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We have investigated the unique interface states (NI) generated by NO annealing at the SiO2/SiC interfaces by low-temperature conductance measurements, which is more suitable for characterization of very fast interface states than high-frequency conductance measurements at room temperature. Although only a part of the NI states can be evaluated by measurements at room temperature, the whole picture of the NI states, especially near the conduction band edge (0.07 eV ≤EC--ET), has been revealed by the low temperature measurements. The NI peak was present at the interface even without NO annealing. The NI density increased with NO annealing temperature. The NI density at the energy levels shallower than 0.2 eV exceeded 1012cm-2eV-1 after NO annealing. The capture cross section of the NI states is uniquely larger than that of conventional interface states. [ABSTRACT FROM AUTHOR]

We present, in our knowledge, the first theoretical demonstration of the possibility to trap and manipulate particles in three dimensions with the radiation pressure exerted by a single acoustical beam. Numerical examples demonstrate that single-beam acoustical tweezers operating in three dimensions are feasible with a large variety of materials and may widely extend the range of forces and operation regions that are currently available with optical tweezers. To do so, a method to model the focusing properties of acoustical beams with complex wavefronts using a spherical transducer is proposed. Then, the radiation forces exerted by various beams going from the classical vortex to the high radial degree spherical vortex beam that we introduce here are studied. While the first is shown to trap moderately small particles, the latter will stiffly trap large solid spheres in three dimensions. Even though this demonstration is carried out using a formalism suited to acoustics, it is easily applicable to trap non-transparent particles with optical tweezers that remain an issue. [ABSTRACT FROM AUTHOR]

The occurrence of acoustic cavitation is incorporating a multitude of interdependent effects that strongly depend on the bubble size. Therefore, bubble size control would be beneficial for biological and industrial processes that rely on acoustic cavitation. A pulsed acoustic field can result in bubble size control and the repeated dissolution and reactivation ('recycling') of potentially active bubbles. As a consequence, a pulsed field can strongly enhance cavitation activity. In this paper, we present a modified methodology for the evaluation of the active bubble size distribution by means of a combination of cavitation noise measurements and ultrasonic pulsing. The key component of this modified methodology is the definition of an upper size limit, below which bubbles-in between subsequent pulses-have to dissolve, in order to be sustainably recycled. This upper limit makes it possible to explain and link the enhancement of cavitation activity to a bubble size distribution. The experimentally determined bubble size distributions for different power densities are interpreted in the frame of numerical calculations of the oscillatory responses of the bubbles to the intermittent driving sound field. The distributions are found to be shaped by the size dependent interplay between bubble pulsations, rectified diffusion, coalescence, and the development of parametrically amplified shape instabilities. Also, a phenomenological reactivation-deactivation model is proposed to explain and quantify the observed enhancement of cavitation activity under pulsed, with respect to continuous sonication. In this model, the pulse-duration determines the magnitude of the reactivation of partially dissolved bubbles and the deactivation of activated bubbles by coalescence. It is shown that the subsequent recycling of previously active bubbles leads to an accumulation of cavitation activity, which saturates after a certain number of pulses. The model is fitted to the experimental data for the cavitation activity measured by means of ultraharmonic cavitation noise as a function of the pulse duration. Measurements of the development of the cavitation noise and sonochemiluminescence over a sequence of pulses for different pulse durations and separations confirm the general validity of the proposed model. Size distributions of the larger, inactive bubbles that were extracted from High-speed images of the cavitation field, relate the deactivation of activated bubbles by coalescence to the increase in volume concentrations of larger bubbles as observed by others. [ABSTRACT FROM AUTHOR]

For over two decades, passive acoustic monitoring (PAM) methods have been successfully employed around the world for studying aquatic megafauna. PAM-driven studies in Indian waters have so far been relatively very scarce. Furthermore, cetacean populations inhabiting the north western Indian Ocean are far less studied than those in many other regions around the world. This work likely constitutes the first systematic study of the vocal repertoire of humpback whales (Megaptera novaeangliae) at a near-shore site along the western coast of India. Analysis of the observed vocalizations provides an insight into the behaviour of the species. This is significant as it assists in developing a better understanding of the habitat use of the non-migratory Arabian Sea humpback whale population. In contrast, other breeding populations such as those around the North Atlantic, South Pacific and Australia have been relatively well studied. Underwater passive acoustic data were collected during March 2017 using an autonomous logger at a shallow-water site off the eastern edge of Grande Island off the coast of Goa. Humpback whale vocalizations were found to occur over multiple days in the recordings. Time–frequency contours of individual units of vocalization were extracted with the aid of an automatic detection technique and the characteristics of the units were measured. Further, successive units were analysed for formation of phrases and themes. Reconstruction of putative songs from the identified units and themes was not possible due to the limitations imposed by the nature of data collection. Detailed analyses of units, phrases and themes are presented. [ABSTRACT FROM AUTHOR]

ABSTRACT. Ocean ambient noise is a crucial habitat feature for marine animals because it represents the lower threshold of their acoustically active space. Ambient noise is affected by noise from [...]

