bluuebunny/crossref_metadata_2025 huggingface dataset:
The relative contribution of dynamic and spectral cues in virtual sound source localization
Authors: Zhang Chengyun , Xie Bosun
Published: September 2012

It is well known that the dynamic cues caused by head turning and pinna-based spectral cue are vital to sound source localization, especially for front-back and vertical localization. A series of localization experiment is carried out via virtual auditory display to explore the relative contribution of dynamic and spectral cues to localization. Virtual sources at different intended spatial locations are recreated using various combinations with noise stimuli at full audio or 4 kHz-lowpass frequency ranges, individualized HRTFs or non-individual HRTFs derived from KEMAR and spherical-head model, and static or dynamic binaural synthesis. Furthermore, statistical analyses are performed on localization performance in terms of the percentage of front-back and up-down confusion as well as the mean angle error in virtual source localization. Primary results indicate that both dynamic and spectral cues contribute to front-back and vertical localization; whereas dynamic cue contributes more to front-back localization, while individualized spectral cue at high frequency above 4 kHz contributes more to vertical localization. Dynamic and high-frequency individualized spectral cues are also helpful to reduce the angle error in localization. [Work supported by the National Natural Science Foundation of China,11174087, 50938003, and State Key Lab of Subtropical Building Science, South China University of Technology.]
Approximately calculate individual near-field head-related transfer function using an ellipsoidal head and pinnae model
Authors: Rui Yuanqing , Yu Guangzheng , Xie Bosun
Published: September 2012

Head-related transfer functions (HRTFs) describe the acoustic transmission from a point source to ears and are individual-dependent. Measurement is an effective way to obtain individual HRTFs, but the workload is very heavy. This is particularly difficult in the near-field HRTFs measurement due to their source distance-dependence within 1.0 m. Numerical calculation is another way to obtain HRTFs. To reduce the calculation load, an ellipsoid head and pinnae (HAP) model is proposed in the present work for approximately calculating the individual near-field HRTFs via the fast multipole boundary method (FMBEM). The dimension of ellipsoid head is defined by head width, head height, head depth, which is selected to approximately fit the individual interaural time difference (ITD) and interaural level difference (ILD) below about 5 kHz. The individual pinna geometry obtained via a 3D laser scanner provides individual HRTF spectral cues above 5 kHz. To validate the proposed method, the HAP of KEMAR and human is constructed, and calculating results are compared with the measurements. Psychoacoustic experiment is also carried out to evaluate the model and results. [Work supported by the National Natural Science Foundation of China for Young Scholars (No 11104082) and the Natural Science Foundation of Guangdong Province.
Prior probabilities tune attentional bandwidth
Authors: Wolmetz Michael , Elhilali Mounya
Published: September 2012

Top-down schemas or prior probabilities associated with different auditory objects and their acoustic features are thought to improve auditory scene analysis and auditory object recognition processes. This study focused on whether listeners implicitly track the prior probabilities associated with different tone frequencies, and whether tracking those probabilities modulates attentional bandwidth to improve detection sensitivity. Using the target-probe paradigm, attentional bandwidth for varying levels of probe probability was estimated for 40 listeners. To estimate the attentional bandwidth, probes were presented in a continuous band-pass noise masker at distances of .75, 1.25, 1.75, and 3.75 Equivalent Rectangular Bandwidths from a target frequency of 250 Hz, at varying probabilities of occurrence. All tones were presented at approximately 90% detection when presented alone as measured during a separate thresholding procedure. Results indicate that prior probability plays a role in attentional bandwidth: the attentional band is more broadly tuned when tones at probe frequencies are more likely, consistent with optimal observer predictions. Sensitivity of attentional bandwidth to endogenous (listener-directed) and exogenous (stimulus-driven) attention will be discussed.
What is a good musical pattern? On acoustic structure and goodness of pattern
Authors: Vigo Ronaldo , Zhang Yu , Barcus Mikayla
Published: September 2012

An open problem in acoustical psychophysics involves the extent to which the structuro-acoustical properties of a set of sounds determine goodness of pattern judgments (i.e., how good a pattern is perceived to be). We investigated this question experimentally and theoretically from the standpoint of the dimensional structure of sets of sounds defined over the dimensions of timbre, tone, and duration. We observed a distinctive goodness of pattern ordering for the structures tested which involved sets of four tones defined over the three dimensions. The results were consistent with predictions from categorical invariance theory (CIT; Vigo, 2009, 2011a, 2012) which posits that humans detect the dimensional invariants of a set of discrete sounds — a process described in the theory as structural kernel extraction. Using CIT to interpret our results, we also found that sets of stimuli in “structural equilibrium” (the condition when all of the dimensions of a stimulus set play the same structural role in the set) were perceived as conveying better sound patterns. Based on these results, we propose that the tradeoff between complexity and invariance described in CIT can account for differences in music preferences and style.
Efficient coding of multiple nonorthogonal redundancies between acoustic dimensions in novel complex sounds
Authors: Stilp Christian , Kluender Keith
Published: September 2012

