple locations. Preliminary results show a greater level of cortical activation in ROIs in response to a single speaker, contrasted to multiple speakers. [Work supported by NSF and NIH.]
What is attention, and what is it doing in speech perception?
Authors: Francis Alexander L.
Published: September 2008

The word attention is beginning to appear more frequently in speech research, but closer examination suggests that this term can refer to a variety of potentially unrelated phenomena. The goal of this talk is to begin a discussion of these various phenomena, in the form of a preliminary, and subjective, overview of some areas of speech perception research in which ‘‘attention’’ is starting to play a more prominent role, including experimentally and contextually determined changes in cue weighting, foreign language acquisition, speech perception by elderly and/or hearing-impaired listeners and cochlear implant users, normal and disordered language development, perception of spoken discourse, and multimodal speech perception. I will also discuss some areas of basic attention research in which speech research may be able to make a unique contribution, including theories of the nature and role of auditory ‘‘objects’’ in spatial and object-based attention, load-based theories, and research on the interaction of attention and long- and short-term (working) memory and perceptual learning. [Work supported by NIH-NIDCD R03DC006811.]
Attending to what is said or who said it
Authors: Fenn Kimberly , Nusbaum Howard
Published: September 2008

Research on selective attention in speech perception has examined how listeners process source information or linguistic information. Much of this work has focused on specifying the boundary conditions under which perceptual processing is limited, and how attention to talker and linguistic information can compete for limited processing resources. In general, listeners can only attend to a single stream of speech, and changing the talker increases demands on attention. The studies that support this claim usually use target detection or speeded classification tasks, activities we seldom engage in during normal conversation which is the protypical form of spoken language use. Is attention to speech as limited in real conversations as it is in stylized laboratory tasks? In a phone survey paradigm, participants engaged in naturalistic conversation rarely detect changes in their interlocutor. Furthermore, when participants expect that a change in interlocutor will occur, they largely believe that they have spoken to two different people even when the same person conducted the entire experiment. During a conversation, attention is directed to speech consistent with listening goals, and perception of a talkers voice is not an automatic consequence of attending the linguistic message.
Selective perception, perceptual modes, and automaticity in first- and second-language speech processing
Authors: Strange Winifred
Published: September 2008

A common phenomenon reported by experienced, late learners of a second language (L2) is that comprehending L2 speech, especially under non-optimal conditions (e.g., in noisy rooms, when driving, when the speaker talks rapidly) is more effortful than processing L1 input. A variety of paradigms have documented this phenomenon experimentally. Many current theories of L1 and L2 speech perception invoke concepts of learned (language-specific) patterns of selective attention or attunement to characterize the processes by which native speakers rapidly and efficiently extract sufficient phonetic information from complex acoustic signals in order to recover phonological sequences (words or word-forms). It is suggested here that there are two modes of language-specific speech processing available to adult listeners, a phonetic mode (requiring attentional resources) and a phonological mode (automatic), that are tapped in the laboratory to different degrees as a function of complex interactions of subject’s linguistic experience, stimulus characteristics, and task demands. Exemplary experiments of L1 and L2 listener’s perception, using perceptual assimilation (cross-language identification) and speeded discrimination tasks, as well as electrophysiological indices of discrimination, illustrate some of these interactions, within the framework of the automatic selective perception model of speech perception. [Work supported by NIH-NIDCD and NSF.]
Levels of selective attention in speech perception
Authors: Francis Alexander L.
Published: September 2008

Listeners’ ability to attend to a specific talker while ignoring simultaneous speech from another talker (the ‘‘cocktail party’’ problem) is not yet well understood. Lavie’s load theory [Trends Cogn. Sci. 9(2), 2005], originally developed in the visual modality, conceptualizes selective attention as a limited-capacity resource and provides a method for investigating levels of attentional operation. Selection may arise at a cognitive level, when intruding irrelevant information is explicitly inhibited, or at a perceptual level when limited capacity is overwhelmed and task-irrelevant information cannot be fully processed. Under high cognitive load, insufficient processing capacity is available to reject intruding distractors, and interference increases. Under high perceptual load, irrelevant stimuli intrude less because fewer processing resources are concentrated on them, decreasing interference. The present study evaluates the predictions of this theory for speech perception. Listeners identified words produced by a central talker while ignoring a peripheral one. Perceptual load was manipulated by presenting a concurrent (non-speech) response cue, with response contingent upon a single feature of the cue (low load) or a conjunction of features (high load). Cognitive load was manipulated using a concurrent digit pre-load task with one (low load) or six digits (high load). [Work supported by NIDCD No. R03DC006811.]
Phonetic cue use and integrality
Authors: McGuire Grant
Published: September 2008

