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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]

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]

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]

The targeted energy transfer (TET) phenomenon is demonstrated and analyzed between an acoustic medium inside a parallelepiped cavity and a thin viscoelastic membrane that is mounted on one wall of the cavity and that is working as a nonlinear absorber or a nonlinear energy sink (NES). Based on a two degrees-of-freedom (DOF) system, which is comprised by an acoustic mode of 3D acoustic cavity and a nonlinear membrane NES, the desired working zone for the membrane NES is defined and the two thresholds of the zone are obtained. In this paper, we investigate a parameter analysis of a nonlinear membrane absorber based on TET technique to design the NES. The physical parameters of the membrane and the place of the membrane on the wall of the cavity are studied. It can finally allow us to determine where is the better place for the membrane NES and which parameter affects mainly the desired working zone for the membrane NES that has to be used for reducing low frequency resonances inside a given acoustic cavity. Numerical simulations are preformed to validate the TET phenomenon and the results of the parameter analysis. [ABSTRACT FROM AUTHOR]

Detailed mapping of the sound field produced by a modern turbofan engine, with its multitude of overlapping noise sources, often requires a large number of microphones to properly resolve the directivity patterns of the constituent tonal and broadband components. This is especially true at high frequencies where the acoustic wavelength is short, or when shielding, scattering, and reflection of the sound field may be present due to installation effects. This paper presents a novel method for measuring the harmonic and broadband content of complex noncompact noise sources using continuously moving (referred to here as continuous-scan (CS)) microphones in conjunction with a state-ofthe- art phase-referencing technique. Because the microphones are moving through the sound field produced by the noise sources, they effectively provide infinite spatial resolution of the sound directivity over the scan path. In this method, harmonic (i.e., shaftcoherent) content at the integer multiples of the instantaneous shaft rotational frequency is first extracted from the time signal using a tachometer signal and the Vold-Kalman (VK) filter. The residual broadband signal is then filtered in the time domain in fractional octave bands. The broadband spectra of the signals from the moving microphones are then computed at arbitrary positions along their scan paths using weighted averages (based on Chebyshev polynomial zero-crossings) and the assumption of a complex envelope that varies slowly over a spatial scale whose lower bound is set by the acoustic wavenumber. A benefit of this method is that the decomposition of the total measured sound field into a stochastic superposition of components preserves a meaningful phase definition for each "partial field" associated with a given shaft order (SO). This preservation of phase data enables the forward or backward pro jection of each of these partial fields using acoustical holography (AH). The benefits of the CS method are demonstrated using acoustic data acquired for a 22-in. scale-model fan stage run at the NASA Glenn Research Center's 9-foot by 15-foot wind tunnel. Two key outcomes of the work include (1) significant improvement in the spatial resolution of the measured sound field and (2) reduction in the overall data acquisition time. Additionally, the methods described here lead to new opportunities for noise source diagnostics and visualization. [ABSTRACT FROM AUTHOR]

Measurements of mean volume backscattering strength (Sv, dB re 1 m-1) at ocean-basin scale were made using 38-kHz hull-mounted echosounders on ships of opportunity as part of Australia's Integrated Marine Observing System. The data were collected on vessels of various designs, none of which were purposely built for collecting high-quality acoustic data. A full range of weather extremes affected the quality of the data and could cause large biases in Sv. To remove first-order biases and improve processing efficiency, a sequence of new and existing data processing filters were applied in a semi-automated procedure. These filters were designed to mitigate the effects of three types of noise: impulsive (less than one ping), transient (multiple pings), and background (hours or longer). A filter was also applied to identify signals that were attenuated by air bubbles beneath the transducer. These filters were applied to data from transits across the Southwest Pacific, Indian, and Southern Oceans to produce quality-controlled Sv datasets that are now available from a publicly accessible repository. These filters may be relevant to other open-ocean acoustic observing endeavours, and one or more could be used to mitigate bias in data from a range of acoustic applications. [ABSTRACT FROM AUTHOR]

For sound studies, the publication of a 593-page handbook, not to mention the establishment of at least one society – the European Sound Studies Association – might seem to signify the emergence of a new academic discipline. Certainly, the books under consideration here, alongside many others, testify to an intensification of concern with the aural dimensions of culture. Some of this work comes from HPS and STS, some from musicology and cultural studies. But all of it should concern members of our disciplines, as it represents a long-overdue foregrounding of the aural in how we think about the intersections of science, technology and culture. [ABSTRACT FROM PUBLISHER]

Interaural correlation change detection was measured in untrained normal-hearing listeners. Narrowband (10-Hz) noises were varied by center frequency (CF; 500 or 4000 Hz) and diotic level roving (absent or present). For the 500-Hz CF, 96% of listeners could achieve threshold (79.4% correct at the easiest testing level) if roving was absent, but only 36% of listeners could if level roving was present. No one could achieve threshold at the 4000-Hz CF, unlike trained listeners in the literature. The results raise questions about how individual differences affect learning and generalization of monaural and binaural cues related to interaural correlation detection. [ABSTRACT FROM AUTHOR]