Stilp and colleagues (Proc. Natl. Acad. Sci. [2010]; JASA [2011]; PLoS One [2012]) provided perceptual evidence for efficient coding of robust covariance between acoustic dimensions in novel complex sounds. Discrimination of sounds that directly violated this redundancy (Orthogonal condition) was initially inferior to that of sounds obeying overall redundancy (Consistent condition). Performance was consistently predicted by principal components analysis (PCA), as experimental conditions aligned with statistical dimensions in PCA. Stilp and colleagues suggested efficient coding may contribute to perceptual organization for speech, but two aspects of their experimental designs qualify this extension: robust evidence for only one statistical regularity between acoustic dimensions was tested while speech possesses many; and, all statistical structure was mutually orthogonal which is often not true of speech sounds. Here, listeners discriminated sounds supporting two concurrent, nonorthogonal regularities (patterns of covariance between acoustic dimensions: attack/decay and spectral shape.) Despite nonorthogonality, these concurrent statistical regularities were efficiently coded, as discrimination of Consistent sound pairs was initially superior to that of Orthogonal sound pairs. Performance did not adhere to predictions made by PCA. Implications for speech perception and auditory ‘category’ acquisition will be discussed. [Supported by NIDCD.]
Modeling normal and hearing-impaired monaural and binaural signal detection in the presence of noise
Authors: Furst Miriam
Published: September 2012

Psychoacoustical investigations have demonstrated that monaural and binaural signal detection is affected by existence of distracting noise. The most prominent phenomena are co-modulation masking release (CMR) in monotic listening, and binaural masking level difference (BMLD) in diotic listening. Both CMR and BMLD are significantly deteriorates in hearing-impaired listeners. Both CMR and BMLD phenomena are tested by a complete model of the auditory system for normal and hearing-impaired systems. Prediction of the amplitude discrimination results are obtained by deriving the Cramer Rao lower bound (CRLB) of the neural activity. The auditory system model includes a complete cochlear model with integrated outer hair cells and tectorial membrane; an inner hair cell-synapse model that transduce the cilia motion to auditory nerve instantaneous rate; Inferior Colliculus that receives inputs from both ears and process them by excitatory-inhibitory (EI) cells. The AN activity is considered as a non-homogeneous Poisson process (NHHP). We have recently showed that EI cells are NHHP as well, if their inputs behave as NHHP. Therefore, CRLB can be derived analytically from both AN and IC outputs. We have successfully predicted major CMR and BMLD properties as a function of frequency and noise properties for normal and impaired auditory system.
Sound reduction from vibrating thin plates using dimpling and beading design
Authors: Myers Kyle R. , Alshabatat Nabeel T. , Naghshineh Koorosh
Published: September 2012

This study presents a design method to minimize the radiated sound power from vibrating beams and plates. The method relies on altering the acoustic characteristics of beam and plate structures passively by forming dimples and beads on their surfaces. The dimples and beads change the local stiffness of the structure without changing its mass. Also, they alter the mode shapes so as to reduce sound power. The vibration response of the dimpled and beaded beams and plates are calculated using the finite element method (i.e., ANSYS parametric design language). Then, the radiated sound power from vibrating structure is calculated based on the Lumped Parameter Model (LPM). Finally, the method of Genetic Algorithm (GA) is used to optimally locate and size the dimples or the beads on the plate to minimize the sound radiation. The sound radiation is minimized either at a single frequency, or over a broad frequency band. The results show that dimples or beads forming on thin beams and plates can achieve effective reductions in radiated sound power.
Fast wave source localization with sparse measurements
Authors: Sabelli Anthony , Aquino Wilkins
Published: September 2012

Source localization problems are encountered in a variety of engineering dis- ciplines. Applications include earthquake localization, damage identification, speaker localization and structural testing. In most realistic settings, measure- ment points are sparse with respect to the physical domain. Moreover, the experimenter may not have control over where to place measurement points. It is in these imperfect settings that we still need to estimate the location of a wave source. In this talk we will outline a method for source localization inspired by the topological derivative used in shape identification. We will draw parallels to phase conjugation mirror techniques and gradient optimization. We will make no assumptions about the nature of the ambient media nor the computational domain. Specifically we allow for energy loss. We will also outline implemen- tation within an existing massively parallel finite element solver. Our proposed algorithm is minimally invasive and fully exploits the underlying optimization in the existing solvers. Moreover we can extend the method to other physical contexts.
Sound radiation of double rotating dipole
Authors: Wang John , Xu Hongan
Published: September 2012

Mechanical devices are modeled as single and double rotating dipoles in this study. The expressions for sound pressure, intensity and power are derived. Sound pressure and sound power level reduction is investigated. The reduction is quantified. The theory is compared with test data. The applications are discussed.
Speech production under real-time simulation of cochlear implant acoustic feedback
Authors: Casserly Elizabeth D. , Pisoni David B.
Published: September 2012