The article thanks journal peer reviewers from 2017 for the journals "Journal of the Acoustical Society of America" ("JASA") and "JASA Express Letters" ("JASA-EL").

The article presents the December 2017 issue of the periodical "Acoustics Australia." Topics include research articles on topics like sound quality experiments in a student hostel with newly designed sonic crystal window, and design and validation of a simulator tool useful for designers and policy makers in urban sound planning; news reports related to Australian Acoustical Society (AAS); and an obituary for professor Neville Fletcher.

Listeners can judge the overall loudness of time-varying sounds quite easily, i.e., assign a single value that represents the loudness of the entire sound. This holds even if the duration is long and the judgment includes memory effects. Different metrics for calculating overall loudness have been developed. They agree that overall loudness is higher than the mean of loudness over time. Percentiles like the N5, the loudness being exceeded 5% of the time, are adopted by ISO 532-1. In the present study the concept of an energy mean known from level measurements (ISO 1996-1) was applied to the loudness domain. This equivalent continuous loudness level, LLP, was compared to the N5 using a set of real-world sounds that was orthogonal between the two metrics over a wide dynamic range of 30 phon. Cross-modality matching with line length was used in three experiments with a focus on either the overall judgment of loudness, continuous judgment while a sound was played, or both. The LLP showed considerably higher correlations with overall judgments than N5. Comparing continuous instantaneous judgment with calculated instantaneous loudness suggests that the participants might have focused on the sounds’ prominent portions. [ABSTRACT FROM AUTHOR]

A unified framework is proposed for analysis and synthesis of two-dimensional spatial sound field in reverberant environments. In the sound field analysis (SFA) phase, an unbaffled 24-element circular microphone array is utilized to encode the sound field based on the plane-wave decomposition. Depending on the sparsity of the sound sources, the SFA stage can be implemented in two manners. For sparse-source scenarios, a one-stage algorithm based on compressive sensing algorithm is utilized. Alternatively, a two-stage algorithm can be used, where the minimum power distortionless response beamformer is used to localize the sources and Tikhonov regularization algorithm is used to extract the source amplitudes. In the sound field synthesis (SFS), a 32-element rectangular loudspeaker array is employed to decode the target sound field using pressure matching technique. To establish the room response model, as required in the pressure matching step of the SFS phase, an SFA technique for nonsparse-source scenarios is utilized. Choice of regularization parameters is vital to the reproduced sound field. In the SFS phase, three SFS approaches are compared in terms of localization performance and voice reproduction quality. Experimental results obtained in a reverberant room are presented and reveal that an accurate room response model is vital to immersive rendering of the reproduced sound field. [ABSTRACT FROM AUTHOR]

Acoustic signal production is affected by allometric relationships, by which the larger the animal, the lower its call frequency. In this paper, three evolutionary acoustic hypotheses were tested: the Signal-to-Noise Ratio Hypothesis (SNRH), in which evolution maximizes call ranges by increasing the signal-to-noise ratio; the Stimulus Threshold Hypothesis (STH), in which evolution maximizes the range of a specific signal threshold; and the Body Size Hypothesis (BSH), in which the emission of long wavelengths is enabled by body size. Three spectral metrics were measured, Dominant Frequency (FDOM), Minimum Fundamental Frequencies (FFMIN), and Maximum Fundamental Frequencies (FFMAX) of Neotropical Parrots, New World Doves, Woodcreepers, Tinamous, and Thrushes. A Ranged Major Axis (RMA) regression showed that body mass is significantly correlated with all of the spectral parameters in Parrots, Doves, and Woodcreepers, but only with the fundamental frequencies of Tinamous. The FDOM of Parrots corroborated the SNRH. The FFMIN of Woodcreepers and Tinamous corroborated the SNRH and BSH. The FFMAX of Parrots corroborated the STH and BSH. Those acoustic hypotheses could shed light on the evolutionary processes involved in avian communication, although results indicate that these depend on the taxa and spectral parameters considered. [ABSTRACT FROM AUTHOR]

Most derivations of acoustic wave equations involve ensuring that causality is satisfied. Here, the consequences of also requiring that the medium should be passive are explored. This is a stricter criterion than causality for a linear system and implies that there are restrictions on the relaxation modulus and its first few derivatives. The viscous and relaxation models of acoustics satisfy passivity and have restrictions on not only a few, but all derivatives of the relaxation modulus. These models are described as a system of springs and dampers with positive parameters and belong to the important class of completely monotone systems. It is shown here that the attenuation as a function of frequency for such media has to increase slower than a linear function. Likewise, the phase velocity has to increase monotonically. This gives criteria on which one may judge whether a proposed wave equation is passive or not, as illustrated by comparing two different versions of the viscous wave equation. [ABSTRACT FROM AUTHOR]