This paper considers the noise performance of microphone preamplifiers at low gain setting. A detailed noise analysis of the classic current-feedback instrumentation amplifier topology often used for transformerless microphone preamplifiers is presented. The noise contributors that are significant at low gain settings are identified. Several techniques to reduce their magnitude are discussed. One novel technique is introduced that is easily implemented at very low cost and requires a modest increase in power consumption only. Furthermore, it offers a benefit with respect to the frequency response and distortion performance of the circuit. The theoretical results are verified with experimental measurement data. [ABSTRACT FROM AUTHOR]

The directionality of ambient noise in an Arctic tidewater glacier bay was measured using two horizontally spaced, broadband hydrophones. Segments of noise were divided into two frequency bands and analyzed for arrival angle. These data show that different classes of source radiate noise in distinct spectral bands and are spatially diverse. A previously unidentified source, the interaction of surface gravity waves with underside of ice ledges at the periphery of icebergs, is described. The generation of noise by ice-wave interaction suggests that surface waves should be measured if ambient noise is to be used to monitor ice dynamics in glacial fjords. [ABSTRACT FROM AUTHOR]

The robustness of the human auditory system to noise is partly due to the peak preserving capability of the periphery and the cortical filtering of spectro-temporal modulations. In this letter, a robust speech feature extraction scheme is developed that emulates this processing by deriving a spectrographic representation that emphasizes the high energy regions. This is followed by a modulation filtering step to preserve only the important spectro-temporal modulations. The features derived from this representation provide significant improvements for speech recognition in noise and language identification in radio channel speech. Further, the experimental analysis shows congruence with human psychophysical studies. [ABSTRACT FROM AUTHOR]

Dispersion curves in ocean environments are accurately estimated from received signals through the extraction of instantaneous modal frequencies and corresponding arrival times for long-range propagation. The ultimate goal is to estimate sediment sound speed using the extracted dispersion pattern. The approach extends work previously conducted in dispersion tracking with sequential filtering, improving on the latter technique. The sequential state-space method that is developed for the extraction of time-frequency information from specific time instances relies on a representation of those as a sum of elemental pulses, resulting from analysis of the received field. The method is tested on synthetic noisy data with different noise levels. After dispersion probability density functions are estimated via a particle filter, they are subsequently employed for sound speed inversion. Correct mode identification is a challenge impacting inversion; this is demonstrated through two examples and a way to remedy the problem is discussed. [ABSTRACT FROM AUTHOR]

Sound source localization accuracy using a sound source identification task was measured in the front, right quarter of the azimuth plane as rms (root-mean-square) error (degrees) for stimulus conditions in which the bandwidth (1/20 to 2 octaves wide) and center frequency (250, 2000, 4000 Hz) of 200-ms noise bursts were varied. Tones of different frequencies (250, 2000, 4000 Hz) were also used. As stimulus bandwidth increases, there is an increase in sound source localization identification accuracy (i.e., rms error decreases). Wideband stimuli (>1 octave wide) produce best sound source localization accuracy (~6°-7° rms error), and localization accuracy for these wideband noise stimuli does not depend on center frequency. For narrow bandwidths (<1 octave) and tonal stimuli, accuracy does depend on center frequency such that highest accuracy is obtained for low-frequency stimuli (centered on 250 Hz), worse accuracy for mid-frequency stimuli (centered on 2000 Hz), and intermediate accuracy for high-frequency stimuli (centered on 4000 Hz). [ABSTRACT FROM AUTHOR]

When phase relationships between partials of a complex masker produce highly modulated temporal envelopes on the basilar membrane, listeners may detect speech information from temporal dips in the within-channel masker envelopes. This source of masking release (MR) is however located in regions of unresolved masker partials and it is unclear how much of the speech information in these regions is really needed for intelligibility. Also, other sources of MR such as glimpsing in between resolved masker partials may provide sufficient information from regions that disregard phase relationships. This study simplified the problem of speech recognition to a masked detection task. Target bands of speech-shaped noise were restricted to frequency regions containing either only resolved or only unresolved masker partials, as a function of masker phase relationships (sine or random), masker fundamental frequency (F0) (50, 100, or 200 Hz), and masker spectral profile (flat-spectrum or speech-shaped). Although masker phase effects could be observed in unresolved regions at F0s of 50 and 100 Hz, it was only at 50-Hz F0 that detection thresholds were ever lower in unresolved than in resolved regions, suggesting little role of envelope modulations for harmonic complexes with F0s in the human voice range and at moderate level. [ABSTRACT FROM AUTHOR]

The hypothesis of this study was that broader patterns of physiological channel interactions in the local region of the cochlea are associated with poorer spectral resolution in the same region. Electrically evoked compound action potentials (ECAPs) were measured for three to six probe electrodes per subject to examine the channel interactions in different regions across the electrode array. To evaluate spectral resolution at a confined location within the cochlea, spectral-ripple discrimination (SRD) was measured using narrowband ripple stimuli with the bandwidth spanning five electrodes: Two electrodes apical and basal to the ECAP probe electrode. The relationship between the physiological channel interactions, spectral resolution in the local cochlear region, and vowel identification was evaluated. Results showed that (1) there was within- and across-subject variability in the widths of ECAP channel interaction functions and in narrowband SRD performance, (2) significant correlations were found between the widths of the ECAP functions and narrowband SRD thresholds, and between mean bandwidths of ECAP functions averaged across multiple probe electrodes and broadband SRD performance across subjects, and (3) the global spectral resolution reflecting the entire electrode array, not the local region, predicts vowel identification. [ABSTRACT FROM AUTHOR]