Phonetic categories are generally assumed to have multiple cues available, which listeners can reliably use for identification. Moreover, there is considerable evidence that such cues are perceived integrally as a unit [Best et al. (1981), Kemler Nelson (1993)]. This paper presents results from two experiments exploring cue perception by comparing Mandarin and American English listeners’ perception of the Polish alveopalatal ∼ retroflex sibilant contrast. This contrast is native to Mandarin listeners [Stevens et al. (2004)] and is difficult for English listeners to discriminate [Lisker (2001)]. In the first experiment, listeners discriminated pairs of syllables having fully correlated cues, pairs differing in a single dimension (fricative noise, formant transition), or fully conflicting cue pairs. Results demonstrate that only Mandarin listeners are negatively affected by conflicting cue information. In a second experiment, the same listeners labeled a two-dimensional continuum varying in fricative noise and formant transition cues. Mandarin listeners reliably used both cues and showed an effect of integration. English listeners relied solely on vocalic information and ignored fricative variation. Together these results provide more evidence that cue use is language specific and suggests that listeners do not automatically generalize cue information used in other contrasts to a new context.
Acoustic measures of sound change in language obsolescence
Authors: Chang Charles B.
Published: September 2008

Following recent acoustic investigations of obsolescing languages such as Babel (2007), I present an acoustic study of phonetic change in Southeastern Pomo (Northern Hokan, Pomoan) based upon recordings of two generations of speakers. This paper focuses on the realization of two phonological contrasts: A velar/post-velar contrast and a dental/alveolar contrast. First, acoustic measurements show that both contrasts have undergone phonetic change convergent with English, the dominant contact language, which contains neither contrast. The contrast between velars and post-velars is considerably diminished in Generation 2 with respect to Generation 1, as seen in the significantly smaller difference between the second formant resonances of the velar and post-velar fricatives. The dental/alveolar contrast undergoes a similar change: The difference between the burst peak frequencies of the dental and alveolar ejectives and the difference between the third formant resonances of the following vowels both shrink. However, the data also indicate that although the acoustic distance between the members of each contrast decreases, the contrast between the two is still maintained in one or more of the above acoustic dimensions. Thus, this study provides a demonstration of how acoustic data can be used to illuminate the status of a sound change in progress.
Perception of phonemically-identical true prefixes and pseudo-prefixes in real words heard in noise
Authors: Baker Rachel , Hawkins Sarah , Smith Rachel
Published: September 2008

This experiment assessed the perceptual salience of acoustic-phonetic differences measured in the first syllables of words with initial syllables that differ in morphological, but not phonemic structure [Baker et al., Proc. 16th ICPhS (2007)]. Words began with true prefixes (Pr: e.g., mistimes, displease) or pseudo-prefixes (Ps: e.g., mistakes, displays). The fourth phoneme was a stop. A woman recorded sentences, identical except for the critical word, e.g., I think she distrusts/destroyed them, in scripted dialogs for naturalness. Critical syllables (mis-, dis-), up to but excluding the stop burst, were cross-spliced either into the ‘‘wrong’’ context (mismatched conditions, PrPsPr, PsPrPs) or into another instance of the same sentence (controls, PrPrPr, PsPsPs). In a counterbalanced design, four groups of ten listeners each heard eight mismatched and eight control sentences in cafeteria noise (SNR 2 dB), and wrote what they heard. Errors were greater for critical words in mismatched than control sentences, though the difference only achieved significance for dis- words. The data suggest that perceptual disruptions reflect the difference between the spectrotemporal pattern heard and that expected on the basis of the critical syllable’s morphological status (true- or pseudo-prefix), the sentence prosody, and possibly the statistical distribution of the critical syllable’s duration.
Perception of word boundaries in English speech by Japanese L2 learners of English
Authors: Ito Kikuyo , Strange Winifred
Published: September 2008

This study investigated the perception of aspiration and glottal stop phonetic cues for word juncture in English phrases by Japanese L2 learners of English, replicating a study of Spanish L2 learners [E. P. Altenberg, Second Language Research. 21, 325–358 (2005)]. Thirty Japanese and 10 American listeners responded to 84 phrases in randomly ordered lists containing 42 contrasting pairs, e.g., keeps talking versus keep stalking (aspiration-cued contrasts) and a nice man versus an ice man (glottal stop-cued contrasts) in a two-choice task with confidence ratings. Results indicated that aspiration cues were less available to Japanese listeners (mean=73% correct) than glottal stop cues (mean=91% correct); Americans performed near ceiling on both types of contrasts (96% and 97% correct). Japanese listeners had the most difficulty perceiving aspiration-cued contrasts containing consonant clusters at word junctures, e.g., keeps canning versus keep scanning, tops crawled versus top scrawled. JapaneseEconfidence ratings were high even when they erred, while those of native listeners better reflected response accuracy. The results support the notion that L1/L2 differences in both phonetic and phonotactic structure significantly affects perception of L2 word boundaries.
Effects of syllable-final duration variation on the identification of synthetic /ba/-/wa/ and /bas/-/was/ speech continua by humans and birds
Authors: Welch Thomas E. , Dent Micheal L. , Sawusch James R.
Published: September 2008