Although previous research on simulation of cochlear implant (CI) processing with normal-hearing listeners relied on offline transformation of pre-recorded acoustic signals, the advent of new simulation technology using a Portable Real-Time Vocoder (PRTV) enables subjects to experience CI simulation not only of interlocutors’ speech during face-to-face interaction, but also of their own speech acoustics. This paper explores the effects of this novel acoustic feedback manipulation on subjects’ speech production. Nine normal-hearing speakers were recorded producing 114 isolated English words and 24 sentences during three time epochs: once under normal conditions, once immediately after being fitted with the PRTV, and again after experiencing a short, natural conversational interaction through the real-time (8-channel, noise-vocoded) CI simulation. Acoustic-phonetic analysis of subjects’ speech revealed substantial, segment-specific shifts in vowel acoustics, alteration of sibilant frication frequencies, and evidence of increased cognitive load (e.g. slower speaking rates) during speech production under conditions of vocoded acoustic feedback. Speakers also appeared to alter their articulatory strategies, spending more time on production of consonants relative to vowels, possibly reflecting a flexible exploitation of reliable somatosensory versus aural feedback cues.
Developmental changes in voiceless fricative production: Influence of position in words
Authors: Nishi Kanae , Graham Elizabeth C.
Published: September 2012

Children master the production of fricative consonants later than other phonemes [Moeller et al., Ear Hear. 28, 605-627 (2007)]. Even though recognizable fricative categories are present before school age, fine-tuning of acoustic properties may continue throughout early adolescence [Nissen & Fox, J. Acoust. Soc. Am. 118, 2570-2578 (2005)]. Previous acoustic studies on the development of fricative production focused on those in word-initial position only. Even in adults’ speech, acoustics of consonants in word-medial or word-final positions vary more compared to those in the word-initial position. The present study hypothesized that adult-like production of fricatives in the word-final position may be achieved later than those in the word-initial position due to the acoustic variability in the adult exemplars children hear. Thirty-six (six each of 4-, 6-, 8-, 10-, 12-year-olds and adults) female native speakers of American English recorded five tokens of 16 consonant-vowel-consonant monosyllabic real words containing voiceless fricative consonants /f θ s ∫/ in initial or final position in /i/ and /ɑ/ vowel contexts. Each token was analyzed for frication duration, amplitude, and several spectral measures. Results will be discussed in terms of fricative position in word, vowel context, and speaker age. [Work supported by NIDCD R03 DC009334 and P30 DC004662.]
Understanding speech acoustics in an era of extreme cue-integration: Multi-dimensional phonetics reveals individual differences in fricative production
Authors: Rhone Ariane E. , Apfelbaum Keith S. , McMurray Bob
Published: September 2012

Phonological features are indicated by many acoustic cues (Lisker, 1986). Listeners must thus combine multiple cues to recognize speech (Nearey, 1990). A recent survey of cues to English fricatives identified 24 distinct, useful cues (McMurray & Jongman, 2011). This multi-dimensional nature of speech raises methodological challenges: cues do not contribute equally, and multiple cues contribute to the same phonetic features (e.g. voicing). We offer a solution, using McMurray and Jongman’s (2011) fricative database as a test. We used a logistic regression to predict the fricatives using measurements of 24 cues. We then sum the product of each cue value and its weight from the regression to determine the strength of the evidence for a given feature/phoneme (e.g. degree of “f-ness” vs. “v-ness”) for each token. By computing this for different subsets of cues, we can measure how classes of cues work in tandem. We illustrate this by examining the relative contribution of cues within the frication and those within the vocoid, as well as spectral and temporal cues, to show individual differences in fricative productions. These analyses offer a straightforward approach to conceptualizing speech perception in high multi-dimensional space, while also giving insights on the production of English fricatives.
Language specificity in the perception of children's productions of /t/ and /k/
Authors: Munson Benjamin , Yoneyama Kiyoko , Edwards Jan
Published: September 2012

The age and order of acquisition of what are ostensibly the 'same' sounds can differ across languages. These differences relate to a number of factors, including frequency in the ambient language, language-specific phonetic instantiations of sounds, and language-specific parsing of children's emerging productions (Beckman & Edwards, 2010; Edwards & Beckman, 2008; Li et al, 2011). The current investigation examines the role of adults' perception of children's speech on the acquisition of /t/ and /k/ in English- and Japanese-speaking children. Previous work has shown that /t/ is acquired earlier than /k/ in English, but that the opposite is true in Japanese (Nakanishi et al., 1972; Smit et al., 1990). We examined whether this tendency was due to cross-linguistic differences in adults' perception of English- and Japanese-acquiring children's speech. Native speakers of English and Japanese labeled a large set of 2- to 5-year-old children's word-initial /t/- and /k/ productions. Japanese-speaking adults perceived English-speaking children's productions of sounds intermediate between /t/ and /k/ as more /k/-like than did English-speaking adults. This suggests that the earlier acquisition of /k/ in Japanese than in English may be due, in part, to Japanese-speaking adults' willingness to label ambiguous sounds as /k/-like.
Voicing, aspiration, and vowel duration in Hindi
Authors: Durvasula Karthik , Luo Qian
Published: September 2012

