The 50 most-cited articles in The Journal of the Acoustical Society of America, 2017

Speech separation can be formulated as learning to estimate a time-frequency mask from acoustic features extracted from noisy speech. For supervised speech separation, generalization to unseen noises and unseen speakers is a critical issue. Although deep neural networks (DNNs) have been successful in noise-independent speech separation, DNNs are limited in modeling a large number of speakers. To improve speaker generalization, a separation model based on long short-term memory (LSTM) is proposed, which naturally accounts for temporal dynamics of speech. Systematic evaluation shows that the proposed model substantially outperforms a DNN-based model on unseen speakers and unseen noises in terms of objective speech intelligibility. Analyzing LSTM internal representations reveals that LSTM captures long-term speech contexts. It is also found that the LSTM model is more advantageous for low-latency speech separation and it, without future frames, performs better than the DNN model with future frames. The proposed model represents an effective approach for speaker- and noise-independent…

Source localization in ocean acoustics is posed as a machine learning problem in which data-driven methods learn source ranges directly from observed acoustic data. The pressure received by a vertical linear array is preprocessed by constructing a normalized sample covariance matrix and used as the input for three machine learning methods: feed-forward neural networks (FNN), support vector machines (SVM), and random forests (RF). The range estimation problem is solved both as a classification problem and as a regression problem by these three machine learning algorithms. The results of range estimation for the Noise09 experiment are compared for FNN, SVM, RF, and conventional matched-field processing and demonstrate the potential of machine learning for underwater source localization.

Machine learning classifiers are shown to outperform conventional matched field processing for a deep water (600 m depth) ocean acoustic-based ship range estimation problem in the Santa Barbara Channel Experiment when limited environmental information is known. Recordings of three different ships of opportunity on a vertical array were used as training and test data for the feed-forward neural network and support vector machine classifiers, demonstrating the feasibility of machine learning methods to locate unseen sources. The classifiers perform well up to 10 km range whereas the conventional matched field processing fails at about 4 km range without accurate environmental information.

Languages show systematic variation in their sound patterns and grammars. Accordingly, they have been classified into typological categories such as stress-timed vs syllable-timed, or Head-Complement (HC) vs Complement-Head (CH). To date, it has remained incompletely understood how these linguistic properties are reflected in the acoustic characteristics of speech in different languages. In the present study, the amplitude-modulation (AM) and frequency-modulation (FM) spectra of 1797 utterances in ten languages were analyzed. Overall, the spectra were found to be similar in shape across languages. However, significant effects of linguistic factors were observed on the AM spectra. These differences were magnified with a perceptually plausible representation based on the modulation index (a measure of the signal-to-noise ratio at the output of a logarithmic modulation filterbank): the maximum value distinguished between HC and CH languages, with the exception of Turkish, while the exact frequency of this maximum differed between stress-timed and syllable-timed languages. An additional…

This study examines a near-field acoustic holography method consisting of a sparse formulation of the equivalent source method, based on the compressive sensing (CS) framework. The method, denoted Compressive-Equivalent Source Method (C-ESM), encourages spatially sparse solutions (based on the superposition of few waves) that are accurate when the acoustic sources are spatially localized. The importance of obtaining a non-redundant representation, i.e., a sensing matrix with low column coherence, and the inherent ill-conditioning of near-field reconstruction problems is addressed. Numerical and experimental results on a classical guitar and on a highly reactive dipole-like source are presented. C-ESM is valid beyond the conventional sampling limits, making wide-band reconstruction possible. Spatially extended sources can also be addressed with C-ESM, although in this case the obtained solution does not recover the spatial extent of the source.

Non-invasive, focal neurostimulation with ultrasound is a potentially powerful neuroscientific tool that requires effective transcranial focusing of ultrasound to develop. Time-reversal (TR) focusing using numerical simulations of transcranial ultrasound propagation can correct for the effect of the skull, but relies on accurate simulations. Here, focusing requirements for ultrasonic neurostimulation are established through a review of previously employed ultrasonic parameters, and consideration of deep brain targets. The specific limitations of finite-difference time domain (FDTD) and k-space corrected pseudospectral time domain (PSTD) schemes are tested numerically to establish the spatial points per wavelength and temporal points per period needed to achieve the desired accuracy while minimizing the computational burden. These criteria are confirmed through convergence testing of a fully simulated TR protocol using a virtual skull. The k-space PSTD scheme performed as well as, or better than, the widely used FDTD scheme across all individual error tests and in the convergence of…

This study starts with a simple question: can the vibration of plates or beams be efficiently reduced using a lightweight structure that occupies a small space? As an efficient technique to damp vibration, the concept of an acoustic black hole (ABH) is adopted with a simple modification of the geometry. The original shape of an ABH is a straight wedge-type profile with power-law thickness, with the reduction of vibration in beams or plates increasing as the length of the ABH increases. However, in real-world applications, there exists an upper bound of the length of an ABH due to space limitations. Therefore, in this study, the authors propose a curvilinear shaped ABH using the simple mathematical geometry of an Archimedean spiral, which allows a uniform gap distance between adjacent baselines of the spiral. In numerical simulations, the damping performance increases as the arc length of the Archimedean spiral increases, regardless of the curvature of the spiral in the mid- and high-frequency ranges. Adding damping material to an ABH can also strongly enhance the damping performance…