In an attempt to test whether experience with or knowledge of language is necessary to show typical human only duration-based tendencies in speech perception tasks, budgerigars (Melopsittacus undulatus) and zebra finches (Taeniopygia guttata) were trained to categorize the endpoint stimuli of four synthetic continua consisting of CV and CVC stimuli, with both short and long syllable-final durations. Humans and birds were then tested on the exact same sets of full ten-step continua and their performance was compared in terms of their identification responses and of phoneme boundaries along the /b/-/w/ stop-glide continua. Operant conditioning was used to train the birds to respond differentially to /b/ and /w/ stimuli. Overall, zebra finches, who unlike budgerigars, cannot mimic human speech, were poor at categorizing the human speech tokens and very few could even categorize two endpoints above criterion levels to be moved onto testing. The birds that were able to learn the task showed similar identification results to humans for the continua. The implications of these results and comparisons of the effects of phoneme duration (as in human speaking rate) between humans and birds will be discussed. For Speech Communication Best Student Paper Award Best ‘‘Outstanding Paper By A Young Presenter’’ Awards.
Characteristics of /h/ in American English: Effects of speaker and phonetic context
Authors: Koenig Laura
Published: September 2008

This work explores the factors associated with variation in /h/ realization in American English, with particular focus on the presence or absence of phonation. Traditional phonetic accounts have suggested that voiced /h/ occurs most commonly in unstressed, intervocalic positions, but limited quantitative data exist. In an earlier study, we assessed /h/ production in the Texas Instruments/MIT [TIMIT] database, considering one utterance produced by all speakers (N=630), along with a few additional utterances. Those results suggested that /h/ voicing varied considerably depending on phonetic context. This study extends that work by measuring all /h/-containing utterances in the TIMIT database that were produced by multiple speakers. Each possible /h/ was inspected to determine whether the /h/ was deleted, showed visible noise, and/or had a phonation break (defined for each speaker as more than the duration of two pitch periods as measured in an adjacent stressed vowel). The utterances were then coded for position (utterance-initial, word-initial, intervocalic); preceding and following phonemes; stress (full or reduced vowel following); and speaker (gender and dialect region). These data will provide a more complete picture of the environments that favor phonation and aspiration noise in an open vocal tract in running speech.
Sub-bottom profiling using time-frequency analysis
Authors: Sanders William , Bibee Dale , Kaminsky Edit
Published: September 2008

Methods for characterizing the returns from acoustic scatterers can roughly be separated into spectral or imaging methods. If the wavelength of the acoustic signal is small compared to the size of the target (geometrical scattering), the object may be imaged to such a fine resolution that its shape or structure may be revealed. This is often achieved using a wide bandwidth (high temporal resolution) signal. Otherwise, if the wavelength of the acoustic signal is large compared to the target size, spectral methods are deemed superior. These utilize relatively narrowband methods (with good spectral resolution) designed to identify characteristic resonances or nulls in spectra. However, practical factors place sub-bottom profiling systems squarely in the middle of these two domains. Hence time-frequency analysis becomes a viable option for characterization of scattering from buried objects. This study utilized simulated and tank-collected data to demonstrate the utility of Wigner-Ville distributions in analyzing and characterizing specific acoustic interactions. Salient features of scattering are revealed in the Wigner-Ville distributions that are not easily identifiable in either time domain or frequency domain only analysis. [This work was supported in part by the Naval Research Laboratory under program element PE62782N.]
High frequency scattering from water-saturated granular sediments: Scaling effects
Authors: Ivakin Anatoliy , Sessarego Jean-Pierre
Published: September 2008

Sound backscattering from water-saturated granular sediments at frequencies from 150 kHz to 8 MHz at normal and oblique incidence was studied in controlled laboratory conditions. Two kinds of sediments, moderately sorted medium and coarse sands, were chosen for the study, degassed, and their surface was flattened. At oblique incidence, intrinsic scattering due to the sediment granular structure can be considered as a dominating mechanism of backscatter. Comparison of frequency dependencies of the backscattering strength for the two sediments with different mean grain size shows the existence of a persistent scaling effect that allows a description of the backscattering strength as a unique scaling function of one parameter, the mean grain size/wavelength ratio. For a wide range of this ratio, 0.15 to 1.3, the scaling function has been measured and presented, which is the major result of this work. Possible applications to remote sensing of marine sediments, and particularly determining the sediment mean grain size, are discussed. [Work supported by ONR and CNRS.]
Comparison of finite element model of scattering with experimental results from the Experimental Validation of Acoustic modeling techniques (EVA) sea test
Authors: Isakson Marcia , Chotiros Nicholas , Piper James , Zampolli Mario
Published: September 2008