The waveguide invariant β is typically discussed in terms of either a range-independent environment or a range-dependent environment under the adiabatic approximation, with few studies considering the effects of mode coupling. In this work, how internal solitary waves (ISWs) affect the waveguide invariant is investigated, and it is shown that mode-coupling effects introduce many additional components in the acoustic interference intensity. It is found that the striation slope and value of β for these additional components are determined not only by the acoustic modal dispersion, but are also dependent on the position where the mode coupling occurs. This can lead to a very complicated acoustic interference pattern and result in multiple peaks in the distribution of β. The sensitivity of β to the parameters of ISWs, such as amplitude, horizontal scale, and position, is analyzed. It is found that although all parameters can affect the energy of the peaks, only the position of the internal wave has an obvious impact on the peak values. This indicates that the peak values of β can be utilized for monitoring the position of the internal wave. [ABSTRACT FROM AUTHOR]

A spherical microphone array has proved effective in reconstructing an enclosed sound field by a superposition of spherical wave functions in Fourier domain. It allows successful reconstructions surrounding the array, but the accuracy will be degraded at a distance. In order to extend the effective reconstruction to the entire cavity, a plane-wave basis in space domain is used owing to its non-decaying propagating characteristic and compared with the conventional spherical wave function method in a low frequency sound field within a cylindrical cavity. The sensitivity to measurement noise, the effects of the numbers of plane waves, and measurement positions are discussed. Simulations show that under the same measurement conditions, the plane wave function method is superior in terms of reconstruction accuracy and data processing efficiency, that is, the entire sound field imaging can be achieved by only one time calculation instead of translations of local sets of coefficients with respect to every measurement position into a global one. An experiment was conducted inside an aircraft cabin mock-up for validation. Additionally, this method provides an alternative possibility to recover the coefficients of high order spherical wave functions in a global coordinate system without coordinate translations with respect to local origins. [ABSTRACT FROM AUTHOR]

This study compared psychoacoustic reverberance parameters to each other, as well as to reverberation time (RT) and early decay time (EDT) under various acoustic conditions. The psychoacoustic parameters were loudness-based RT (TN), loudness-based EDT [EDTN; Lee, Cabrera, and Martens, J. Acoust. Soc. Am. 131, 1194–1205 (2012a)], and parameter for reverberance [PREV; van Dorp Schuitman, de Vries, and Lindau., J. Acoust. Soc. Am. 133, 1572–1585 (2013)]. For the comparisons, a wide range of sound pressure levels (SPLs) from 20 dB to 100 dB and RTs from 0.5 s to 5.0 s were evaluated, and two sets of subjective data from the previous studies were used for the cross-validation and comparison. Results of the comparisons show that the psychoacoustic reverberance parameters provided better matches to reverberance than RT and EDT; however, the performance of these psychoacoustic reverberance parameters varied with the SPL range, the type of audio sample, and the reverberation conditions. This study reveals that PREV is the most relevant for estimating a relative change in reverberance between samples when the SPL range is small, while EDTN is useful in estimating the absolute reverberance. This study also suggests the use of PREV and EDTN for speech and music samples, respectively. [ABSTRACT FROM AUTHOR]

This report is an addendum to a previous report by Burns [(2009). J. Acoust. Soc. Am. 125, 3166–3176] that measured spontaneous otoacoustic emissions (SOAEs) in 18 subjects, whose ages at the time of initial measurement ranged from 6 to 42 yr, for follow-up periods of up to 19.5 yr. The major finding of that report was that the frequencies of all SOAEs, in all subjects, declined over time, with an average decline of 0.25% per year. In this report seven SOAEs in the oldest subject were measured for an additional 13.7 yr, for a total follow-up of 33 yr, to age 75. [ABSTRACT FROM AUTHOR]

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 components. A multiple discriminant analysis yields classification accuracy greatly exceeding chance level, confirming the reliability of the acoustic patterns. [ABSTRACT FROM AUTHOR]

Numerical computations of the Navier-Stokes equations governing acoustic propagation are performed to investigate infrasound propagation in the troposphere and into acoustic shadow zones. An existing nonlinear finite-difference, time-domain (FDTD) solver that constrains input sound speed models to be axisymmetric is expanded to allow for advection and rigid, stair-step topography. The FDTD solver permits realistic computations along a given azimuth. It is applied to several environmental models to examine the effects of nonlinearity, topography, advection, and two-dimensional (2D) variations in wind and sound speeds on the penetration of infrasound into shadow zones. Synthesized waveforms are compared to a recording of a rocket motor fuel elimination event at the Utah Test and Training Range. Results show good agreement in the amplitude, duration, and spectra of synthesized and recorded waveforms for propagation through 2D atmospheric models whether or not topography, advection, or nonlinearity is explicitly included. However, infrasound propagation through a one-dimensional, range-averaged, atmospheric model yields waveforms with lower amplitudes and frequencies, suggesting that small-scale atmospheric variability causes significant scatter within the troposphere, leading to enhanced infrasound penetration into shadow zones. Thus, unresolved fine-scale atmospheric dynamics are not required to explain infrasound propagation into shadow zones. [ABSTRACT FROM AUTHOR]