Ultrasound computed tomography (USCT) is a non-invasive imaging technique that provides information about the acoustic properties of soft tissues in the body, such as the speed of sound (SS) and acoustic attenuation (AA). Knowledge of these properties can improve the discrimination between benign and malignant masses, especially in breast cancer studies. Full wave inversion (FWI) methods for image reconstruction in USCT provide the best image quality compared to more approximate methods. Using FWI, the SS is usually recovered in the time domain, and the AA is usually recovered in the frequency domain. Nevertheless, as both properties can be obtained from the same data, it is desirable to have a common framework to reconstruct both distributions. In this work, an algorithm is proposed to reconstruct both the SS and AA distributions using a time domain FWI methodology based on the fractional Laplacian wave equation, an adjoint field formulation, and a gradient-descent method. The optimization code employs a Compute Unified Device Architecture version of the software k-Wave, which…

Band gaps in conventional phononic crystals (PCs) are attractive for applications such as vibration control, wave manipulation, and sound absorption. Their practical implementations, however, are hampered by several factors, among which the large number of cells required and their impractically large size to ensure the stopbands at reasonably low frequencies are on the top of the list. This paper reports a type of beam carved inside with two double-leaf acoustic black hole indentations. By incorporating the local resonance effect and the Bragg scattering effect generated by a strengthening stud connecting the two branches of the indentations, ultrawide band gaps are achieved. Increasing the length of the stud or reducing the residual thickness of the indentation allows the tuning of the band gaps to significantly enlarge the band gaps, which can exceed 90% of the entire frequency range of interest. Experimental results show that with only three cells, the proposed beam allows considerable vibration energy attenuation within an ultra-broad frequency range including the low frequency…

The multi-snapshot, multi-frequency sparse Bayesian learning (SBL) processor is derived and its performance compared to the Bartlett, minimum variance distortionless response, and white noise constraint processors for the matched field processing application. The two-source model and data scenario of interest includes realistic mismatch implemented in the form of array tilt and data snapshots not exactly corresponding to the range-depth grid of the replica vectors. Results demonstrate that SBL behaves similar to an adaptive processor when localizing a weaker source in the presence of a stronger source, is robust to mismatch, and exhibits improved localization performance when compared to the other processors. Unlike the basis or matching pursuit methods, SBL automatically determines sparsity and its solution can be interpreted as an ambiguity surface. Because of its computational efficiency and performance, SBL is practical for applications requiring adaptive and robust processing.

Elastic wave scattering at grain boundaries in polycrystalline media can be quantified to determine microstructural properties. The amplitude drop observed for coherent wave propagation (attenuation) as well as diffuse-field scattering events have been extensively studied. In all cases, the scattering shows a clear dependence on grain size, grain shape, and microstructural texture. Models used to quantify scattering experiments are often developed assuming dependence on a single spatial length scale, usually, mean grain diameter. However, several microscopy studies suggest that most metals have a log normal distribution of grain sizes. In this study, grain size distribution is discussed within the context of previous attenuation models valid for arbitrary crystallite symmetries. Results are presented for titanium using a range of distribution means and widths assuming equiaxed grains and no preferred crystallographic orientation. The longitudinal and shear attenuations are shown to vary with respect to the frequency dependence for varying distribution widths even when the volumetric…

An inhomogeneous acoustic metamaterial lens based on spatial variation of refractive index for broadband focusing of underwater sound is reported. The index gradient follows a modified hyperbolic secant profile designed to reduce aberration and suppress side lobes. The gradient index (GRIN) lens is comprised of transversely isotropic hexagonal microstructures with tunable quasi-static bulk modulus and mass density. In addition, the unit cells are impedance-matched to water and have in-plane shear modulus negligible compared to the effective bulk modulus. The flat GRIN lens is fabricated by cutting hexagonal centimeter scale hollow microstructures in aluminum plates, which are then stacked and sealed from the exterior water. Broadband focusing effects are observed within the homogenization regime of the lattice in both finite element simulations and underwater measurements (20-40 kHz). This design approach has potential applications in medical ultrasound imaging and underwater acoustic communications.