There is extensive evidence that consonantal laryngeal features modulate adjacent vowel duration (Chen 1970). However, it is not clear if both consonant voicing and aspiration affect vowel duration. Previous studies (on Hindi) produced inconsistent results with respect to the effect of consonant aspiration on vowel duration, while finding a clear positive correlation with consonant voicing (Maddieson & Gandour 1976; Ohala & Ohala 1992; Lampp & Reklis 2004). We conducted an experiment on 7 native standard Hindi speakers, who produced 10 repetitions of 12 nonsense words ending in [d̪, d̪h, t̪, t̪h] that had 3 different CVCVV contexts. The results of the experiment show that there is a statistically significant main effect of coda voicing on vowel duration and a marginally significant main effect of aspiration on vowel duration. Furthermore, the effect of the aspirated coda consonants on vowel duration appears to be modulated by the surrounding segmental context. The results suggest that both consonant voicing and aspiration increase adjacent vowel duration. The results also suggest that the inconsistent findings in previous studies with respect to the effect of aspiration could be the result of differing phonetic contexts of the relevant segments.
Dialectal and age-related acoustic variation in vowels in spontaneous speech
Authors: Jacewicz Ewa , Fox Robert A.
Published: September 2012

Our knowledge of the acoustic characteristics of vowels is based mainly on productions obtained under fine experimental control (e.g., [hVd] tokens produced in isolation). Analyzing spontaneous speech data has its obvious challenges because the vowels of interest occur in various segmental and prosodic contexts and are additionally affected by sudden changes in speech tempo. These variations may compromise the accuracy of interpretation of linguistic phenomena such as sound change. In this study we examine if more natural productions of vowels in two dialects (as compared to those produced in citation form and read speech) show evidence of the existence of cross-generational vowel changes—including corresponding changes in spectral dynamics. A subset of vowels from a large corpus of spontaneous conversations was analyzed. The vowels occurred in variable consonantal contexts in both mono- and polysyllabic words. The obtained patterns of vowel change were consistent with those in read and citation form speech. Although the measured spectral changes were smaller due to shorter vowel durations, the dialect-specific nature of formant dynamics was maintained. These results demonstrate that the patterns of vowel change, including variation in formant dynamics, do not diminish under the circumstances of greater contextual/prosodic variability found in spontaneous speech.
Cross-dialectal and cross-generational changes in point-vowel locations in English
Authors: Fox Robert A. , Jacewicz Ewa
Published: September 2012

The positions of individual vowels in the acoustic vowel space often change over time in languages such as English. This study examines the changes in the location of four point vowels [i u ɑ æ] for three English dialects (North, Midland and Inland South) as produced by three generations of speakers (children, adults aged 35-50 and 65+ years). We determined the speaker-specific spectral centroids of the midpoints of a set of 12 monophthongs in the F1 x F2 vowel space produced by each individual speaker. These formant values were then normalized using Nearey’s formula. For both the raw and normalized F1 x F2 spaces, polar coordinates for productions of each point vowel were calculated with this centroid as the origin (producing a radius and an angle—a variation of Chung et al.’s [2012]) along with vowel space areas. Compass plots were drawn for the vowels of each speaker and angle (polar) histograms for each vowel were created for each speaker group. Analysis of the histograms and radii showed significant differences among the groups in terms of the location of /æ/ and /u/ as well as area differences. The implications these data have for sound change in progress will be discussed.
A production and perception study on tonal neutralization in Nanchang Chinese
Authors: Liu Jiang , Zhang Jie
Published: September 2012

In a production study of tonal contrasts in lexically stressed but grammatically stressless syllables vs. lexically stressless syllables in Nanchang, a Gan dialect spoken in southeastern China, we found that tonal neutralization only occurs in lexically stressless syllables. We argue that the main phonetic ground for such a tonal contrast distribution lies in the rhyme duration difference between syllables with and without lexical stress, namely, lexically stressless syllables have shorter rhyme duration than lexically stressed but grammatically stressless syllables, and the shorter the rhyme duration of a syllable is the fewer tonal contrasts the syllable allows. In terms of perception, we found that different tonal contrasts indeed become neutralized in lexically stressless syllables. However, the neutralization pattern at the perception level is not the same as the one at the production level due to word specific effects.
Pitch and intensity in the speech of Japanese speakers' of English: Comparison with L1 speakers
Authors: Okada Jun , Wilson Ian L. , Yoshizawa Miyuki
Published: September 2012

The speech of L2 learners of English is often difficult to understand because of intonation problems and misplaced word stress. In this research, we investigated whether or not the intonation patterns of Japanese speakers of English show common patterns based on proficiency level. First, we recorded “The North Wind and the Sun” from 50 Japanese undergraduate students (aged 18 to 24). We recorded native English speakers and also obtained such native data online. Next, we labeled each word and analyzed the pitch and intensity using Praat. Data was separated by gender and by proficiency in English, results were plotted, and statistical analysis was undertaken. Preliminary results show that pitch (and to a lesser extent, intensity) showed a common pattern across native speakers, but that L2 speakers relied on intensity much more than pitch in the production of stress.
An experimental study of Korean rhythm structure on the basis of rhythm metrics
Authors: Tark Eun-Sun
Published: September 2012