Otoacoustic emission (OAE) tests of the medial-olivocochlear reflex (MOCR) in humans were assessed for viability as clinical assays. Two reflection-source OAEs [TEOAEs: transient-evoked otoacoustic emissions evoked by a 47 dB sound pressure level (SPL) chirp; and discrete-tone SFOAEs: stimulus-frequency otoacoustic emissions evoked by 40 dB SPL tones, and assessed with a 60 dB SPL suppressor] were compared in 27 normal-hearing adults. The MOCR elicitor was a 60 dB SPL contralateral broadband noise. An estimate of MOCR strength, MOCR%, was defined as the vector difference between OAEs measured with and without the elicitor, normalized by OAE magnitude (without elicitor). An MOCR was reliably detected in most ears. Within subjects, MOCR strength was correlated across frequency bands and across OAE type. The ratio of across-subject variability to within-subject variability ranged from 2 to 15, with wideband TEOAEs and averaged SFOAEs giving the highest ratios. MOCR strength in individual ears was reliably classified into low, normal, and high groups. SFOAEs using 1.5 to 2 kHz tones and TEOAEs in the 0.5 to 2.5 kHz band gave the best statistical results. TEOAEs had more clinical advantages. Both assays could be made faster for clinical applications, such as screening for individual susceptibility to acoustic trauma in a hearing-conservation program. [ABSTRACT FROM AUTHOR]

A limitation currently facing active structural acoustic control (ASAC) researchers is that an ideal minimization quantity for use in the control algorithms has not been developed. A novel parameter termed the "weighted sum of spatial gradients" (WSSG) was recently developed for use in ASAC and shown to effectively attenuate acoustic radiation from a vibrating flat simply supported plate in computer simulations. This paper extends this research from computer simulations and provides experimental test results. The results presented show that WSSG is a viable control quantity and provides better results than the volume velocity approach. The paper also investigates several of the challenges presented by the use of WSSG. These include determining a method to measure WSSG experimentally, an analysis of the influence of noise on WSSG control results and complications presented when degenerate modes exist. Results are shown and discussed for several experimental configurations. [ABSTRACT FROM AUTHOR]

Structural-change detection and characterization, or structural-health monitoring, is generally based on modal analysis, for detection, localization, and quantification of changes in structure. Classical methods combine both variations in frequencies and mode shapes, which require accurate and spatially distributed measurements. In this study, the detection and localization of a local perturbation are assessed by analysis of frequency changes (in the fundamental mode and overtones) that are combined with a perturbation-based linear inverse method and a deconvolution process. This perturbation method is applied first to a bending beam with the change considered as a local perturbation of the Young's modulus, using a one-dimensional finite-element model for modal analysis. Localization is successful, even for extended and multiple changes. In a second step, the method is numerically tested under ambient-noise vibration from the beam support with local changes that are shifted step by step along the beam. The frequency values are revealed using the random decrement technique that is applied to the time-evolving vibrations recorded by one sensor at the free extremity of the beam. Finally, the inversion method is experimentally demonstrated at the laboratory scale with data recorded at the free end of a Plexiglas beam attached to a metallic support. [ABSTRACT FROM AUTHOR]

In September 2012, the free-falling, deep-diving instrument platform Deep Sound III descended to the bottom of the Tonga Trench, where it resided at a depth of 8515 m for almost 3 h, recording ambient noise data on four hydrophones arranged in a vertical L-shaped configuration. The time series from each of the hydrophones yielded the power spectrum of the noise over the frequency band 3 Hz to 30 kHz. The spatial coherence functions, along with the corresponding cross-correlation functions, were recovered from all available hydrophone pairs in the vertical and the horizontal. The vertical coherence and cross-correlation data closely follow the predictions of a simple theory of sea-surface noise in a semi-infinite ocean, suggesting that the seabed in the Tonga Trench is a very poor acoustic reflector, which is consistent with the fact that the sediment at the bottom of the trench consists of very-fine-grained material having an acoustic impedance similar to that of sea-water. The horizontal coherence and cross-correlation data are a little more complicated, showing evidence of (a) bathymetric shadowing of the noise by the walls of the trench and (b) highly directional acoustic arrivals from the research vessel supporting the experiment. [ABSTRACT FROM AUTHOR]

This work develops the so-called compensated impedance boundary conditions that enable stable time domain simulations of sound propagation in a lined duct with uniform mean flow, which has important practical interest for noise emission by aero-engines. The proposed method is developed analytically from an unusual perspective of control that shows impedance boundary conditions act as closed-loop feedbacks to an overall duct acoustic system. It turns out that those numerical instabilities of time domain simulations are caused by deficient phase margins of the corresponding control-oriented model. A particular instability of very low frequencies in the presence of steady uniform background mean flow, in addition to the well known high frequency numerical instabilities at the grid size, can be identified using this analysis approach. Stable time domain impedance boundary conditions can be formulated by including appropriate phaselead compensators to achieve desired phase margins. The compensated impedance boundary conditions can be simply designed with no empirical parameter, straightforwardly integrated with ordinary linear acoustic models, and efficiently calculated with no need of resolving sheared boundary layers. The proposed boundary conditions are validated by comparing against asymptotic solutions of spinning modal sound propagation in a duct with a hard-soft interface and reasonable agreement is achieved. [ABSTRACT FROM AUTHOR]