An approximate analytical model is presented to investigate sound transmission, reflection and absorption of a rubber-like medium comprising a single layer of periodic cylindrical voids attached to a steel backing. The layer of voids is modelled as a homogeneous medium with effective material and geometric properties. A numerical model based on the finite element method is developed to validate results from the homogenization model, as well as to show further insights into the physical mechanisms associated with the system acoustic performance. Monopole resonance of the voids is shown to reduce sound transmission through the voided medium due to increased reflection, resulting in poor sound absorption around this frequency. Peaks of high sound absorption are attributed to Fabry-Pérot resonance with the frequency of the first peak derivable by a lumped spring-mass analogy. Sound absorption for a single layer of voids in a soft elastic medium with a steel backing is shown to be similar to the sound absorption in the same elastic medium but without the steel backing, for a single layer…

This study investigates the interaction between voice quality and pitch by revisiting the well-known case of Mandarin creaky voice. This study first provides several pieces of experimental data to assess whether the mechanism behind allophonic creaky voice in Mandarin is tied to tonal categories or is driven by phonetic pitch ranges. The results show that the presence of creak is not exclusively limited to tone 3, but can accompany any of the low pitch targets in the Mandarin tones; further, tone 3 is less creaky when the overall pitch range is raised, but more creaky when the overall pitch range is lowered. More importantly, tone 3 is not unique in this regard, and other tones such as tone 1 are also subject to similar variations. In sum, voice quality is quite systematically tied to F0 in Mandarin. Results from a pitch glide experiment further suggest that voice quality overall covaries with pitch height in a wedge-shaped function. Non-modal voice tends to occur when pitch production exceeds certain limits. Voice quality, thus, has the potential to enhance the perceptual…

Underwater radiated noise from merchant ships was measured opportunistically from multiple spatial aspects to estimate signature source levels and directionality. Transiting ships were tracked via the Automatic Identification System in a shipping lane while acoustic pressure was measured at the ships' keel and beam aspects. Port and starboard beam aspects were 15°, 30°, and 45° in compliance with ship noise measurements standards [ANSI/ASA S12.64 (2009) and ISO 17208-1 (2016)]. Additional recordings were made at a 10° starboard aspect. Source levels were derived with a spherical propagation (surface-affected) or a modified Lloyd's mirror model to account for interference from surface reflections (surface-corrected). Ship source depths were estimated from spectral differences between measurements at different beam aspects. Results were exemplified with a 4870 and a 10 036 twenty-foot equivalent unit container ship at 40%–56% and 87% of service speeds, respectively. For the larger ship, opportunistic ANSI/ISO broadband levels were 195 (surface-affected) and 209 (surface-corrected) dB…

Matched field processing is a generalized beamforming method that matches received array data to a dictionary of replica vectors in order to locate one or more sources. Its solution set is sparse since there are considerably fewer sources than replicas. Using compressive sensing (CS) implemented using basis pursuit, the matched field problem is reformulated as an underdetermined, convex optimization problem. CS estimates the unknown source amplitudes using the replica dictionary to best explain the data, subject to a row-sparsity constraint. This constraint selects the best matching replicas within the dictionary when using multiple observations and/or frequencies. For a single source, theory and simulations show that the performance of CS and the Bartlett processor are equivalent for any number of snapshots. Contrary to most adaptive processors, CS also can accommodate coherent sources. For a single and multiple incoherent sources, simulations indicate that CS offers modest localization performance improvement over the adaptive white noise constraint processor. SWellEx-96…

Salience describes the phenomenon by which an object stands out from a scene. While its underlying processes are extensively studied in vision, mechanisms of auditory salience remain largely unknown. Previous studies have used well-controlled auditory scenes to shed light on some of the acoustic attributes that drive the salience of sound events. Unfortunately, the use of constrained stimuli in addition to a lack of well-established benchmarks of salience judgments hampers the development of comprehensive theories of sensory-driven auditory attention. The present study explores auditory salience in a set of dynamic natural scenes. A behavioral measure of salience is collected by having human volunteers listen to two concurrent scenes and indicate continuously which one attracts their attention. By using natural scenes, the study takes a data-driven rather than experimenter-driven approach to exploring the parameters of auditory salience. The findings indicate that the space of auditory salience is multidimensional (spanning loudness, pitch, spectral shape, as well as other acoustic…

This study reconsiders the number of effective channels in contemporary cochlear implants. Subjects listened to matrix sentences with a competing talker using their clinical map (up to 22 electrodes) and reduced-channel maps using 12, 8, and 4 electrodes. Spectro-temporal modulation thresholds and reading span were measured to explore intersubject variability. Results show that speech understanding significantly improved with increasing active electrodes up to 22, particularly for subjects with better spectro-temporal resolution. These findings suggest some listeners may be able to utilize the full electrode array and may not be limited to eight channels of information as indicated in previous studies.