This paper investigates the rhythm structure of Korean. Metrics used in this study included %V, ΔC, Varco V, nPVI-V, and rPVI-C, which have been shown to reflect differences in rhythm structure (stress-, syllable-, and mora-timing) across languages. Ten female native Koreans each produced 20 short declarative Korean sentences. The rhythm metric results of Korean were compared to those of English, Spanish, and Japanese using raw data from previous studies [Ramus et al. 1999, Cognition 73, 265-292; White and Mattys, 2007, Journal of Phonetics 35, 501-522; Grenon and White, 2008, BUCLD 32, 155-166]. Results indicate that Korean combines aspects of both syllable timing and mora timing. Korean has a similar ΔC, Varco V, and nPVI-V to syllable-timed languages. Korean has a similar %V and nPVI-V to mora-timed languages. These data show that instead of belonging to one of three distinct rhythm categories, languages may be placed along a rhythm structure continuum. On such a continuum, Korean is placed between syllable-timed and mora-timed languages, and distinct from stress-timed languages.
Random matrix theory in signal processing: Performance analyses and algorithm design
Authors: Richmond Christ D.
Published: September 2012

Estimation of covariance matrices from a finite sample set of data observations plays a central role in signal processing. The true data covariance is rarely known in practice, but optimized algorithms inevitably depend on such statistics to enable robust system performance; for example, environments plagued by dominant interference, and/or those challenged by complex propagation. Classical (finite) random matrix theory (RMT) facilitates assessment of finite sample effects surrounding covariance estimation. Implements shown to be very useful in this regard under a circular complex Gaussian data assumption are reviewed. Recent advances in RMT explore limiting behavior of eigenvalues and eigenvectors of random matrices as dimensions become increasingly large (referred to as infinite RMT). Defining the notion of an empirical distribution for the eigenvalues deviates from classical treatments, but yields amazing convergence toward deterministic distributions. Coupled with the Stieltjes transform, powerful tools emerge that can provide new insights especially for signal processing methods intimately tied to the eigen-decomposition, e.g. diagonal loading and dominant mode rejection used in adaptive beamforming. Although many of the theorems are based on asymptotic convergence as dimensionality increases, they describe performance of finite systems quite well. Aspects of infinite RMT are also reviewed and contrasted with classical RMT.
Constructing acoustic timefronts using random matrix theory
Authors: Hegewisch Katherine C. , Tomsovic Steven
Published: September 2012

In a recent letter [Europhys.Lett. 97, 34002 (2012)], random matrix theory is introduced for long-range acoustic propagation in the ocean. The theory is expressed in terms of unitary propagation matrices that represent the scattering between acoustic modes due to sound speed fluctuations induced by the ocean's internal waves. The scattering exhibits a power-law decay as a function of the differences in mode numbers thereby generating a power-law, banded, random unitary matrix ensemble. This talk describes that approach and extends the methods to the construction of an ensemble of acoustic timefronts. The result is a very efficient method for studying the statistical properties of timefronts at various propagation ranges that agrees well with propagation based on the parabolic equation. It helps identify which information about the ocean environment survives in the timefronts and how to connect features of the data to the surviving environmental information. It also makes direct connections to methods used in other disordered wave guide contexts where the use of random matrix theory has a multi-decade history. [This work was supported by ONR and NSF.]
Random matrix theory and performance prediction of subspace methods
Authors: Nadakuditi Raj R.
Published: September 2012

Subspace methods constitute a powerful class of techniques for detection, estimation and classification of signals buried in noise. Recent results in random matrix theory precisely quantify the accuracy of subspaces estimates from finite, noisy data in both the white noise and colored noise setting. This advance facilitates unified performance analysis of signal processing methods that rely on these empirical subspaces. We discuss the pertinent theory and its application to the characterization of the performance of direction-of-arrival estimation, matched subspace detection and subspace clustering for large arrays in the sample-starved setting for both white and colored noise.
Eigenvalues of the sample covariance matrix for a towed array
Authors: Gerstoft Peter , Menon Ravishankar , Hodgkiss William S. , Mecklenbrauker Christoph
Published: September 2012

Observations of the spatial sample covariance matrix (SCM) reveal that the ordered noise eigenvalues of the SCM decay steadily. Using a stochastic model for the sample covariance matrix, the empirical eigenvalue distribution can be derived using random matrix theory. The eigenvalue spectrum is directly related to the quantile function of the empirical eigenvalue distribution. These ordered eigenvalues have a decay that resembles what is observed in real data. Noise on the array is considered either incoherent self-noise or propagating acoustic noise that is coherent across the array. Using conventional 2D or 3D isotropic noise models, realizations of the SCM eigenvalues are generated using random matrix theory. Deep-water towed-array data are analyzed and it is shown that the eigenvalues of the SCM and compares well with theory.
Random matrix theory analysis of the dominant mode rejection beamformer
Authors: Wage Kathleen E. , Buck John R.
Published: September 2012