Recent studies of auditory streaming have suggested that repeated synchronous onsets and offsets over time, referred to as "temporal coherence," provide a strong grouping cue between acoustic components, even when they are spectrally remote. This study uses a measure of auditory stream formation, based on comodulation masking release (CMR), to assess the conditions under which a loss of temporal coherence across frequency can lead to auditory stream segregation. The measure relies on the assumption that the CMR, produced by flanking bands remote from the masker and target frequency, only occurs if the masking and flanking bands form part of the same perceptual stream. The masking and flanking bands consisted of sequences of narrowband noise bursts, and the temporal coherence between the masking and flanking bursts was manipulated in two ways: (a) By introducing a fixed temporal offset between the flanking and masking bands that varied from zero to 60 ms and (b) by presenting the flanking and masking bursts at different temporal rates, so that the asynchronies varied from burst to burst. The results showed reduced CMR in all conditions where the flanking and masking bands were temporally incoherent, in line with expectations of the temporal coherence hypothesis. [ABSTRACT FROM AUTHOR]

In the present paper, the authors present a numerical analysis of the sound reduction provided by simple acoustic protective measures to attenuate the noise emitted by equipment placed near a facade. The noise source is assumed to be surrounded by an enclosed space, defined as a parallelepiped chamber with a rectangular opening. To perform the numerical analysis, a 3D Boundary Element formulation is implemented. This formulation makes use of the domain decomposition, together with Green's functions, specifically defined to reduce the size of the involved system matrices and to allow consideration for surface absorption. Indeed, these Green's functions, defined using the image-source technique, allow modelling of the building's facade and the ground as infinite surfaces with a given absorption coefficient. The numerical model is verified against an analytical solution known for the case of a point load acting within a parallelepiped space; additionally, it is validated by comparing its results with those obtained experimentally for a simple case. The implemented model is then used to perform a number of numerical simulations, illustrating the effect of different configurations of the protective surfaces in the sound reduction. [ABSTRACT FROM AUTHOR]

Personal audio refers to the creation of a listening zone within which a person, or a group of people, hears a given sound program, without being annoyed by other sound programs being reproduced in the same space. Generally, these different sound zones are created by arrays of loudspeakers. Although these devices have the capacity to achieve different sound zones in an anechoic environment, they are ultimately used in normal rooms, which are reverberant environments. At high frequencies, reflections from the room surfaces create a diffuse pressure component which is uniform throughout the room volume and thus decreases the directional characteristics of the device. This paper shows how the reverberant performance of an array can be modeled, knowing the anechoic performance of the radiator and the acoustic characteristics of the room. A formulation is presented whose results are compared to practical measurements in reverberant environments. Due to reflections from the room surfaces, pressure variations are introduced in the transfer responses of the array. This aspect is assessed by means of simulations where random noise is added to create uncertainties, and by performing measurements in a real environment. These results show how the robustness of an array is increased when it is designed for use in a reverberant environment. [ABSTRACT FROM AUTHOR]

Eight normal-hearing listeners practiced a tone-detection task in which a 1-kHz target was masked by a spectrally unpredictable multitone complex. Consistent learning was observed, with mean masking decreasing by 6.4 dB over five sessions (4500 trials). Reverse-correlation was used to estimate how listeners weighted each spectral region. Weight-vectors approximated the ideal more closely after practice, indicating that listeners were learning to attend selectively to the task relevant information. Once changes in weights were accounted for, no changes in internal noise (psychometric slope) were observed. It is concluded that this task elicits robust learning, which can be understood primarily as improved selective attention. [ABSTRACT FROM AUTHOR]

The relative independence of time-unit processing during speech reception was examined. It was found that temporally interpolated noise, even at very high levels, had little effect on sentence recognition using masking-release conditions similar to those of Kwon et al. [(2012). J. Acoust. Soc. Am. 131, 31113119]. The current data confirm the earlier conclusions of Kwon et al. involving masking release based on the relative timing of speech and noise. These data also indicate substantial levels of independence in the time domain, which has implications for current theories of C speech perception in noise. 2014 Acoustical Society of America. [ABSTRACT FROM AUTHOR]

This paper investigates to what extent users of bilateral and bimodal fittings should expect to benefit from all three different binaural advantages found to be present in normal-hearing listeners. Head-shadow and binaural squelch are advantages occurring under spatially separated speech and noise, while summation emerges when speech and noise coincide in space. For 14 bilateral or bimodal listeners, speech reception thresholds in the presence of four-talker babble were measured in sound-field under various speech and noise configurations. Statistical analysis revealed significant advantages of head-shadow and summation for both bilateral and bimodal listeners. Squelch was significant only for bimodal listeners. [ABSTRACT FROM AUTHOR]