An adaptive procedure for controlling the signal-to-noise ratio (SNR) when rating the subjectively perceived listening effort (Adaptive Categorical Listening Effort Scaling) is described. For this, the listening effort is rated on a categorical scale with 14 steps after the presentation of three sentences in a background masker. In a first phase of the procedure, the individual SNR range for ratings from "no effort" to "extreme effort" is estimated. In the following phases, stimuli with randomly selected SNRs within this range are presented. One or two linear regression lines are fitted to the data describing subjective listening effort as a function of SNR. The results of the adaptive procedure are independent of the initial SNR. Although a static procedure using fixed, predefined SNRs produced similar results, the adaptive procedure avoided lengthy pretests for suitable SNRs and limited possible bias in the rating procedures. The adaptive procedure resolves individual differences, as well as differences between maskers. Inter-individual standard deviations are about three times as…

Perceptual experiments with infants show that they adapt their perception of speech sounds toward the categories of the native language. How do infants learn these categories? For the most part, acoustic analyses of natural infant-directed speech have suggested that phonetic categories are not presented to learners as separable clusters of sounds in acoustic space. As a step toward explaining how infants begin to solve this problem, the current study proposes that the exaggerated prosody characteristic of infant-directed speech may highlight for infants certain speech-sound tokens that collectively form more readily identifiable categories. A database is presented, containing vowel measurements in a large sample of natural American English infant-directed speech. Analyses of the vowel space show that prosodic exaggeration in infant-directed speech has the potential to support distributional vowel learning by providing the learner with a subset of "high-quality" tokens that infants might attend to preferentially. Categorization models trained on prosodically exaggerated tokens…

To provide constraints on the inversion of ocean sound speed profiles (SSPs), SSPs are often modeled using empirical orthogonal functions (EOFs). However, this regularization, which uses the leading order EOFs with a minimum-energy constraint on the coefficients, often yields low resolution SSP estimates. In this paper, it is shown that dictionary learning, a form of unsupervised machine learning, can improve SSP resolution by generating a dictionary of shape functions for sparse processing (e.g., compressive sensing) that optimally compress SSPs; both minimizing the reconstruction error and the number of coefficients. These learned dictionaries (LDs) are not constrained to be orthogonal and thus, fit the given signals such that each signal example is approximated using few LD entries. Here, LDs describing SSP observations from the HF-97 experiment and the South China Sea are generated using the K-SVD algorithm. These LDs better explain SSP variability and require fewer coefficients than EOFs, describing much of the variability with one coefficient. Thus, LDs improve the resolution…

Although much is known about the linguistic function of vowel nasality, whether contrastive (as in French) or coarticulatory (as in English), and much effort has gone into identifying potential correlates for the phenomenon, this study examines these proposed features to find the optimal acoustic feature(s) for nasality measurement. To this end, a corpus of 4778 oral and nasal vowels in English and French was collected, and data for 22 features were extracted. A series of linear mixed-effects regressions highlighted three promising features with large oral-to-nasal feature differences and strong effects relative to normal oral vowel variability: A1-P0, F1's bandwidth, and spectral tilt. However, these three features, particularly A1-P0, showed considerable variation in baseline and range across speakers and vowels within each language. Moreover, although the features were consistent in direction across both languages, French speakers' productions showed markedly stronger effects, and showed evidence of spectral tilt beyond the nasal norm being used to enhance the oral-nasal…

The acoustic waves being generated during the motion of a bubble in water near a solid boundary are calculated numerically. The open source package OpenFOAM is used for solving the Navier-Stokes equation and extended to include nonlinear acoustic wave effects via the Tait equation for water. A bubble model with a small amount of gas is chosen, the gas obeying an adiabatic law. A bubble starting from a small size with high internal pressure near a flat, solid boundary is studied. The sequence of events from bubble growth via axial microjet formation, jet impact, annular nanojet formation, torus-bubble collapse, and bubble rebound to second collapse is described. The different pressure and tension waves with their propagation properties are demonstrated.

A membrane absorber usually requires a large back cavity to achieve low-frequency sound absorption. This paper describes the design of a membrane acoustic metamaterial absorber in which magnetic negative stiffness is employed to reduce the size of the back cavity. As a baseline for the present research, analysis of a typical membrane sound absorber based on an equivalent circuit model is presented first. Then, a theoretical model is established by introducing negative stiffness into a standard absorber. It is demonstrated that a small cavity with negative stiffness can achieve the acoustic impedance of a large cavity and that the absorption peak is shifted to lower frequencies. Experimental results from an impedance tube test are also presented to validate this idea and show that negative stiffness can be employed to design compact low-frequency membrane absorbers.

The accuracy, resolution, and economic cost of near-field acoustic holography (NAH) are highly dependent on the number of spatial sampling points. Generally, higher accuracy and resolution require more spatial sampling points, which may increase the workload of measurement or the hardware cost. Compressive sensing (CS) is able to solve the underdetermined problems by utilizing the sparsity of signals, and thus it can be applied to NAH to reduce the number of spatial sampling points but at the same time provide a high-resolution reconstruction image. Based on the CS theory, this paper proposes a compressed modal equivalent point source method (CMESM). In the method, a sparse basis that is obtained from the eigen-decomposition of the power resistance matrix is introduced to compress the equivalent point source strengths, and the ℓ1–norm minimization is used to promote sparse solutions. Both numerical simulation and experimental results demonstrate the validity of the proposed CMESM and show its advantage over the existing methods when the number of spatial sampling points is reduced.