The Dominant Mode Rejection (DMR) beamformer developed by Abraham and Owsley [Proc. Oceans, 1990] determines the beamformer weights from the sensor covariance matrix eigendecomposition. The weights are designed to reject signals contained in the dominant subspace, which is defined by the eigenvectors associated with the largest eigenvalues. In previous work, we developed a model for the mean notch depth (ND) of the DMR beamformer from random matrix theory (RMT) results on the sample eigenvector fidelity [IEEE Stat. Sig. Proc. workshop, 2012]. While ND is useful, other metrics such as white noise gain (WNG) and signal to interference and noise ratio (SINR) are of great interest. WNG characterizes the beamformer robustness to mismatch, and SINR quantifies overall performance. SINR loss is defined as the ratio of the SINR for a beamformer designed using sample statistics to the SINR for the optimal beamformer designed using ensemble statistics. This talk extends our previous work by considering the relationship among ND, WNG, and SINR for the DMR beamformer. A surprising result obtained from RMT is that for a single loud interferer and twice as many snapshots as sensors, the expected SINR loss depends only on the number of snapshots. [Work supported by ONR.]
Aerial hearing sensitivity in a southern sea otter (Enhydra lutris nereis)
Authors: Ghoul Asila , Reichmuth Colleen
Published: September 2012

The lack of information concerning auditory sensitivity in sea otters has been recognized by biologists and resource managers as a priority research need for this threatened species. Noise-generating human activity in near-shore marine environments occurs as a result of construction, transportation, exploration and recreation. These activities may degrade critical habitat or cause behavioral or auditory effects that are harmful to individuals. As direct measures of hearing are not presently available for sea otters, we obtained psychophysical hearing thresholds from a trained individual. Audiometric testing was conducted with an adult male sea otter using 500 ms frequency-modulated narrow-band sweeps under quiet conditions. Absolute aerial thresholds were collected at eleven frequencies ranging from 0.125 to 45 kHz. The sea otter showed a broad functional range of hearing, extending from 0.250 to ~40 kHz, with best sensitivity between 2 and 16 kHz. The lowest measured threshold was -1 dB re 20 μPa at 8 kHz. The high-frequency hearing data was similar to that of terrestrial carnivores, while hearing thresholds below 1 kHz showed a relative decrease in sensitivity. Measurements of underwater sensitivity in the same sea otter are ongoing, and will inform explorations of amphibious hearing capabilities in marine mammals, as well as provide insight into the effects of anthropogenic noise on this vulnerable species.
Auditory thresholds in marine vertebrates conform to natural ambient noise levels
Authors: Stocker Michael A. , Reuterdahl John T.
Published: September 2012

Auditory thresholds are often displayed in a manner that reveals what is commonly called a “U-curve.” But if the threshold curves are displayed on the x axis on a true Log10 scale the profile is shaped differently. For marine mammals the shape is more like a “hockey stick.” If these curves are overlaid on the “Wenz ambient noise spectra curves” there appears to be shape conformance. This makes sense as auditory sensitivity would naturally evolve to exclude ambient environmental noise. This paper evaluates 120 legacy auditory threshold curves from 18 species of marine mammals and 60 threshold curves from 32 species of fish. The auditory threshold curves from the fish do not conform to the Wenz curves. Given that both the auditory thresholds and the Wenz curves were expressed as pressure gradient energy it is possible that the profile of the fish threshold curves express sound in either the particle velocity, or both particle velocity and pressure gradient energy. This paper extrapolates the particle velocity data from the fish threshold conditions to determine if there is some conformity to ambient noise levels in either or both the particle and pressure gradient realms.
Animal-borne active acoustic tags: A new paradigm to conduct minimally invasive behavioral response studies?
Authors: Klinck Holger , Horning Markus , Mellinger David K. , Costa Daniel P. , Fregosi Selene , Mann David A. , Sexton Kenneth , Huckstadt Luis
Published: September 2012

In 2011 a pilot study was begun to evaluate the potential of animal-borne active acoustic tags for conducting minimally-invasive behavioral response studies on pinnipeds. A basic prototype tag was developed and tested on juvenile northern elephant seals (Mirounga angustirostris) during translocation experiments at Año Nuevo State Park, CA, USA in spring 2012. The principal scientific questions of this pilot study were these: (1) do sounds emitted from an animal-borne low acoustic intensity tag elicit behavioral responses, and (2) are potential animal responses related to signal content (e.g., threatening vs. non-threatening). Although the sample size was small, preliminary results indicate that (1) low-intensity sounds emitted by animal-borne tags elicit distinct behavioral responses, (2) these responses appear related to signal content, and (3) the responses may differ based on depth, bathymetry, and location. The results of the conducted study show the promise of this approach as a minimally-invasive and cost-effective method to investigate animal responses to underwater sounds, as well as a method to develop mitigation strategies. Future efforts would increase the sample size, range of acoustic stimuli, and age/sex classes of tagged seals. [Funding from NOAA/NMFS Ocean Acoustics Program.]
Passive acoustic monitoring on the North Atlantic right whale calving grounds
Authors: Soldevilla Melissa , Garrison Lance , Clark Christopher
Published: September 2012