Perception of warning sounds, such as vehicle backup alarms, is reduced when hearing protection devices (HPDs) are worn. A cross-correlation approach is employed to detect a pre-selected warning sound and enable it to bypass the attenuation of the HPD while still attenuating the environmental noise. Computer simulation shows that the algorithm can detect the specified alarm at signal-to-environmental-noise ratios as low as -30 dB. Human subject testing of the algorithm, implemented on a modified commercial HPD, confirms the minimum detection threshold obtained in simulation, and demonstrates a 7 dB improvement in detection threshold compared with the unmodified HPD. [ABSTRACT FROM AUTHOR]

A tonal thickness noise and loading noise model of rotating blades has been developed as an extension of the exact frequency-domain solutions for rotating monopole and dipole point sources. The present model has two advantages over the previous methods and models for noise prediction. The first is the unified expression for sources in subsonic and supersonic rotation even at rest. The second is that the present model has no limit on the location of the observer and no interpolation error. Two test cases are carried out to validate the present model and emphasize its advantage at the noise prediction for sources in supersonic rotation. Moreover, as a specified application of the present model for the rotating blades whose tip radius is acoustically compact, acoustic energy distribution at different frequencies and in different directions is analyzed. Result shows that the acoustic energy of acoustically compact rotating blades is mainly concentrated at the source frequency while propagating along the axial direction, leaving the rest propagating along the radial direction at the other frequencies. [ABSTRACT FROM AUTHOR]

This study investigated the extent to which noise impacts normal-hearing young adults' speech processing of sentences that vary in intelligibility. Intelligibility and recognition memory in noise were examined for conversational and clear speech sentences recorded in quiet (quiet speech, QS) and in response to the environmental noise (noise-adapted speech, NAS). Results showed that (1) increased intelligibility through conversational-to-clear speech modifications led to improved recognition memory and (2) NAS presented a more naturalistic speech adaptation to noise compared to QS, leading to more accurate word recognition and enhanced sentence recognition memory. These results demonstrate that acoustic-phonetic modifications implemented in listener-oriented speech enhance speech-in-noise processing beyond word recognition. Effortful speech processing in challenging listening environments can thus be improved by speaking style adaptations on the part of the talker. In addition to enhanced intelligibility, a substantial improvement in recognition memory can be achieved through speaker adaptations to the environment and to the listener when in adverse conditions. [ABSTRACT FROM AUTHOR]

Automatic and accurate detection of the closure-burst transition events of stops and affricates serves many applications in speech processing. A temporal measure named the plosion index is proposed to detect such events, which are characterized by an abrupt increase in energy. Using the maxima of the pitch-synchronous normalized cross correlation as an additional temporal feature, a rule-based algorithm is designed that aims at selecting only those events associated with the closure-burst transitions of stops and affricates. The performance of the algorithm, characterized by receiver operating characteristic curves and temporal accuracy, is evaluated using the labeled closure-burst transitions of stops and affricates of the entire TIMIT test and training databases. The robustness of the algorithm is studied with respect to global white and babble noise as well as local noise using the TIMIT test set and on telephone quality speech using the NTIMIT test set. For these experiments, the proposed algorithm, which does not require explicit statistical training and is based on two one-dimensional temporal measures, gives a performance comparable to or better than the state-of-the-art methods. In addition, to test the scalability, the algorithm is applied on the Buckeye conversational speech corpus and databases of two Indian languages. [ABSTRACT FROM AUTHOR]

Using the data presented in the accompanying paper [Hilkhuysen et al., J. Acoust. Soc. Am. 131, 531-539 (2012)], the ability of six metrics to predict intelligibility of speech in noise before and after noise suppression was studied. The metrics considered were the Speech Intelligibility Index (SII), the fractional Articulation Index (fAI), the coherence intelligibility index based on the midlevels in speech (CSIImid), an extension of the Normalized Coherence Metric (NCM+), a part of the speech-based envelope power model (pre-sEPSM), and the Short Term Objective Intelligibility measure (STOI). Three of the measures, SII, CSIImid, and NCM+, overpredicted intelligibility after noise reduction, whereas fAI underpredicted these intelligibilities. The pre-sEPSM metric worked well for speech in babble but failed with car noise. STOI gave the best predictions, but overall the size of intelligibility prediction errors were greater than the change in intelligibility caused by noise suppression. Suggestions for improvements of the metrics are discussed. [ABSTRACT FROM AUTHOR]

Background reverberation can obscure useful features of the target echo response in broadband low-frequency sonar images, adversely affecting detection and classification performance. This paper describes a resolution and phase-preserving means of separating the target response from the background reverberation noise using a coherence-based wavelet shrinkage method proposed recently for de-noising magnetic resonance images. The algorithm weights the image wavelet coefficients in proportion to their coherence between different looks under the assumption that the target response is more coherent than the background. The algorithm is demonstrated successfully on experimental synthetic aperture sonar data from a broadband low-frequency sonar developed for buried object detection [ABSTRACT FROM AUTHOR]