Previous research suggests that, in open-plan offices, noise complaints may be related to the high intelligibility of speech. Distraction distance, which is based on the Speech Transmission Index, can be used to objectively describe the acoustic quality of open-plan offices. However, the relation between distraction distance and perceived noise disturbance has not been established in field studies. The aim of this study was to synthesize evidence from separate studies covering 21 workplaces (N = 883 respondents) and a wide range of room acoustic conditions. The data included both questionnaire surveys and room acoustic measurements [ISO 3382-3 (2012) (International Organization for Standardization, Geneva, Switzerland]. Distraction distance, the spatial decay rate of speech, speech level at 4 m from the speaker, and the average background noise level were examined as possible predictors of perceived noise disturbance. The data were analyzed with individual participant data meta-analysis. The results show that distracting background speech largely explains the overall perception of…

Children perform more poorly than adults on a wide range of masked speech perception paradigms, but this effect is particularly pronounced when the masker itself is also composed of speech. The present study evaluated two factors that might contribute to this effect: the ability to perceptually isolate the target from masker speech, and the ability to recognize target speech based on sparse cues (glimpsing). Speech reception thresholds (SRTs) were estimated for closed-set, disyllabic word recognition in children (5-16 years) and adults in a one- or two-talker masker. Speech maskers were 60 dB sound pressure level (SPL), and they were either presented alone or in combination with a 50-dB-SPL speech-shaped noise masker. There was an age effect overall, but performance was adult-like at a younger age for the one-talker than the two-talker masker. Noise tended to elevate SRTs, particularly for older children and adults, and when summed with the one-talker masker. Removing time-frequency epochs associated with a poor target-to-masker ratio markedly improved SRTs, with larger effects for…

Otoacoustic emissions (OAEs) provide an acoustic fingerprint of the inner ear, and changes in this fingerprint may indicate changes in cochlear function arising from efferent modulation, aging, noise trauma, and/or exposure to harmful agents. However, the reproducibility and diagnostic power of OAE measurements is compromised by the variable acoustics of the ear canal, in particular, by multiple reflections and the emergence of standing waves at relevant frequencies. Even when stimulus levels are controlled using methods that circumvent standing-wave problems (e.g., forward-pressure-level calibration), distortion-product otoacoustic emission (DPOAE) levels vary with probe location by 10-15 dB near half-wave resonant frequencies. The method presented here estimates the initial outgoing OAE pressure wave at the eardrum from measurements of the conventional OAE, allowing one to separate the emitted OAE from the many reflections trapped in the ear canal. The emitted pressure level (EPL) represents the OAE level that would be recorded were the ear canal replaced by an infinite tube with…

There has been little research on the acoustic correlates of emotional expression in the singing voice. In this study, two pertinent questions are addressed: How does a singer's emotional interpretation of a musical piece affect acoustic parameters in the sung vocalizations? Are these patterns specific enough to allow statistical discrimination of the intended expressive targets? Eight professional opera singers were asked to sing the musical scale upwards and downwards (using meaningless content) to express different emotions, as if on stage. The studio recordings were acoustically analyzed with a standard set of parameters. The results show robust vocal signatures for the emotions studied. Overall, there is a major contrast between sadness and tenderness on the one hand, and anger, joy, and pride on the other. This is based on low vs high levels on the components of loudness, vocal dynamics, high perturbation variation, and a tendency for high low-frequency energy. This pattern can be explained by the high power and arousal characteristics of the emotions with high levels on these…

Killer whales (Orcinus orca) are one of the most cosmopolitan marine mammal species with potential widespread exposure to anthropogenic noise impacts. Previous audiometric data on this species were from two adult females [Szymanski, Bain, Kiehl, Pennington, Wong, and Henry (1999). J. Acoust. Soc. Am. 108, 1322-1326] and one sub-adult male [Hall and Johnson (1972). J. Acoust. Soc. Am. 51, 515-517] with apparent high-frequency hearing loss. All three killer whales had best sensitivity between 15 and 20 kHz, with thresholds lower than any odontocete tested to date, suggesting this species might be particularly sensitive to acoustic disturbance. The current study reports the behavioral audiograms of eight killer whales at two different facilities. Hearing sensitivity was measured from 100 Hz to 160 kHz in killer whales ranging in age from 12 to 52 year. Previously measured low thresholds at 20 kHz were not replicated in any individual. Hearing in the killer whales was generally similar to other delphinids, with lowest threshold (49 dB re 1 μPa) at approximately 34 kHz, good hearing…

A simulation that is perceptually indistinguishable from the corresponding real sound field could be termed authentic. Using binaural technology, such a simulation would theoretically be achieved by reconstructing the sound pressure at a listener's ears. However, inevitable errors in the measurement, rendering, and reproduction introduce audible degradations, as it has been demonstrated in previous studies for anechoic environments and static binaural simulations (fixed head orientation). The current study investigated the authenticity of individual dynamic binaural simulations for three different acoustic environments (anechoic, dry, wet) using a highly sensitive listening test design. The results show that about half of the participants failed to reliably detect any differences for a speech stimulus, whereas all participants were able to do so for pulsed pink noise. Higher detection rates were observed in the anechoic condition, compared to the reverberant spaces, while the source position had no significant effect. It is concluded that the authenticity mainly depends on how…