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 comprehensive the spectral cues are provided by the audio content, and the amount of reverberation, whereas the source position plays a minor role. This is confirmed by a broad qualitative evaluation, suggesting that remaining differences mainly affect the tone color rather than the spatial, temporal or dynamical qualities. [ABSTRACT FROM AUTHOR]

Sound transmission and resulting airway wall vibration in a complex multiscale viscoelastic model of the subglottal bronchial tree was calculated using a modified one-dimensional (1D) branching acoustic waveguide approach. This is an extension of previous work to enable use of complex airway trees that are partially derived from subject-specific medical images, without the need for self-similarity in the geometric structure. The approach was validated numerically for simplified airway geometries, as well as experimentally by comparison to previous studies. A comprehensive conducting airway tree with about 60 000 branches was then modified to create fibrotic, bronchoconstrictive, and pulmonary infiltrate conditions. The fibrotic case—systemic increase in soft tissue stiffness—increased the Helmholtz resonance frequency due to the increased acoustic impedance. Bronchoconstriction, with geometric changes in small conducting airways, decreased acoustic energy transmission to the peripheral airways due in part to the increased impedance mismatch between airway orders. Pulmonary infiltrate significantly altered the local acoustic field in the affected lobe. Calculation of acoustic differences between healthy versus pathologic cases can be used to enhance the understanding of vibro-acoustic changes correlated to pathology, and potentially provide improved tools for the diagnosis of pulmonary diseases that uniquely alter the acoustics of the airways. [ABSTRACT FROM AUTHOR]

A corpus of recordings of deaf speech is introduced. Adults who were pre- or post-lingually deafened as well as those with normal hearing read standardized speech passages totaling 11 h of .wav recordings. Preliminary acoustic analyses are included to provide a glimpse of the kinds of analyses that can be conducted with this corpus of recordings. Long term average speech spectra as well as spectral moment analyses provide considerable insight into differences observed in the speech of talkers judged to have low, medium, or high speech intelligibility. [ABSTRACT FROM AUTHOR]

This article seeks to explore the proposed notion of ‘context-based composition’ by examining the nature of ‘real-world’ context. It does this by studying the way in which listeners interpret sounds, working towards a deeper understanding of what it is that we mean by ‘real-world’ sound and context-based composition. These discussions are then utilised to explore the concept of what it means to compose context-based works and suggests that new potentials are opened up by a closer examination of the definition of context-based composition, one which liberates itself from a concern over an absolute physical nature of sounds and which embraces the use of both abstract and referential sounds. This journey highlights the importance of memory and experience within processes of interpretation and the creation of context-based compositions, and questions divisions between the virtual and the real. [ABSTRACT FROM PUBLISHER]

In this article I outline the idea of an empirical/experiential reconnection to the natural non-human world through the practice of deep listening. I believe that the aesthetic experience is central to a more ecological positioning of the human being on earth and that aesthetic experience should involve a ‘rewilding of the ear’. To discuss this concept, I build an argument from Edgard Varèse’s music as ‘organised sound’ and approach it from a perceptual point of view. This leads to the discussion of other concepts, such as David Dunn’s ‘grief of incommunicability’ (Dunn 1997) and Jean-François Augoyard and Henry Torgue’s ‘sharawadgi effect’ (Bick 2008). Further to this I discuss parallels between Truax’s continuum (Speech–Music–Soundscape) and Peirce’s semiotic system. Taking points from these theories, we can discuss the possibility of the re-tuning of our ears to the wider sound palette of the world. I consider George Monbiot’s concepts of ‘rewilding’ and ‘rewilding of the human life’ (Monbiot 2014), in order to create a parallel to our relationship with the soundscape. [ABSTRACT FROM PUBLISHER]

This study examines acoustic features of speech production in speakers of Mandarin with Parkinson's disease (PD) and relates them to intelligibility outcomes. Data from 11 participants with PD and 7 controls are compared on several acoustic measures. In speakers with PD, the strength of association between these measures and intelligibility is investigated. Speakers with PD exhibited significant differences in fundamental frequency, pitch variation, vowel space, and rate relative to controls. However, in contrast to the English studies, speech rate was consistently slow and most strongly correlated with intelligibility. Thus, acoustic cues that strongly influence intelligibility in PD may vary cross-linguistically. [ABSTRACT FROM AUTHOR]