Reflection coefficient measurements from 5 50 kHz at grazing angles from 10–80 degrees were taken as part of the Experiments for Validation of Acoustic modeling techniques (EVA) sea test on the north shore of Isola dElba, Italy. The microtopography of the seafloor was also measured using an ROV mounted laser line scan system. The seafloor roughness measurements were used as inputs in a finite element model to calculate the effects of scattering on the reflection coefficient measurement. The variance of the calculated reflection coefficient is compared with the measured variance to determine the contribution from interface scattering at different frequencies and angles. [Work supported by ONR, Ocean Acoustics.]
Extension of the grain-shearing theory of wave propagation in marine sediments to include pore-fluid viscosity
Authors: Buckingham Michael J.
Published: September 2008

Buckingham’s grain-shearing (GS) theory of wave propagation in saturated granular materials predicts a sound speed showing weak dispersion and an attenuation that scales essentially linearly with the frequency. Measurements of the dispersion curves made during the SAX99 experiment in the Gulf of Mexico match the GS theory at higher frequencies, above 10 kHz, but show a lower sound speed and a higher attenuation than predicted by the GS theory at lower frequencies. A generalized version of the GS theory has now been developed, designated the VGS theory, which takes into account the effective viscosity of the molecularly thin layer of pore fluid between contiguous grains. In effect, the viscosity of the pore fluid limits the degree strain-hardening that occurs as grains slide against one another. The resultant VGS dispersion curves are the same as those of the GS theory, except for the appearance of a simple algebraic function that has an effect only at frequencies below about 10 kHz. This low-frequency function gives rise to VGS compressional-wave dispersion curves that match the sound speed and attenuation measurements made during SAX99 across the frequency band from 1 kHz to 400 kHz. [Research supported by ONR.]
A finite-element analysis of reflection coefficient variability due to interface roughness
Authors: Yarbrough R. Abe , Isakson Marcia J.
Published: September 2008

The effect of sea floor roughness on the reflection of sound diverging from a spherical source is investigated. An ensemble of water-sediment interface realizations are modeled, each with height statistics matching those of a rough surface prepared in a laboratory experiment. Using the finite-element method to solve the Helmholtz equation, time harmonic solutions from 40 to 60 kHz are calculated. From these frequency domain solutions, the band limited impulse response time domain solution is constructed via Fourier synthesis. The first two moments of the spherical wave reflection coefficient are determined as a function of grazing angle, and results are compared to measurements from the laboratory experiment. [Work supported by ONR, Ocean Acoustics.]
Application of generalized polynomial chaos method to describe uncertainty in acoustic scattering from random rough interfaces
Authors: Keiffer Richard
Published: September 2008

Recently, the transformed field expansion (TFE) boundary perturbation method [D. P. Nicholls and F. Reitich, J. Comput. Phys. 170, 276–298 (2001)] has been extended to random rough interfaces using the method of generalized polynomial chaos (gPC) [D. Xiu and J. Shen, Comm. in Comp. Physics 2, 54–72 (2007)] to compute statistics of the scattered field. The TFE boundary perturbation method is similar to the usual small wave height approximation approach, but starts with a coordinate transformation that avoids ill-conditioning in the perturbation series and maps the original deterministic Helmholtz equation on a random domain to a stochastic Helmholtz equation on a deterministic domain. When dealing with large multiscale rough interfaces like the ocean surface, the number of expansion terms in the gPC representation of the field can grow large. One strategy to keep the number of random variables manageable is to pose the scattering problem in the time domain, allow for the excitation of the rough interface by an impulse, and consider only those random surface features that can be causally linked to the received field. [This work was supported by ONR through NRL.]
An experimental test of a model for scattered envelope statistics
Authors: Lyons Anthony P. , Parks Susan E. , Weber Thomas C.
Published: September 2008

Co-located acoustic and high-resolution environmental data were collected at Seneca Lake in the summers of 2005 and 2006. The first acoustic data set was acquired in August of 2005 near Dresden, New York, over an approximately 5 square km section of Seneca Lake near the Naval Undersea Warfare Center’s System Measurement Platform and consisted of large numbers of high bandwidth high-frequency pings (tens of kHz). Analysis of the acoustic data showed areas of distinctly nonRayleigh statistics. In mid-July, 2006, high-resolution environmental ground truth data were collected including bathymetry and scattering strength data obtained using a RESON 8101 SeaBat 240 kHz calibrated multibeam system, video from ROV, and sediment grab samples. The ground truth data showed the dominant scattering mechanism to be large numbers of Quagga mussels. The spatial distribution of the mussels can be used to predict the broadband statistics using a recently developed model [D. A. Abraham and A. P. Lyons, IEEE J. Ocean. Eng., 27, 800–813, (2002)], providing an excellent opportunity to test the model. Results and model/data comparisons are presented in this paper. [Work performed under ONR Grant Nos. N00014-04-1-0013 and N00014-06-1-0245.]
Sound fluctuation in a non-Gaussian random field: A wavelet-based Wiener chaos expansion approach
Authors: Xing Hao , Ma Li
Published: September 2008