A cue indicating when in time to listen can improve adults' tone detection thresholds, particularly for conditions that produce substantial informational masking. The purpose of this study was to determine if 5- to 13-yr-old children likewise benefit from a light cue indicating when in time to listen for a masked pure-tone signal. Each listener was tested in one of two continuous maskers: Broadband noise (low informational masking) or a random-frequency, two-tone masker (high informational masking). Using a single-interval method of constant stimuli, detection thresholds were measured for two temporal conditions: (1) Temporally-defined, with the listening interval defined by a light cue, and (2) temporally-uncertain, with no light cue. Thresholds estimated from psychometric functions fitted to the data indicated that children and adults benefited to the same degree from the visual cue. Across listeners, the average benefit of a defined listening interval was 1.8 dB in the broadband noise and 8.6 dB in the random-frequency, two-tone masker. Thus, the benefit of knowing when in time to listen was more robust for conditions believed to be dominated by informational masking. An unexpected finding of this study was that children's thresholds were comparable to adults' in the random-frequency, two-tone masker. [ABSTRACT FROM AUTHOR]

This study examined the ability of human listeners to detect the presence and judge the strength of a statistical dependency among the elements comprising sequences of sounds. The statistical dependency was imposed by specifying transition matrices that determined the likelihood of occurrence of the sound elements. Markov chains were constructed from these transition matrices having states that were pure tones/noise bursts that varied along the stimulus dimensions of frequency and/or interaural time difference. Listeners reliably detected the presence of a statistical dependency in sequences of sounds varying along these stimulus dimensions. Furthermore, listeners were able to discriminate the relative strength of the dependency in pairs of successive sound sequences. Random variation along an irrelevant stimulus dimension had small but significant adverse effects on performance. A much greater decrement in performance was found when the sound sequences were concurrent. Likelihood ratios were computed based on the transition matrices to specify Ideal Observer performance for the experimental conditions. Preliminary modeling efforts were made based on degradations of Ideal Observer performance intended to represent human observer limitations. This experimental approach appears to be useful for examining auditory 'stream' formation and maintenance over time based on the predictability of the constituent sound elements. [ABSTRACT FROM AUTHOR]

A novel methodology is presented for the simultaneous measurement of noise and personal annoyance at the exact times that the affected person is annoyed. The system is described and applied to a test case, a farmhouse close to a wind farm where the resident claims to be annoyed by noise. The system was successfully able to characterise the level and spectral content of the noise in the house when the resident was annoyed, and there was some correlation with personally recorded annoyance level. As the system cannot identify noise sources, no conclusions can be made about noise source; however, the methodology is shown to be a useful aid for diagnosing the type and severity of an indoor noise problem. To help interpret the results from this type of testing, a discussion concerning the subjective nature of noise annoyance is presented before some suggestions are made for further improvements in the measurement system. [ABSTRACT FROM AUTHOR]

Large-eddy simulations are performed using high-fidelity numerical schemes developed for turbulence simulations and computational aeroacoustics to study near-ideally expanded and tone-producing supersonic jets vertically impinging on a large flat plate. The simulations compute the complex interaction of the supersonic jet with the flat impingement plane and directly provide the noise generated by this interaction. Two jet-operating temperatures, which correspond to isothermal and heated conditions, are considered in the simulations. The simulation results from both cases are examined to investigate the intense tonal noise generation arising from the jet impingement. The comparisons between the simulation results and the corresponding experimental measurements generally show a reasonable agreement and provide credibility to the quality of the simulation predictions. The simulations reveal important details about the problem of interest, which are not observable from the experiments. A dynamic mode decomposition is performed to identify the dynamically important modes of the supersonic impinging-jet flowfield for the simulated conditions. The most dominant mode of the flowfield is found to possess a temporal frequency that corresponds to the fundamental impingement tone frequency. Coherent nearly axisymmetric structures in the flowfield are also identified, which are generated at the fundamental impingement tone frequency. [ABSTRACT FROM AUTHOR]

A year-long experiment (March 2010 to April 2011) measuring ambient sound at a shallow water site (50 m) on the central OR coast near the Port of Newport provides important baseline information for comparisons with future measurements associated with resource development along the inner continental shelf of the Pacific Northwest. Ambient levels in frequencies affected by surf-generated noise (f < 100 Hz) characterize the site as a high-energy end member within the spectrum of shallow water coastal areas influenced by breaking waves. Dominant sound sources include locally generated ship noise (66% of total hours contain local ship noise), breaking surf, wind induced wave breaking and baleen whale vocalizations. Additionally, an increase in spectral levels for frequencies ranging from 35 to 100 Hz is attributed to noise radiated from distant commercial ship commerce. One-second root mean square (rms) sound pressure level (SPLrms) estimates calculated across the 10-840 Hz frequency band for the entire year long deployment show minimum, mean, and maximum values of 84 dB, 101 dB, and 152 dB re 1 μPa. [ABSTRACT FROM AUTHOR]

Several measures of sound source localization performance of 45 listeners with normal hearing were obtained when loudspeakers were in the front hemifield. Localization performance was not statistically affected by filtering the 200-ms, 2-octave or wider noise bursts (125 to 500, 1500 to 6000, and 125 to 6000 Hz wide noise bursts). This implies that sound source localization performance for noise stimuli is not differentially affected by which interaural cue (interaural time or level difference) a listener with normal hearing uses for sound source localization, at least for relatively broadband signals. This sound source localization task suggests that listeners with normal hearing perform with high reliability/repeatability, little response bias, and with performance measures that are normally distributed with a mean root-mean-square error of 6.2° and a standard deviation of 1.79°. [ABSTRACT FROM AUTHOR]