Purpose: The reading deficit for people with dyslexia is typically associated with linguistic, memory, and perceptual-discrimination difficulties, whose relation to reading impairment is disputed. We proposed that automatic detection and usage of serial sound regularities for individuals with dyslexia is impaired (anchoring deficit hypothesis), leading to the formation of less reliable sound predictions. Agus, Carrión- Castillo, Pressnitzer, and Ramus, (2014) reported seemingly contradictory evidence by showing similar performance by participants with and without dyslexia in a demanding auditory task that contained task-relevant regularities. To carefully assess the sensitivity of participants with dyslexia to regularities of this task, we replicated their study. Method: Thirty participants with and 24 without dyslexia performed the replicated task. On each trial, a 1-s noise stimulus was presented. Participants had to decide whether the stimulus contained repetitions (was constructed from a 0.5-s noise segment repeated twice) or not. It is implicit in this structure that some of the stimuli with repetitions were themselves repeated across trials. We measured the ability to detect within-noise repetitions and the sensitivity to cross-trial repetitions of the same noise stimuli. Results: We replicated the finding of similar mean performance. However, individuals with dyslexia were less sensitive to the cross-trial repetition of noise stimuli and tended to be more sensitive to repetitions in novel noise stimuli. Conclusion: These findings indicate that online auditory processing for individuals with dyslexia is adequate but their implicit retention and usage of sound regularities is indeed impaired. [ABSTRACT FROM AUTHOR]

Purpose: Noise reduction and dynamic-range compression are generally applied together in hearing aids but may have opposite effects on amplification. This study evaluated the acoustical and perceptual effects of separate and combined processing of noise reduction and compression. Design: Recordings of the output of 4 hearing aids for speech in babble noise at +4 dB signal-to-noise ratio were used in 3 experiments: (a) acoustical measurements to determine the influence of processing on speech and noise levels; (b) perceptual measurements to determine the detectability of processing differences for 16 listeners with hearing impairment; and (c) perceptual measurements to determine the effect of processing on speech intelligibility, noise annoyance, speech naturalness, and overall preference. Results: Noise reduction and compression processing differed between hearing aids. The combined processing (noise reduction with compression) most strongly reduced noise and speech levels. The combined processing was detectably different between hearing aids, but compression processing alone was not. The combined processing did not influence speech intelligibility. Preference for combined processing was lower than previously observed for noise reduction without compression. Conclusions: Differences in processing between hearing aids are perceptually salient. The effect of compression should be taken into account during the development and evaluation of hearing aid noise reduction., Noise reduction and dynamic-range compression are common features in modern hearing aids. The role of noise reduction is to reduce hearing aid gain for background noises and to preserve the [...]

Purpose: The acoustic measure relative fundamental frequency (RFF) was investigated as a potential objective measure to track variations in vocal effort within and across individuals. Method: Twelve speakers with healthy voices created purposeful modulations in their vocal effort during speech tasks. RFF and an aerodynamic measure of vocal effort, the ratio of sound pressure level to subglottal pressure level, were estimated from the aerodynamic and acoustic signals. Twelve listeners also judged the speech samples for vocal effort using the visual sort and rate method. Results: Relationships between RFF and both the aerodynamic and perceptual measures of vocal effort were weak across speakers ([R.sup.2] = .06-.26). Within speakers, relationships were variable but much stronger on average ([R.sup.2] = .45-56). Conclusions: RFF showed stronger relationships between both the aerodynamic and perceptual measures of vocal effort when examined within individuals versus across individuals. Future work is necessary to establish these relationships in individuals with voice disorders across the therapeutic process., Vocal hyperfunction is defined as 'conditions of abuse and/or misuse of the vocal mechanism due to excessive and/or 'imbalanced' muscular forces' (Hillman, Holmberg, Perkell, Walsh, & Vaughan, 1989, p. 373), [...]

Purpose: The objective of the study was to investigate the influence of noise (energetic) and speech (energetic plus informational) maskers on the head shadow (HS), squelch (SQ), and binaural summation (SU) effect in bilateral and bimodal cochlear implant (CI) users. Method: Speech recognition was measured in the presence of either a competing talker or modulated speech-shaped noise in 10 bimodal and 10 bilateral adult CI users. HS, SQ, and SU effects were calculated. The interfering signals were manipulated with respect to F0 to consider the influence of different speaker voices. Results: The effects HS, SQ, and SU differed depending on the type of masker. A detailed analysis of errors was used to dissociate energetic and informational masking effects. The analysis showed a release from energetic than from informational masking. Conclusion: Noise interferers are not sufficient to reflect difficulties experienced with speech understanding in noise for bilateral and bimodal CI users. Key Words: bilateral hearing, bimodal hearing, hearing aid, cochlear implant, informational masking, fundamental frequency, To communicate successfully in a listening situation with more than one talker, listeners must be able to attend selectively to a target talker while ignoring other competing talkers. In a [...]