Recently Wiener chaos derived stochastic expansion methods, especially polynomial chaos (PC), have caught great attention in the underwater acoustics community for its success in modeling random properties of underwater acoustic field. However, most research until now concern a Gaussian random field, while fluctuations in ocean medium are not always Gaussian. For non-Gaussian probability density function (PDF) with a finite support, it is reported that a Wiener chaos decomposition, which use wavelets as basis functions instead of orthogonal polynomials, is more suitable than PC. This paper concentrates on non-Gaussian high frequency internal waves [T. Wang and T. Gao, Chin. J. Ocean. Limnol. 20, 16–21 (2002)] and presents a multi-resolution stochastic expansion approach to handle acoustic field fluctuation in a non-Gaussian internal wave field. The depth dependent PDF of sound speed fluctuation is determined empirically from experiment results. Spatial coherence function and scintillation index of acoustic field were calculated for Gaussian and non-Gaussian cases. In addition, a comparison between both situations was made, which shows that in the non-Gaussian case, the acoustic field has a more complex coherence structure. [Work partially supported by NSFC.]
Sound transmission loss of nontraditional building materials and redundancies
Authors: Ermann Michael , Carneal James , Mennitt Daniel , Jackson Christopher , Karmarkar Bharti , Helveston Matthew , Clay Patrick
Published: September 2008

This line of inquiry aims to provide sound transmission loss data, both lab- and field-tested, for novel transparent and translucent wall assemblies. Special attention is paid to: (1) Redundant assemblies featuring two layers of material with a large gap or small room between them, (2) assemblies featuring emerging materials, and (3) assemblies with high thermal resistance values. To that end, composite configurations of acrylic, polycarbonate, aerogel, and glass were tested in a laboratory, and an acrylic room and glass vestibule were tested in the field. Results will be presented with an eye toward future use of the data. Indeed, it was the unavailability of the data in existing publications that initially drove the research.
The acoustics and psychoacoustics of sound localization in the cat
Authors: Tollin Daniel
Published: September 2008

Cats are a common model system for the anatomy, physiology, and psychophysics of sound localization. There are three main cues to location: Interaural differences in time (ITD) and level (ILD) and monaural spectral shape. Here, the cues were derived from measurements of directional transfer functions, the directional components of head-related transfer functions. In the frontal hemisphere, spectral notches were present for frequencies from ∼8–16 kHz; the notch frequency increased with increases in source elevation and in azimuth towards the ipsilateral ear. The maximum ITDs were ∼380 s. Maximum ILDs depended on source azimuth and frequency, but were as large as 35 dB at some frequencies. For the psychophysics of localization, cats were trained using operant conditioning to indicate the apparent locations of sounds via gaze shift (combined eye and head movements); eye and head position were measured using the search coil technique. Auditory targets consisted of noise bursts of varying duration, intensities, and spectral content. Localization of long duration, moderately intense, and broadband targets were accurate and precise, but deteriorated for sounds that were short-duration, narrow-band, or that were very intense or faint. Cat sound localization capabilities are in many ways similar to humans. [NIH grants DC006865 and DC02840.]
Neural computations in the owl’s sound localization
Authors: Pena Jose L.
Published: September 2008

Neural models predict mathematical operations taking place in systems of neurons. The plausibility of such operations is a matter of debate. The owl’s auditory system offers a good substrate to approach this question. To determine what and where acoustic objects are, the brain must compare and combine noisy information coming from each ear. Comparison, combination, and noise reduction can be represented by equations. A common tool used to extract signals from variable sets of stimuli is averaging. Similarly, cross-correlation and multiplication are mathematical correlates of comparison and combination selectivity, respectively. In the owl’s auditory pathway, we can find neurons that effectively perform these types of processing. The nucleus Laminaris neurons perform a running cross-correlation that indicates the degree of similarity between right- and left-side inputs. Midbrain neurons become selective to combinations of spatial cues by effectively multiplying inputs from independent processing pathways. In the inferior colliculus, an average-like process produces a tuning to interaural time difference that would require at least five repetitions of the same stimulus one stage before. Thus, the owl’s auditory system appears to perform the computations that models predict, representing an interesting example of how close to the nature of neural processing mathematical operations are.
The building blocks of ranging: What laboratory studies of distance perception can tell us
Authors: Phillmore Leslie , Sturdy Christopher , Weisman Ronald
Published: September 2008