Binaural reproduction aims at recreating a realistic audio scene at the ears of the listener using headphones. In the real acoustic world, sound sources tend to be externalized (that is, perceived to be emanating from a source out in the world) rather than internalized (that is, perceived to be emanating from inside the head). Unfortunately, several studies report a collapse of externalization, especially with frontal and rear virtual sources, when listening to binaural content using non-individualized Head-Related Transfer Functions (HRTFs). The present study examines whether or not head movements coupled with a head tracking device can compensate for this collapse. For each presentation, a speech stimulus was presented over headphones at different azimuths, using several intermixed sets of non-individualized HRTFs for the binaural rendering. The head tracker could either be active or inactive, and the subjects could either be asked to rotate their heads or to keep them as stationary as possible. After each presentation, subjects reported to what extent the stimulus had been…

A method to characterize macroscopically homogeneous rigid frame porous media from impedance tube measurements by deterministic and statistical inversion is presented. Equivalent density and bulk modulus of the samples are reconstructed with the scattering matrix formalism, and are then linked to its physical parameters via the Johnson-Champoux-Allard-Lafarge model. The model includes six parameters, namely the porosity, tortuosity, viscous and characteristic lengths, and static flow and thermal permeabilities. The parameters are estimated from the measurements in two ways. The first one is a deterministic procedure that finds the model parameters by minimizing a cost function in the least squares sense. The second approach is based on statistical inversion. It can be used to assess the validity of the least squares estimate, but also presents several advantages since it provides valuable information on the uncertainty and correlation between the parameters. Five porous samples with a range of pore properties are tested, and the pore parameter estimates given by the proposed…

This paper describes practical integration of the remote microphone technique with a head-tracking device in a local active noise control system. The formulation is first reviewed for the optimized observation filter and nearfield pressure estimation. The attenuation performance and stability of an adaptive active headrest system combined with the remote microphone technique are then studied. The accuracy of the nearfield estimation and the effect of the head-tracking on the control performance are investigated in real-time experiments. The regularization factor of the observation filter is selected as a trade-off between its accuracy and its robustness. The integrated active headrest system is used to estimate and attenuate disturbance signals at a listener's ears from a single tonal primary source, while a commercial head-tracking device detects and provides the real-time head position to the active headrest system whose responses are updated accordingly.

The unsteady nature of wind turbine noise is a major reason for annoyance. The variation of far-field sound pressure levels is not only caused by the continuous change in wind turbine noise source levels but also by the unsteady flow field and the ground characteristics between the turbine and receiver. To take these phenomena into account, a consistent numerical technique that models the sound propagation from the source to receiver is developed. Large eddy simulation with an actuator line technique is employed for the flow modelling and the corresponding flow fields are used to simulate sound generation and propagation. The local blade relative velocity, angle of attack, and turbulence characteristics are input to the sound generation model. Time-dependent blade locations and the velocity between the noise source and receiver are considered within a quasi-3D propagation model. Long-range noise propagation of a 5 MW wind turbine is investigated. Sound pressure level time series evaluated at the source time are studied for varying wind speeds, surface roughness, and ground…

The study examined whether the benefit of deactivating stimulation sites estimated to have broad neural excitation was attributed to improved spectral resolution in cochlear implant users. The subjects' spatial neural excitation pattern was estimated by measuring low-rate detection thresholds across the array [see Zhou (2016). PLoS One 11, e0165476]. Spectral resolution, as assessed by spectral-ripple discrimination thresholds, significantly improved after deactivation of five high-threshold sites. The magnitude of improvement in spectral-ripple discrimination thresholds predicted the magnitude of improvement in speech reception thresholds after deactivation. Results suggested that a smaller number of relatively independent channels provide a better outcome than using all channels that might interact.

This paper describes the extension of a wave and finite element (WFE) method to the prediction of noise transmission through, and radiation from, infinite panels. The WFE method starts with a conventional finite element model of a small segment of the panel. For a given frequency, the mass and stiffness matrices of the segment are used to form the structural dynamic stiffness matrix. The acoustic responses of the fluids surrounding the structure are modelled analytically. The dynamic stiffness matrix of the segment is post-processed using periodic structure theory, and coupled with those of the fluids. The total dynamic stiffness matrix is used to obtain the response of the medium to an incident acoustic pressure. Excitation of the structure by oblique plane waves and a diffuse sound field are considered. The response to structural excitation and the consequent radiation are determined. Since the size of the WFE model is small, computational times are small. Various example applications are presented to illustrate the approach, including a thin isotropic panel, an antisymmetric,…