The nonlinear self-refraction of acoustic pulsed beams, which include shock fronts, is studied. The medium of sound propagation is a gas where thermodynamically non-equilibrium processes take place, such as exothermic chemical reaction or excitation of vibrational degrees of a molecule's freedom. Comparative analysis of the features of sound propagation over gases where pure nonlinear attenuation of the shock wave occurs, and gases with non-equilibrium and equilibrium processes, is performed. PACS Nos.: 43.25, 43.35.Fj. Nous etudions l'auto-refraction non lineaire d'un faisceau acoustique pulse qui inclut une onde de choc. Le milieu de propagation du son est un gaz dans lequel se produisent des transformations thermodynamiquement hors d'equilibre, comme des reactions chimiques exothermiques ou l'excitation des degres de liberte vibrationnels des molecules. Nous completons avec une analyse comparative des caracteristiques de propagation du son dans des gaz ou l'onde de choc est soumise a une pure attenuation non lineaire et dans des gaz ou on trouve des transformations hors d'equilibre et a l'equilibre. [Traduit par la Redaction], 1. Introduction There are different kinds of self-action of sound, which have been observed in experiments, among them nonlinear self-action and inertial types of self-action. The first kind reflects the [...]

Purpose: This study systematically assessed the effects of select linear predictive coding (LPC) analysis parameter manipulations on vowel formant measurements for diverse speaker groups using 4 trademarked Speech Acoustic Analysis Software Packages (SAASPs): CSL, Praat, TF32, and WaveSurfer. Method: Productions of 4 words containing the corner vowels were recorded from 4 speaker groups with typical development (male and female adults and male and female children) and 4 speaker groups with Down syndrome (male and female adults and male and female children). Formant frequencies were determined from manual measurements using a consensus analysis procedure to establish formant reference values, and from the 4 SAASPs (using both the default analysis parameters and with adjustments or manipulations to select parameters). Smaller differences between values obtained from the SAASPs and the consensus analysis implied more optimal analysis parameter settings. Results: Manipulations of default analysis parameters in CSL, Praat, and TF32 yielded more accurate formant measurements, though the benefit was not uniform across speaker groups and formants. In WaveSurfer, manipulations did not improve formant measurements. Conclusions: The effects of analysis parameter manipulations on accuracy of formant-frequency measurements varied by SAASP, speaker group, and formant. The information from this study helps to guide clinical and research applications of SAASPs. [ABSTRACT FROM AUTHOR]

The dispersion pattern of a received signal is critical for understanding physical properties of the propagation medium. The objective of this work is to estimate accurately sediment sound speed using modal arrival times obtained from dispersion curves extracted via time-frequency analysis of acoustic signals. A particle filter is used that estimates probability density functions of modal frequencies arriving at specific times. Employing this information, probability density functions of arrival times for modal frequencies are constructed. Samples of arrival time differences are then obtained and are propagated backwards through an inverse acoustic model. As a result, probability density functions of sediment sound speed are estimated. Maximum a posteriori estimates indicate that inversion is successful. It is also demonstrated that multiple frequency processing offers an advantage over inversion at a single frequency, producing results with reduced variance. [ABSTRACT FROM AUTHOR]

Numerical modeling of wave propagation in acoustic tubes is a subject of longstanding interest, particularly for enclosures of varying cross section, and especially when viscothermal losses due to boundary layer effects are taken into consideration. Though steady-state, or frequency domain methods, are a common avenue of approach, recursive time domain methods are an alternative, allowing for the generation of wideband responses, and offer a point of departure for more general modeling of nonlinear wave propagation. The design of time-domain methods is complicated by numerical stability considerations, and to this end, a passive representation is a useful design principle leading to simple stable and explicit numerical schemes, particularly in the case of viscothermal loss modeling. Such schemes and the accompanying energy and stability analysis are presented here. Numerical examples are presented for a variety of duct profiles, illustrating strict energy dissipation, and for comparison of computed input impedances against frequency-domain results. [ABSTRACT FROM AUTHOR]

Some studies of concert hall acoustics consider the acoustics in a hall as a single entity. Here, it is shown that the acoustics vary between different seats, and the choice of music also influences the perceived acoustics. The presented study compared the acoustics of six unoccupied concert halls with extensive listening tests, applying two different music excerpts on three different seats. Twenty eight assessors rated the halls according to the subjective preference of the assesors and individual attributes with a paired comparison method. Results show that assessors can be classified into two preference groups, which prioritize different perceptual factors. In addition, the individual attributes elicited by assessors were clustered into three latent classes. [ABSTRACT FROM AUTHOR]

This paper presents a theoretical study on "seeing" through an elastic structure to uncover the root cause of sound and vibration by using nearfield acoustical holography (NAH) and normal modes expansion. This approach is of generality because vibro-acoustic responses on the surface of a vibrating structure can always be reconstructed, exactly or approximately. With these vibroacoustic responses, excitation forces acting on the structure can always be determined, analytically or numerically, given any set of boundary conditions. As an example, the explicit formulations for reconstructing time-harmonic excitation forces, including point, line and surface forces, and their arbitrary combinations acting on a rectangular thin plate in vacuum mounted on an infinite baffle are presented. The reason for choosing this example is that the analytic solutions to vibro-acoustic responses are available, and in-depth analyses of results are possible. Results demonstrate that this approach allows one to identify excitation forces based on measured acoustic pressures and reveal their characteristics such as locations, types and amplitudes, as if one could "see" excitation forces acting behind the plate based on acoustic pressure measured on the opposite side. This approach is extendable to general elastic structures, except that in such circumstance numerical results must be sought. [ABSTRACT FROM AUTHOR]