Ranging (estimating the distance to) conspecifics is an important skill for songbirds. For example, territorial males must determine, often by acoustic information alone, whether a rival male is either within territorial boundaries, requiring an aggressive response or outside boundaries, requiring him to withhold response to conserve energy. Most field studies of ranging in songbirds measure males’ behavioral responses to playback of vocalizations seeming to originate from within or outside an established territory. However, this approach is inadequate for investigating ranging abilities in species that do not defend territories or for assessing the extent to which ranging abilities are dependent on early experience with distance. In a series of studies, a go/no-go operant task requiring birds to discriminate vocalizations recorded at various distances was developed to ask some of these comparative questions about auditory distance perception. Results showed that a territorial species, black-capped chickadees (Poecile atricapillus), learned to discriminate more quickly than a nonterritorial species, zebra finches (Taeniopygia guttata), that both species learned to discriminate chickadee vocalizations more quickly, and that chickadees raised without experience with ranging could perform the distance cue discrimination task as well as field-reared birds. [Research completed at Queens University, Kingston, Canada; supported by NSERC.]
High-resolution marine seismic reflection profiling: Systems and capabilities
Authors: Hart Patrick E.
Published: September 2008

Seismic reflection profiling is the most effective and commonly used geophysical tool for sub-seafloor geologic investigations. A broad variety of seismic data acquisition systems are available, depending on the imaging depth and resolution required. High-resolution methods utilize controlled sound sources that generate signals in the frequency range of approximately 1,000 to 10,000 Hz. Data acquired with piezoelectric swept-frequency (chirp) transducers, mini-sparkers, and boomer-plate sources routinely image up to 100 m subbottom in marine sediments, with vertical resolution better than 50 cm. Small air-gun systems with frequencies between 100 Hz and 500 Hz achieve as much as 1 km of penetration and resolution of a few meters. Imaging depths are highly dependent on the geologic environment; surveys in fine-grained marine sediments usually produce the best results. Free gas from decaying organic matter in shallow sediments can dramatically degrade data quality. High-resolution methods are contrasted by current hydrocarbon exploration industry systems that utilize large arrays of air guns to generate high-energy signals between l0 Hz and 100 Hz recorded by multiple hydrophone streamers as much as 10 km in length. The resulting three-dimensional data volumes can provide detailed images to depths greater than 15 km.
Prestack full waveform inversion for estimating elastic properties of marine sediments
Authors: Sen Mrinal
Published: September 2008

Prestack seismic waveform inversion is a highly challenging task. The nonlinearity and nonuniqueness together with computed intensive forward modeling make the problem intractable. Here we report on a new approach to seismic waveform inversion based on a gradient descent scheme that makes use of innovative gradient calculation, adaptive regularization, and an efficient conjugate gradient algorithm. We exploit sparsity of the gradient matrix based on physics in designing an algorithm that is computationally very efficient. The results from our approach are highly encouraging in that we are able to invert large 3-D dataset with nearly 1,000 model parameters fairly rapidly. Not only that, because of the use of adaptive regularization, we are able to track smooth as well as sharp variations in the impedance, Vp and Vs profiles when realistic starting solutions are used. We have applied the technique successfully to numerous 3-D exploration and academic datasets. Of particular interest is our ability to estimate density from seismic data that can be used to map porosity in an effective manner. Application to seismic data collected over offshore Oregon resulted in estimates of Vp and Vs that could be used in estimating hydrate concentration in the shallow sediment.
Performance of a chirp sonar sub-bottom profiler for different bottom types
Authors: Turgut Altan
Published: September 2008

Performance of a 0.5–12 KHz chirp sonar sub-bottom profiler is evaluated for different bottom types. The bottom types at different survey sites include mud (Malta Plateau and Gulf of Mexico), silt (Santa Barbara Channel), sand (New Jersey Shelf), and mud stone (Oregon Shelf). Excellent vertical resolution and penetration revealed several geological features (e.g., buried river channels, mud slides, seep alterations, etc.) at muddy, silty, and sandy sites. Sub-bottom imaging at the Oregon Shelf site was relatively poor due to insufficient penetration and interference from sidelobes over a rough terrain of mud-stone bottom. Several improvements on the instrument reponse and source-receiver beam patterns are discussed. Methods for estimation of bottom types and measurements of reflection and attenuation coefficients are also presented. [Work supported by ONR.]
Infrasonic instrumentation
Authors: Rasmussen Gunnar
Published: September 2008