During the 1980 s, acoustic studies of Upper Palaeolithic imagery in French caves-using the technology then available-suggested a relationship between acoustic response and the location of visual motifs. This paper presents an investigation, using modern acoustic measurement techniques, into such relationships within the caves of La Garma, Las Chimeneas, La Pasiega, El Castillo, and Tito Bustillo in Northern Spain. It addresses methodological issues concerning acoustic measurement at enclosed archaeological sites and outlines a general framework for extraction of acoustic features that may be used to support archaeological hypotheses. The analysis explores possible associations between the position of visual motifs (which may be up to 40 000 yrs old) and localized acoustic responses. Results suggest that motifs, in general, and lines and dots, in particular, are statistically more likely to be found in places where reverberation is moderate and where the low frequency acoustic response has evidence of resonant behavior. The work presented suggests that an association of the location…

The purpose of this work is to systematically study the effect of the throat and the pore sizes on the sound absorbing properties of open-cell foams. The three-dimensional idealized unit cell used in this work enables to mimic the acoustical macro-behavior of a large class of cellular solid foams. This study is carried out for a normal incidence and also for a diffuse field excitation, with a relatively large range of sample thicknesses. The transport and sound absorbing properties are numerically studied as a function of the throat size, the pore size, and the sample thickness. The resulting diagrams show the ranges of the specific throat sizes and pore sizes where the sound absorption grading is maximized due to the pore morphology as a function of the sample thickness, and how it correlates with the corresponding transport parameters. These charts demonstrate, together with typical examples, how the morphological characteristics of foam could be modified in order to increase the visco-thermal dissipation effects.

In recent years acoustic black holes (ABHs) have been developed and demonstrated as an effective method for developing lightweight, high loss structures for noise and vibration control. ABHs employ a local thickness change to tailor the speed and amplitude of flexural bending waves and create concentrated regions of high strain energy which can be effectively dissipated through conventional damping treatments. These regions act as energy sinks which allow for effective broadband vibration absorption with minimal use of applied damping material. This, combined with the reduced mass from the thickness tailoring, results in a treated structure with higher loss and less mass than the original. In this work, the transmission loss (TL) of plates with embedded ABHs was investigated using experimental and numerical methods in order to assess the usefulness of ABH systems for TL applications. The results demonstrated that damped ABH plates offer improved performance compared to a uniform plate despite having less mass. The result will be useful for applying ABHs and ABH systems to practical…

The inter-aural time difference (ITD) is a fundamental cue for human sound localization. Over the past decades several methods have been proposed for its estimation from measured head-related impulse response (HRIR) data. Nevertheless, inter-method variations in ITD calculation have been found to exceed the known just noticeable differences (JNDs), leading to possible perceptible artifacts in virtual binaural auditory scenes, when personalized HRIRs are being used. In the absence of an objective means for validating ITD estimations, this paper examines which methods lead to the most perceptually relevant results. A subjective lateralization study compared objective ITDs to perceptually evaluated inter-aural pure delay offsets. Results clearly indicate the first-onset threshold detection method, using a low relative threshold of -30 dB, applied on 3 kHz low-pass filtered HRIRs as consistently the most perceptually relevant procedure across various metrics. Several alternative threshold values and methods based on the maximum or centroid of the inter-aural cross correlation of…

This document reviews the history, development, and use of auditory weighting functions for noise impact assessment in humans and marine mammals. Advances from the modern era of electroacoustics, psychophysical studies of loudness, and other related hearing studies are reviewed with respect to the development and application of human auditory weighting functions, particularly A-weighting. The use of auditory weighting functions to assess the effects of environmental noise on humans-such as hearing damage-risk criteria-are presented, as well as lower-level effects such as annoyance and masking. The article also reviews marine mammal auditory weighting functions, the development of which has been fundamentally directed by the objective of predicting and preventing noise-induced hearing loss. Compared to the development of human auditory weighting functions, the development of marine mammal auditory weighting functions have faced additional challenges, including a large number of species that must be considered, a lack of audiometric information on most species, and small sample sizes…

A database of acoustic radiation patterns was recorded, modeled, and analyzed for 41 modern or authentic orchestral musical instruments. The generation of this database included recordings of each instrument over the entire chromatic tone range in an anechoic chamber using a surrounding spherical microphone array. Acoustic source centering was applied in order to align the acoustic center of the sound source to the physical center of the microphone array. The acoustic radiation pattern is generated in the spherical harmonics domain at each harmonic partial of each played tone. An analysis of the acoustic radiation pattern complexity has been performed in terms of the number of excitation points using the centering algorithm. The database can be used both for studying the radiation of musical instruments itself, as well as for the implementation of radiation patterns in room acoustical simulations and auralization in order to obtain a spatial excitation of the room closer to reality.

Early language input is far from uniform, even among children learning the same language. For instance, while some children are exposed to a single accent in their linguistic environment, others have routine exposure to multiple accents. Nonetheless, few studies have taken this into account when examining word recognition, and none has examined this issue in infants prior to the emergence of phonological constancy (∼18 months). This study demonstrates that daily exposure to multiple accents strongly impacts infants' performance in a laboratory word form recognition task. Accent variability in the input thus needs to be carefully considered when studying speech development.