Shallow water environments, such as the North Atlantic right whale calving grounds, pose a challenge to cetacean passive acoustic monitoring due to high variability in ambient noise and environmental conditions. In this region of high shipping traffic and increased ship-strike risk, passive acoustic monitoring may reduce right whale ship strikes. This study describes temporal variability in right whale call detections, ambient noise sources, and environmental conditions on the right whale calving grounds during 2009-2010 and 2010-2011. Right whale detections occurred between November 19 and March 11, on up to 25% of days per deployment with increased nocturnal call detections, and increased acoustic presence off Jacksonville, FL during 2010-2011. Shipping noise was most common off Jacksonville, detected in up to 74% of minutes, with a diurnal peak, while tidally-associated broadband impulses were detected in up to 43% of minutes off Savannah GA. Environmental conditions including SST, wind, waves, and tidal height varied on daily and semi-diurnal scales. While sightings were higher in 2009-2010, the fewer sightings in 2010-2011 were more narrowly distributed within the depth range of the acoustic instruments. Passive acoustic monitoring is effective for detecting right whales in this environment, especially at night when they cannot be seen.
New directions for manipulation of sound using acoustic metamaterials
Authors: Naify Christina J. , Orris Gregory J. , Martin Theodore P. , Layman Christopher N.
Published: September 2012

Manipulation of sound waves using acoustic metamaterials has expanded significantly in recent years. Acoustic metamaterials are a class of materials that use sub-wavelength structures to achieve effective bulk properties under acoustic excitation. Unusual effective physical properties, including negative bulk modulus, negative mass density, and negative index have been achieved using metamaterials. Additionally, the development of structures based on transformational acoustics has resulted in designs for scattering reduction and sound focusing. Current research emphases include expansion from narrowband, resonant structures to broadband structures, as well as the design and construction challenges of three-dimensional structures. Active or tunable structures are also being explored. Examples will be given of negative index and three-dimensional metamaterial structures. [Work is supported by the Office of Naval Research.]
Continuous and impulsive noise attenuation performance of passive level dependent earplugs
Authors: McKinley Richard L. , Gallagher Hilary L. , Theis Melissa , Murphy William J.
Published: September 2012

Level dependent hearing protectors, earplugs and earmuffs, have advanced in technology due to the needs of military personnel and others to reduce the risk of hearing damage from impulsive noise. These hearing protectors were developed to preserve ambient listening capabilities therefore improving situational awareness while reducing the risk of noise induced hearing loss by attenuating both continuous and impulsive noise. Four commercially available passive level dependent earplugs were assessed for both continuous noise attenuation and impulsive insertion loss performance. The continuous noise attenuation results were collected using American National Standard Institute (ANSI) S12.6-2008 Methods for Measuring the Real-Ear Attenuation of Hearing Protectors while the impulsive insertion loss results were collected using ANSI S12.42-2010 Methods for the Measurement of Insertion Loss of Hearing Protection Devices in Continuous or Impulsive Noise Using Microphone-in-Real-Ear (MIRE) or Acoustic Test Fixture Procedures. The presentation will include the passive noise attenuation performance of level dependent earplugs for both continuous and impulsive noise. The impulsive insertion loss results for these particular hearing protectors will be applied to impulsive noise damage risk criteria for an estimate of allowable impulsive noise exposure.
Effective attenuation performance of passive hearing protectors: A temporary threshold shift study
Authors: McKinley Richard L. , Gallagher Hilary L. , Theis Melissa
Published: September 2012

Passive hearing protectors have been used for decades to reduce the risk of noise induced hearing loss. Hearing protectors (earmuffs, earplugs, helmets) have traditionally been the first line of defense for personnel working in hazardous noise environments. According to ANSI S12.68-2007, the “gold standard” method of estimating effective A-weighted sound pressure levels when hearing protectors are worn is the classical octave band method. The octave band method subtracts the hearing protector noise attenuation from the ambient noise level for each relevant octave band to estimate the noise exposure at the ear, under the hearing protector. ANSI S12.6-2008 Methods for Measuring the Real-Ear Attenuation of Hearing Protectors was used to measure the attenuation of the hearing protectors. The purpose of this study was to measure the effective attenuation of a hearing protector in terms of temporary threshold shift (TTS) response for individual human subjects with and without hearing protection. This presentation will include the TTS response curves for subjects exposed to various noise levels and durations in a controlled laboratory environment. The passive hearing protectors evaluated in this study included an earplug, earmuff, and a headphone with minimal attenuation as determined by REAT.
Measurements of bone-conducted impulse noise from weapons using a head simulator
Authors: Clavier Odile H. , Dietz Anthony J. , Wilbur Jed C. , Zechmann Edward L. , Murphy William J.
Published: September 2012