Infrasonic measurements and calibration. Infrasonic measurements call for special instrumentation in order to cover the low frequency range. Below 100 Hz the microphones present a special problem. The microphone performance is very different between free field conditions with wave motion and closed conditions without propagating waves. The primary method for pressure calibration of standard microphones by the reciprocity technique described in IEC 1094-2 assumes that the microphone is only exposed to the pressure variation on the front of the diaphragm. A true pressure calibration will therefore present an increase in sensitivity below the air equalization time constant. In a free field the sound pressure has access to the back of the diaphragm through the air equalization vent and the sensitivity of the microphone will go towards zero at low frequencies. The uncertainty therefore at low frequencies goes towards infinity, when standards for free field sensitivity by comparison with a laboratory standard microphone or working standard microphone as presented in IEC 61094, part 3-7-8, are applied. A low frequency calibrator which will present a constant pressure, independent of frequency and adiabatic-isothermal conditions at frequencies lower than 250 Hz, will be discussed.
Comparison of measurement and theory for the acoustic pressure field in an infrasonic calibrator through the isentropic-to-isothermal transition
Authors: Marston Timothy M. , Gabrielson Thomas B.
Published: September 2008

In a previous paper [T. M. Marston and T. B. Gabrielson, J. Acoust. Soc. Am. 119, 3378 (2006)], a simple total-immersion calibrator for infrasonic microphones was described. This calibrator has been improved and so has the correspondence with theory. The Rott equations for thermoviscous acoustics have been incorporated in a stepwise solution for the acoustic field. These equations account for the transition between boundary-layer flow in which the main field is almost isentropic, and the very-low-frequency approach to isothermal pressure oscillations. The entire microphone body is placed in the calibrator so that the pressure-equalization leak functions normally. The calibrator operates from 0.02 Hz to 20 Hz and includes a DC-coupled piezoresistive differential pressure sensor for comparison calibration; however, the model, which incorporates the geometry of the calibrator pipe, the temperature, and the piston stroke, permits response estimation without dependence on the reference sensor. [Research funded by the Applied Research Laboratory Educational and Foundational Fund.]
Acoustical properties of porous hose wind noise filters
Authors: Howard Wheeler , Dillion Kevin , Shields F. Douglas
Published: September 2008

Observation of infrasound signals in the atmosphere is often masked by wind noise. A common means of filtering out the wind noise is to connect a commercial porous (or soaker) hose to the manifold of a micro barometer. The filtering effect is attributed to the ability of the porous hose to average the pressure variations over its length. The pressure variations due to the turbulent wind field are incoherent on a scale equal to the hose length and, therefore, are reduced in the averaging process. Infrasound signals, with wavelengths much longer than the hose, are reduced little in the averaging process. There remains the question, ‘‘How does the porous hose respond to infrasound signals that have wavelengths comparable to the hose length?’’ To answer this question, measurements have been made of infrasound signals radiated from a jet engine during takeoff using an infrasound sensor with a porous hose connected to its port, and an array of infrasound sensors distributed along the length of the porous hose. Measurements are reported for the hose oriented both in the direction and perpendicular to the direction of the sound source. [Work supported by the Space and Missile Defense Command is gratefully acknowledged.]
Simple calculation models for wind noise spectral contributions in outdoor turbulent flows
Authors: Yu Jiao , Webster Jeremy , Raspet Richard
Published: September 2008

We have previously reported on the predictions of the stagnation pressure fluctuations, turbulence-turbulence interaction pressure fluctuations and turbulence-mean shear interaction pressure fluctuations in outdoor turbulent flows [J. Acoust. Soc. Am. 120(5), 3032 (2006)]. The predictions of these contributions to the full pressure spectrum involve double integrals and are computationally complex. In this talk, we present simple algebraic fits to the predictions, and use these fits to discuss dominant terms under atmospheric conditions different from the measurement conditions reported earlier. Turbulence percentage and scale are shown to be the determining factor influencing the relative contributions to the wind noise outdoors. The simple calculation models provide a quick way to predict wind noise contributions from wind velocity measurements. [Research supported by the U.S. Army TACOM-ARDEC at Picatinny Arsenal, NJ.]
An investigation of the correlation of infrasound signals as a function of sensor separation and wind velocity
Authors: Howard Wheeler , Dillion Kevin , Shields F. Douglas , Hetzer Claus
Published: September 2008

Wind noise is a common obstacle to the detection and analysis of infrasonic signals. The masking caused by the wind noise is often reduced by spatial averaging via pipe or porous hose arrays. An alternate method is to average the signals from multiple sensors in distributed arrays. In order to reduce the wind noise, the sensors in the distributed array must be far enough apart so that their wind noise signals are incoherent, but not so far apart that the sound loses its coherence. Few measurements have been reported of the infrasound correlation distance in the presence of wind. This paper reports results of measurements of this infrasound correlation when the infrasound sensors are separated by distances up to hundreds of meters. Techniques for combining infrasound signals from widely separated sensors and measuring their correlation in the presence of wind noise are discussed.
Microphone array images of low-frequency wind noise
Authors: Wilson D. Keith , Greenfield Roy , White Michael J.
Published: September 2008