This article presents a new approach to interactive spatial sonification of multidimensional data as a tool for spatial sound synthesis, for composing temporal--spatial musical materials, and as an auditory display for scientists to analyze multidimensional data sets in time and space. The approach applies parameter-mapping sonification and is currently implemented in an application called Cheddar, which was programmed in Max/MSP. Cheddar sonifies data in real time, where the user can modify a wide variety of temporal, spatial, and sonic parameters during the listening process, and thus more easily uncover patterns and processes in the data than when applying non-real-time, noninteractive techniques. The design draws on existing literature concerning perception and acoustics, and it applies the author's practical experience in acousmatic composition, spectromorphology, and sound semantics, while addressing accuracy, flexibility, and ease of use. Although previous sonification applications have addressed some degree of real-time control and spatialization, this approach integrates space and sound in an interactive framework. Spatial information is sonified in high-order 3-D ambisonics, where the user can interactively move the virtual listening position to reveal details easily missed from fixed or noninteractive spatial views. Sounds used as input to the sonification take advantage of the rich spectra and extramusical attributes of acoustic sources, which, although previously theorized, are investigated here in a practical context thoroughly tested alongside acoustic and psychoacoustic considerations. Furthermore, when using Cheddar, no specialized knowledge of programming, acoustics, or psychoacoustics is required. These approaches position Cheddar at the junction between science and art. With one application serving both disciplines, the patterns and processes of science are more fluently appropriated into music or sound art, and vice versa for scientific research, science public outreach, and education. [ABSTRACT FROM AUTHOR]

The term roughness is used to describe a specific sound sensation which may occur when listening to stimuli with more than one spectral component within the same critical band. It is believed that the spectral components interact inside the cochlea, which leads to fluctuations in the neural signal and, in turn, to a sensation of roughness. This study presents a roughness model composed of two successive stages: peripheral and central. The peripheral stage models the function of the peripheral ear. The central stage predicts roughness from the temporal envelope of the signal processed by the peripheral stage. The roughness model was shown to account for the perceived roughness of various types of acoustic stimuli, including the stimuli with temporal envelopes that are not sinusoidal. It thus accounted for effects of the phase and the shape of the temporal envelope on roughness. The model performance was poor for unmodulated bandpass noise stimuli. [ABSTRACT FROM AUTHOR]

Annoyance ratings for artificially created noises, resembling the main characteristics of temporal wind turbine noise, were studied by means of a listening experiment involving 21 participants with normal hearing. Three types of stimuli were examined: broadband noise (−4 dB/octave), noise generated by moving cars, and narrowband noise. All stimuli had the sound level fluctuations typical for wind turbine noise. The magnitude of the sound level fluctuations was measured in a quantitative way, by using the characteristics of amplitude modulated sound: modulation rate and modulation depth. Our aim was to examine how the modulation rate and the modulation depth influence the noise annoyance assessment of broadband and narrowband amplitude modulated noises. Three different modulation rates, 1, 2 and 4 Hz, and sound level fluctuations (a measure of the modulation depth), 3, 6, 9 dB, were applied to each type of stimuli (with exception of noise generated by the moving cars) and investigated. The participants in the listening experiment were presented with sound stimuli in laboratory conditions and asked to rate their annoyance on a numerical scale. The results have shown a significant difference between the investigated conditions. The effect was particularly strong between the annoyance judgments of different types of noise (narrow and broadband), and modulated versus unmodulated noises. Temporal fluctuations occurring in wind turbine noise are very pertinent to the perception of annoyance and could be responsible for its being a relatively annoying noise source. The obtained results were discussed and compared to the typical modulation rates and level changes that occur in recordings of real wind turbine noise. [ABSTRACT FROM AUTHOR]

This study presents the indoor soundscape framework in detail by describing the variables and factors that form an indoor soundscape study. The main objective is to introduce a new indoor soundscaping framework and systematically explain the variables that contribute to the overall evaluation of an indoor soundscape. Hence, the dependencies of physical and psychoacoustical factors of the sound environment and the spatial factors of the built entity are statistically tested. The new indoor soundscaping framework leads to an overarching evaluation perspective of enclosed sound environments, combining objective room acoustics research and noise control engineering with architectural analysis. Therefore, it is hypothesised that case spaces with certain plan organisations, volumetric relations, and spatial referencing lead to differentiated sound pressure level (SPL) and loudness (N) values. SPL and N parametric variances of the sound environments are discussed through the statistical findings with respect to the architectural characteristics of each library case space. The results show that the relation between crowd level variances and sound environment parametric values is statistically significant. It is also found that increasing the atrium height and atrium void volume, the atrium's presence as a common architectural element, and its interpenetrating reference and domain containment results in unwanted variances and acoustic formations, leading to high SPL and N values. [ABSTRACT FROM AUTHOR]