Abstract: This study proposes a prediction method for assessing the sound of a passing vehicle that is transmitted through a glass plate, which employs the vibro-acoustic finite-difference time-domain (FDTD) method. The noise that is transmitted through the glass in a building façade, which is affected by the sound-insulation characteristics of its glass plates, can have a psychological influence such as a sleep disorder on the residents. In this study, a prediction method for the spectral characteristics of the transmitting sound through glass plates is proposed. The sound-insulation characteristics of glass plates are obtained using a vibro-acoustic FDTD method, and are then synthesized with the sound of passing vehicles obtained by in situ measurement. Firstly, the sound transmitted through several kinds of glass plates is simulated using the proposed method. Then, in order to confirm the validity of this method, the simulated results are compared to measured sounds transmitted by passing vehicles into a room near the street. [Copyright &y& Elsevier]

The ability to automatically recognize a wide range of sound events in real-world conditions is an important part of applications such as acoustic surveillance and machine hearing. Our approach takes inspiration from both audio and image processing fields, and is based on transforming the sound into a two-dimensional representation, then extracting an image feature for classification. This provided the motivation for our previous work on the spectrogram image feature (SIF). In this paper, we propose a novel method to improve the sound event classification performance in severe mismatched noise conditions. This is based on the subband power distribution (SPD) image—a novel two-dimensional representation that characterizes the spectral power distribution over time in each frequency subband. Here, the high-powered reliable elements of the spectrogram are transformed to a localized region of the SPD, hence can be easily separated from the noise. We then extract an image feature from the SPD, using the same approach as for the SIF, and develop a novel missing feature classification approach based on a nearest neighbor classifier (kNN). We carry out comprehensive experiments on a database of 50 environmental sound classes over a range of challenging noise conditions. The results demonstrate that the SPD-IF is both discriminative over the broad range of sound classes, and robust in severe non-stationary noise. [ABSTRACT FROM PUBLISHER]

The effects of forward and backward noise fringes on binaural signal detectability were investigated. Masked thresholds for a 12-ms, 250-Hz, sinusoidal signal masked by Gaussian noise, centered at 250 Hz, with bandwidths from 3 to 201 Hz, were obtained in N0S0 and N0Sπ configurations. The signal was (a) temporally centered in a 12-ms noise burst (no fringe), (b) presented at the start of a 600-ms noise burst (backward fringe), or (c) temporally centered in a 600-ms noise burst (forward-plus-backward fringe). For noise bandwidths between 3 and 75 Hz, detection in N0S0 improved with the addition of a backward fringe, improving further with an additional forward fringe; there was little improvement in N0Sπ. The binaural masking-level difference (BMLD) increased from 0 to 8 dB with a forward-plus-backward fringe as noise bandwidths increased to 100 Hz, increasing slightly to 10 dB at 201 Hz. This two-stage increase was less pronounced with a backward fringe. With no fringe, the BMLD was about 10-14 dB at all bandwidths. Performance appears to result from the interaction of across-time and across-frequency listening strategies and the possible effects of gain reduction and suppression, which combine in complex ways. Current binaural models are, as yet, unable to account fully for these effects. [ABSTRACT FROM AUTHOR]

The extent of noise and its impacts continues to grow globally indicating a different approach from regulating individual noise sources is needed. We pose the argument that soundscapes, or the acoustic environment, should be managed as a common-pool resource (CPR). Using CPR theory, we argue that soundscapes possess key features of CPRs: (1) multiple soundscape users, (2) difficulty of exclusion, and (3) subtractability and degradation. Using Ostrom's Social-Ecological Systems (SES) framework, we describe the main elements of soundscapes to consider for their sustainable management. In order to assess noise issues and challenges in managing national park soundscapes, we conducted interviews with U.S. National Park Service managers at parks identified as having air tour overflight impacts. While most managers indicated that aircraft overflights posed the most serious impacts to park resources and visitor experiences, the park units also experienced several other types of noise impacts including traffic on park roads, park maintenance operations, and different types of motorized recreational vehicles. Addressing single sources of noise is necessary, as is the case with air tour overflights, but we argue that a more comprehensive approach is needed to protect park soundscapes. From this study several SES framework variables emerged that need to be addressed for sustainable management, such as the lack of clear soundscape boundaries, availability of acoustic monitoring and data, and the number and types of soundscape users. Based on CPR theory and using the SES framework, the challenges and a potential new approach for sustainable management are discussed. [ABSTRACT FROM AUTHOR]

Throughout the United States, opportunities to experience noise-free intervals are disappearing. Rapidly increasing energy development, infrastructure expansion, and urbanization continue to fragment the acoustical landscape. Within this context, the National Park Service endeavors to protect acoustical resources because they are essential to park ecology and central to the visitor experience. The Park Service monitors acoustical resources in order to determine current conditions, and forecast the effects of potential management decisions. By community noise standards, background sound levels in parks are relatively low. By wilderness criteria, levels of noise audibility are remarkably high. A large percentage of the noise sources measured in national parks (such as highways or commercial jet traffic) originates outside park boundaries and beyond the management jurisdiction of NPS. Many parks have adopted noise mitigation plans, but the regional and national scales of most noise sources call for conservation and management efforts on similar scales. [ABSTRACT FROM AUTHOR]