Transient, propagating pressure disturbances associated with turbulent eddies (wind noise) and sound waves are imaged by wavelet processing of data from a horizontal, 49-microphone array. It is found that the choice of analyzing wavelet has a strong effect on the appearance of the pressure images; because of the rapid spectral roll-off of the wind noise, there exist important trade-offs between temporal/spatial resolution and frequency resolution. For a given time scale, the wind disturbances are much smaller and less spatially coherent than sound waves. Since the wind disturbances move much more slowly than the sound waves, the spatial pattern they exhibit is independent of the look direction of a conventional acoustic beamformer. The wind disturbances also do not resemble plane waves. Conventional beamforming techniques are particularly sensitive to wind noise when sensor separations are small compared to the acoustic wavelengths of interest.
My voice looks like that? Analysis and synthesis of vowel sounds in a PC-based lab
Authors: Piacsek Andrew A.
Published: September 2008

How is it that we can recognize familiar vowel sounds regardless of who is speaking? How does my voice compare to your voice? The acoustics laboratory exercise described here appeals to our natural curiosity about everyday phenomena and about ourselves. In the first part, students use a PC equipped with a built-in microphone and easy-to-use sound analysis software to record themselves ‘‘singing’’ a variety of vowel sounds, each with the same steady pitch. With the spectrum of each vowel displayed, students initially confirm the presence of the harmonic series and identify the fundamental frequency, reinforcing the association of these characteristics with the pitch of the vowel sound. Working in small groups, students then investigate a correlation between the specific vowel and its spectrum envelope. By comparing the spectra of similar vowels sung by different students, as well as the spectra of different vowels, students learn to recognize formants within the spectrum. In the second part, students apply their newfound understanding of the role of formants by attempting to synthesize a vowel sound by adding harmonic components. The sound synthesis is also done with commonly available PC software.
Supervised separation of musical sources
Authors: Smaragdis Paris
Published: September 2008

Sounds, especially in the musical context, are often presented in mixtures. Given that traditional signal processing theory is not well equipped to work with concurrent sound sources, this limits the analysis and creative options on musical audio. As it was recently shown, sound mixtures can be analyzed with latent variable models of time-frequency distributions and, thus, reveal the additive structure of acoustic scenes. Unfortunately, such processes separate acoustic elements in an indiscriminant manner that does not always result in extracting the desired sources. However, in the case where a user can provide examples of the sources to extract, the aforementioned algorithms can become powerful tools for supervised source separation. Performing experiments on real single-channel recordings, it was shown that by simply indicating the regions of time where a particular source was active, it is possible to extract that source with minimal distortion, even though it may constantly be part of a sound mixture. Since this is an example-based procedure, it is able to deal with arbitrary sources, regardless of their acoustic properties, without requiring any tedious heuristic modeling. This extraction method facilitates the analysis of musical data, and also allows creative manipulation of music.
Methods for stereo music source separation
Authors: Ehmann Andreas
Published: September 2008

Musical source separation from stereo signals has been growing in popularity recently. Algorithms (including the author’s) and the results of a recent stereo audio source separation evaluation campaign (SASSEC) will be presented. In general, such approaches involve extracting mixing parameters from time-frequency (T-F) analyses of both channels. Each corresponding T-F point of the two channels is compared and interchannel intensity differences (IID) and (in the case of stereo mixing), interchannel phase differences (IPD) are computed. A histogram of signal power in the IID and IPD space is constructed, with sources showing strong peaks at the locations of their mixing parameters. Masks are then created for each potential source, corresponding to T-F points that share common mixing parameters. These masks are applied to the T-F analysis and resynthesized as probable sources. The underlying assumption of such systems is that of W-disjoint orthogonality, that is, T-F points of the sources do not overlap. In music, however, this is rarely the case, as common musical intervals often contain overalapping harmonic ‘‘collisions.’’ The use of an exponentially damped sinusoidal model, that in limited cases has the ability to resolve very closely spaced, beating, sinusoids is explored as a means of separating harmonic collisions.
Approaching polyphonic transcription of piano sounds
Authors: Ortiz-Berenguer Luis I. , Casajus-Quiros Francisco J.
Published: September 2008

Polyphonic transcription of piano is a challenging task due to the specific characteristics of its sound spectrum. Pattern matching method, which compares the spectrum with a set of spectral patterns, has proven to give good results, although some limitations still exist mainly when analyzing notes of the lower octaves. Present research is oriented to improve both the accuracy of the spectral patterns, especially for lower octaves, and the matching-score calculation. The former is being carried out by developing a model of the vibration of piano. The last is aimed to decide whether a simple internal product or a more complex MSE calculation are needed and how the spectral pattern has to be modeled regarding partial levels. The model of the vibration permits calculating the partial frequencies, for which it takes into account the vibration of the stiff string and the effect of the soundboard impedance. The soundboard impedance modeling is under a deeper study using both real measurements and FEM modeling. As further work, once the model is completed, it is expected to generate the patterns by using a waveguide synthesis method. [Work supported by Spanish National Project TEC2006-13067-C03-01/TCM.]