Circular differential microphone arrays (CDMAs) have been extensively studied in speech and audio applications for their steering flexibility, potential to achieve frequency-invariant directivity patterns, and high directivity factors (DFs). However, CDMAs suffer from both white noise amplification and deep nulls in the DF and in the white noise gain (WNG) due to spatial aliasing, which considerably restricts their use in practical systems. The minimum-norm filter can improve the WNG by using more microphones than required for a given differential array order; but this filter increases the array aperture (radius), which exacerbates the spatial aliasing problem and worsens the nulls problem in the DF. Through theoretical analysis, this research finds that the nulls of the CDMAs are caused by the zeros in the denominators of the filters' coefficients, i.e., the zeros of the Bessel function. To deal with both the white noise amplification and deep nulls problems, this paper develops an approach that combines different rings of microphones together with appropriate radii. The resulting…

An anatomically based three-dimensional finite-element human middle-ear (ME) model is used to test the sensitivity of ME sound transmission to tympanic-membrane (TM) material properties. The baseline properties produce responses comparable to published measurements of ear-canal input impedance and power reflectance, stapes velocity normalized by ear-canal pressure (PEC), and middle-ear pressure gain (MEG), i.e., cochlear-vestibule pressure (PV) normalized by PEC. The mass, Young's modulus (ETM), and shear modulus (GTM) of the TM are varied, independently and in combination, over a wide range of values, with soft and bony TM-annulus boundary conditions. MEG is recomputed and plotted for each case, along with summaries of the magnitude and group-delay deviations from the baseline over low (below 0.75 kHz), mid (0.75–5 kHz), and high (above 5 kHz) frequencies. The MEG magnitude varies inversely with increasing TM mass at high frequencies. Increasing ETM boosts high frequencies and attenuates low and mid frequencies, especially with a bony TM annulus and when GTM varies in proportion to…

Compressive beamforming realizes the direction-of-arrival (DOA) estimation and strength quantification of acoustic sources by solving an underdetermined system of equations relating microphone pressures to a source distribution via compressive sensing. The conventional method assumes DOAs of sources to lie on a grid. Its performance degrades due to basis mismatch when the assumption is not satisfied. To overcome this limitation for the measurement with plane microphone arrays, a two-dimensional grid-free compressive beamforming is developed. First, a continuum based atomic norm minimization is defined to denoise the measured pressure and thus obtain the pressure from sources. Next, a positive semidefinite programming is formulated to approximate the atomic norm minimization. Subsequently, a reasonably fast algorithm based on alternating direction method of multipliers is presented to solve the positive semidefinite programming. Finally, the matrix enhancement and matrix pencil method is introduced to process the obtained pressure and reconstruct the source distribution. Both…

The popularity of plane-wave transmits at multiple transmit angles for synthetic transmit aperture (or coherent compounding) has spawned a number of adaptations and new developments of ultrasonic imaging. However, the coherence properties of backscattered signals with plane-wave transmits at different angles are unknown and may impact a subset of these techniques. To provide a framework for the analysis of the coherence properties of such signals, this article introduces the angular coherence theory in medical ultrasound imaging. The theory indicates that the correlation function of such signals forms a Fourier transform pair with autocorrelation function of the receive aperture function. This conclusion can be considered as an extended form of the van Cittert Zernike theorem. The theory is validated with simulation and experimental results obtained on speckle targets. On the basis of the angular coherence of the backscattered wave, a new short-lag angular coherence beamformer is proposed and compared with an existing spatial-coherence-based beamformer. An application of the theory…

Although stiffness conditions in the multi-layered vocal folds are generally considered to have a large impact on voice production, their specific role in controlling vocal fold vibration and voice acoustics is unclear. Using a three-dimensional body-cover continuum model of phonation, this study shows that changes in vocal fold stiffness have a large effect on F0 and the means and amplitudes of the glottal area and flow rate. However, varying vocal fold stiffness, particularly along the anterior-posterior direction, has a much smaller effect on the closed quotient, vertical phase difference, and the spectral shape of the output acoustics, which are more effectively controlled by changes in the vertical thickness of the medial surface. These results suggest that although changes in vocal fold stiffness are often correlated with production of different voice types, there is no direct cause-effect relation between vocal fold stiffness and voice types, and the correlation may simply result from the fact that both vocal fold stiffness and geometry are regulated by the same set of…

The synthesis of binaural signals from spherical microphone array recordings has been recently proposed. The limited spatial resolution of the reproduced signal due to order-limited reproduction has been previously investigated perceptually, showing spatial perception ramifications, such as poor source localization and limited externalization. Furthermore, this spatial order limitation also has a detrimental effect on the frequency content of the signal and its perceived timbre, due to the rapid roll-off at high frequencies. In this paper, the underlying causes of this spectral roll-off are described mathematically and investigated numerically. A digital filter that equalizes the frequency spectrum of a low spatial order signal is introduced and evaluated. A comprehensive listening test was conducted to study the influence of the filter on the perception of the reproduced sound. Results indicate that the suggested filter is beneficial for restoring the timbral composition of order-truncated binaural signals, while conserving, and even improving, some spatial properties of the signal.