In this article the authors investigated and presented the experiments on the sentence boundaries annotation from Polish speech using acoustic cues as a source of information. The main result of the investigation is an algorithm for detection of the syntactic boundaries appearing in the places of punctuation marks. In the first stage, the algorithm detects pauses and divides a speech signal into segments. In the second stage, it verifies the configuration of acoustic features and puts hypotheses of the positions of punctuation marks. Classification is performed with parameters describing phone duration and energy, speaking rate, fundamental frequency contours and frequency bands. The best results were achieved for Naive Bayes classifier. The efficiency of the algorithm is 52% precision and 98% recall. Another significant outcome of the research is statistical models of acoustic cues correlated with punctuation in spoken Polish. [ABSTRACT FROM AUTHOR]

The aim of this work was to measure subjective speech intelligibility in an enclosure with a long reverberation time and comparison of these results with objective parameters. Impulse Responses (IRs) were first determined with a dummy head in different measurement points of the enclosure. The following objective parameters were calculated with Dirac 4.1 software: Reverberation Time (RT), Early Decay Time (EDT), weighted Clarity (C50) and Speech Transmission Index (STI). For the chosen measurement points, a convolution of the IRs with the Polish Sentence Test (PST) and logatome tests was made. PST was presented at a background of a babble noise and speech reception threshold - SRT (i.e. SNR yielding 50% speech intelligibility) for those points were evaluated. A relationship of the sentence and logatome recognition vs. STI was determined. It was found that the final SRT data are well correlated with speech transmission index (STI), and can be expressed by a psychometric function. The difference between SRT determined in condition without reverberation and in reverberation conditions appeared to be a good measure of the effect of reverberation on speech intelligibility in a room. In addition, speech intelligibility, with and without use of the sound amplification system installed in the enclosure, was compared. [ABSTRACT FROM AUTHOR]

This paper describes the development phases of a numerical-experimental integrated approach aimed at obtaining sufficiently accurate predictions of the noise field emitted by an external gear pump by means of some vibration measurements on its external casing. Harmonic response methods and vibroacoustic analyses were considered as the main tools of this methodology. FFT acceleration spectra were experimentally acquired only in some positions of a 8.5 cc/rev external gear pump casing for some working conditions and considered as external excitation boundary conditions for a FE quite simplified vibroacoustic model. The emitted noise field was computed considering the pump as a 'black box', without taking into account the complex dynamics of the gear tooth meshing process and the consequent fluid pressure and load distribution. Sound power tests, based on sound intensity measurements, as well as sound pressure measurements in some positions around the pump casing were performed for validation purposes. The comparisons between numerical and experimental results confirmed the potentiality of this approach in offering a good compromise between noise prediction accuracy and reduction of experimental and modelling requirements. [ABSTRACT FROM AUTHOR]

The paper presents the possible applications of using acoustic diagnostics in inspecting the technical condition of an internal combustion engine with autoignition on the example of the Fiat drive unit with common rail system. As a result of measuring the sound pressure level for specific faults and comparing the noise generated by the motor running smoothly, the detailed maps of changes in the acoustic spectrum are possible to generate. These results may be helpful in the future diagnostics of internal combustion engines. In the paper, the results of scientific work in the area of research, design and operation of internal combustion engines, conducted at the Department of Automotive Engineering, in cooperation with the Laboratory of Hydraulic Drives & Vibroacoustics of Machines at the Wroclaw University of Technology are included. [ABSTRACT FROM AUTHOR]

The paper presents two theoretical models for traffic noise level distribution on curved horizontal roads. In the case of vehicles moving on a given route, one can consider, in terms of sound field, that the granular traffic is equivalent for short periods with a quasi-continuous noise flow. When computing and modelling the noise level generated by traffic on roads with complex trajectory, it is common to treat the route as a sum of small length road segments, each being assimilated with a linear noise source. This paper started from the assumption that the route can be decomposed into a sequence of linear and arc-shaped road segments, each of which is treated as a linear respectively curved noise source. An arc-shaped road segment is modelled by a tubular vibrating surface, of circular or rectangular section. In the case of rectangular section, the vibrating blade emits complex sounds on its both vertical sides and the generated sound field can be described more clearly, qualitatively and quantitatively, through intensity distribution. The theoretical models presented in the paper have direct application to the traffic noise prediction and noise maps drawing [ABSTRACT FROM AUTHOR]

This study presents the first acoustic description of the vowel space of a Papuan language-Nambo, spoken in southern Papua New Guinea-based on duration and first and second formant measurements from 19 adult male and female speakers across three age groups (young, middle-aged, senior). Phonemically, Nambo has six full vowels /i, e, æ, A, o, u/ and a reduced vowel tentatively labeled /@/. Unlike the full vowels, the quality of /@/ showed great variation: seniors' and young females' realizations tended to be more open and retracted than those by young males, while middle-aged speakers' productions fell between these two variants. [ABSTRACT FROM AUTHOR]