The extensive literature documenting the ecological effects of roads has repeatedly implicated noise as one of the causal factors. Recent studies of wildlife responses to noise have decisively identified changes in animal behaviors and spatial distributions that are caused by noise. Collectively, this research suggests that spatial extent and intensity of potential noise impacts to wildlife can be studied by mapping noise sources and modeling the propagation of noise across landscapes. Here we present models of energy extraction, aircraft overflight and roadway noise as examples of spatially extensive sources and to present tools available for landscape scale investigations. We focus these efforts in US National Parks (Mesa Verde, Grand Teton and Glacier) to highlight that ecological noise pollution is not a threat restricted to developed areas and that many protected natural areas experience significant noise loads. As a heuristic tool for understanding past and future noise pollution we forecast community noise utilizing a spatially-explicit land-use change model that depicts the intensity of human development at sub-county resolution. For road noise, we transform effect distances from two studies into sound levels to begin a discussion of noise thresholds for wildlife. The spatial scale of noise exposure is far larger than any protected area, and no site in the continental US is free form noise. The design of observational and experimental studies of noise effects should be informed by knowledge of regional noise exposure patterns. [ABSTRACT FROM AUTHOR]

Slow envelope fluctuations in the range of 2-20 Hz provide important segmental cues for processing communication sounds. For a successful segmentation, a neural processor must capture envelope features associated with the rise and fall of signal energy, a process that is often challenged by the interference of background noise. This study investigated the neural representations of slowly varying envelopes in quiet and in background noise in the primary auditory cortex (Al) of awake marmoset monkeys. We characterized envelope features based on the local average and rate of change of sound level in envelope waveforms and identified envelope features to which neurons were selective by reverse correlation. Our results showed that envelope feature selectivity of Al neurons was correlated with the degree of nonmonotonicity in their static rate-level functions. Nonmonotonic neurons exhibited greater feature selectivity than monotonic neurons in quiet and in background noise. The diverse envelope feature selectivity decreased spike-timing correlation among Al neurons in response to the same envelope waveforms. As a result, the variability, but not the average, of the ensemble responses of Al neurons represented more faithfully the dynamic transitions in low-frequency sound envelopes both in quiet and in background noise. [ABSTRACT FROM AUTHOR]

Over the last few years, several field attenuation measurement systems (FAMS) have been introduced to the industrial marketplace to measure the individual end-user attenuation for some hearing protection devices (HPDs). Although individual measurement is necessary to determine whether a given user is properly protected in his or her real-life noise environment (assuming that the exposure level is known), one unknown remains with a FAMS measurement: how reliable are the predictions made from the instantaneous measurement (over a few minutes), for determining the attenuation that will be achieved later in the field (over months or years) by the end-user who may fit them slightly differently every time? This paper will address that question for one FAMS, the field microphone-in-real-ear (F-MIRE) measurement technique, and we will study, in the laboratory, how consistently subjects can fit and refit HPDs without assistance. A new metric, the intra-subject fit variability, will be introduced and will be quantified for custom-molded earplugs as fitted by inexperienced test subjects. This paper will present the experimental process used and statistical calculations performed to quantify the intra-subject fit variability. The number of successive refits required for a given prediction accuracy will also be presented, as well as the uncertainty component associated with the intra-subject fit variability when using an F-MIRE field attenuation measurement system. [ABSTRACT FROM AUTHOR]

Abstract: There are four main air-related noise generation mechanisms at the tyre/road interface, which were all categorised more than 20years ago. The first one is the so-called ‘air pumping’ mechanism. Two other air-related phenomena that occur when there are air movements near the contact patch of the tyre are ‘air resonant radiation’ and ‘pipe resonances’ which appear at the footprint of the tyre. In addition to these, there is a forth effect, which is mentioned in the literature, that is occurring due to turbulence effects of the air surrounding the spinning tyre. There has been less focus on the air-related mechanisms than on other types of tyre noise generation mechanisms. This paper attempts to add some detail to current understanding of the air-related noise generation at the tyre road interface and gives some further information on how to identify the differences due to these mechanisms. Specifically in the present paper, a solid rubber tyre running on a vehicle chassis dynamometer is used to study the first two mechanisms. This is done with emphasis on the time history of the recorded signal and not on the frequency spectrum, as is more commonly used. A comparison with existing theoretical models of these mechanisms reveals some of the strength and weaknesses of the current understanding of these phenomena. [Copyright &y& Elsevier]

The article presents studies related to effects of noise. The study "Coping Strategies for Low Frequency Noise," by G. Leventhall et al., conducted intervention sessions aimed at improving the participants' coping strategies and quality of life amid low frequency noise. In "Shock and Impact Levels on North American Locomotives," by N. Cooperrider and J. Gordon, the health risks posed by isolated shocks are discussed. The study "Computation of Trunk Muscle Forces, Spinal Loads, and Stability in Whole-Body Vibration," by B. Bazrgari et al., is presented.