High-intensity impulse sounds are generally considered to be more damaging than continuous sounds, so understanding the attenuation performance of hearing protection devices against impulse noise is key to providing adequate protection for exposed persons. The maximum attenuation of hearing protection devices is limited by bone-conducted sound. Weapon fire noise in the form of short duration impulses can reach peak levels of 170 dB SPL at the shooter’s ear, a sound level for which maximum hearing protection is recommended and for which bone-conducted sound will be a significant factor. However, current acoustic test fixtures do not capture the bone-conducted sound paths. In this study, an anatomically correct head simulator built specifically to measure bone-conducted sound was used to evaluate the effects of impulse noise generated by hand guns and rifles at several peak sound pressure levels ranging between 120 dB SPL and 170 dB SPL. Time histories of the acceleration of the temporal bones and the sound pressure transmitted into the cranial cavity were recorded. Results investigating the linearity of the bone-conducted response to impulse noise at high peak levels and the effects of hearing protection on the sound level and vibrations inside the head are presented.
Adaptive feedforward control for active noise cancellation in-ear headphones
Authors: Priese Sylvia , Bruhnken Christoph , Voss Daniel , Peissig Jürgen , Reithmeier Eduard
Published: September 2012

Noise can be disturbing, stressful or even harmful. Headphones with active noise cancellation (ANC) can enhance the user’s comfort, especially when travelling. On a plane or a train, in the street or at work, these headphones give the possibility to reduce unwanted noise. The range of ANC headphones on the market is constantly increasing. Circumaural and supra-aural headphones with different control strategies have been available for a long time; over the last few years the product lines have been expanded to in-ear headphones. These headphones already have quite a good passive attenuation and are equipped with feedforward control for active noise cancellation. The best results in attenuation are achieved by semi-adaptive digital controls, which choose the best filter depending on the noise spectrum and can be manually adapted to the user. A fully adaptive control has already been proven to be very effective in aviation headsets and other ANC applications. Besides the market analysis of ANC headphones we would like to present an adaptive feedforward control for in-ear headphones and highlight the advantages compared to a static feedforward control.
Design of a feedback controller for active noise control with in-ear headphones
Authors: Bruhnken Christoph , Priese Sylvia , Foudhaili Hatem , Peissig Jürgen , Reithmeier Eduard
Published: September 2012

Nowadays mobility is an important factor in many jobs. Therefore, there is an increased use of planes, trains and cars, and the associated exposure to noise. Good acoustic insulation is often hard to realize due to the involved extra weight. Ear protection or headphones with active noise control (ANC) may be a possible solution. Today circumaural and supra-aural ANC headphones with good attenuation are commercially available. However, their weight and the necessary headband can impair the wearing comfort. ANC in-ear headphones do not have these disadvantages and, therefore, there is a need of further research in the field of ANC. In ANC headphones, disturbing noise is minimized by an out-of-phase anti-noise. Therefore, the noise is recorded by microphones next to each ear, and filtered by an analog or digital platform to generate the anti-noise. There are two main control strategies depending on the position of the microphones, feedforward control with an external reference microphone and feedback control with an internal error microphone. The presentation will focus on the design of feedback controllers and the main problem regarded to in-ear headphones, interpersonal variances, which make the design of stable controllers with high noise attenuation difficult. A model-based solution will be presented.
The application of differential spread spectrum technology in underwater acoustic communication
Authors: Han Xiao , Yin Jingwei , Du Pengyu , Guo Xiaoyu
Published: September 2012

In underwater acoustic channel, the Doppler effect produced by relative movement between Source and information destination is very complex. Currently, the spread spectrum system typically uses PSK modulation. As the transmission characteristic in water sound channel is phase rapid changing, spread spectrum systems based on PSK modulation need high precision in estimating the carrier and need continuous tracking of the carrier, which make the performance in practical applications limited. Differential spread spectrum acoustic communication technology is studied in this paper. Using differential coherent demodulation method at the receiving end, which solves the problem of estimating the carrier in underwater acoustic communication, can overcome the frequency and phase error due to the drift of the carrier in transfer process. This method is verified Through computer simulation studies and Lake test.
Volterra series-based non-linearity analysis of shallow water acoustic channels
Authors: Huang Xiaopeng
Published: September 2012

Most of existing underwater acoustic (UWA) communication systems are based on the linear UWA channels. However, some environmental factors (e.g., bubble plumes) in complicated shallow water environments will contribute to the non-linearity of channels. Therefore, In order to fully understand the properties of shallow water acoustic channels, and develop more bandwidth-efficient communication systems in complicated shallow water environments, we adopt the Volterra series to analyze the non-linearity of shallow water acoustic channels for the first time, and its completed theoretical derivations will be presented. Volterra series combines the representations of a nonlinear system without memory and a linear, casual system with memory to describe a nonlinear system with memory. Generally speaking, the central problem in using a Volterra approach to the analysis of nonlinear channels with momory consists of estimating the Volterra kernels, which represent a nonparametric characterization of the channel.