Volume 137, Issue 4, April 2015
- jasa express letters
- animal bioacoustics
- architectural acoustics
- biomedical acoustics
- engineering acoustics
- musical acoustics
- physical acoustics
- psychological and physiological acoustics
- signal processing in acoustics
- speech communication
- structural acoustics and vibration
- underwater acoustics
- book reviews
- reviews of acoustical patents
Index of content:
- JASA EXPRESS LETTERS
A comparison of finite element and analytic models of acoustic scattering from rough poroelastic interfaces137(2015); http://dx.doi.org/10.1121/1.4914947View Description Hide Description
The finite element method is used to model acoustic scattering from rough poroelastic surfaces. Both monostatic and bistatic scattering strengths are calculated and compared with three analytic models: Perturbation theory, the Kirchhoff approximation, and the small-slope approximation. It is found that the small-slope approximation is in very close agreement with the finite element results for all cases studied and that perturbation theory and the Kirchhoff approximation can be considered valid in those instances where their predictions match those given by the small-slope approximation.
137(2015); http://dx.doi.org/10.1121/1.4914946View Description Hide Description
Traditional loudspeaker equalization algorithms cannot decide the order of an equalizer before the whole equalization procedure has been completed. Designers have to try many times before they determine a proper order of the equalization filter. A method which solves this drawback is presented for loudspeaker equalization using balanced model truncation. The order of the equalizer can be easily decided using this algorithm and the error between the model and the loudspeaker can also be readily controlled. Examples are presented and the performance of the proposed method is discussed with comparative experiments.
137(2015); http://dx.doi.org/10.1121/1.4915064View Description Hide Description
It has been argued that, to ensure accurate spectral feature estimates for sibilants, the spectral estimation method should include a low-variance spectral estimator; however, no empirical evaluation of estimation methods in terms of feature estimates has been given. The spectra of /s/ and /ʃ/ were estimated with different methods that varied the pre-emphasis filter and estimator. These methods were evaluated in terms of effects on two features (centroid and degree of sibilance) and on the detection of four linguistic contrasts within these features. Estimation method affected the spectral features but none of the tested linguistic contrasts.
Localization of low-frequency coherent sound sources with compressive beamforming-based passive synthetic aperture137(2015); http://dx.doi.org/10.1121/1.4915003View Description Hide Description
The localization of low-frequency coherent sources requires a proper aperture to ensure a high spatial resolution. Attaining a large aperture is difficult in practice when the conditions involved are limited. This letter investigated a compressive beamforming-based passive synthetic aperture approach with a reference sensor in a fixed position. Localization findings on acoustic sources in a semi-anechoic chamber were compared with conventional beamforming, compressive beamforming, passive synthetic aperture, and compressive beamforming-based passive synthetic aperture. Results suggest that the proposed method can produce a higher spatial resolution and higher detection ability than the others.
Experimental validation of a coprime linear microphone array for high-resolution direction-of-arrival measurementsa)137(2015); http://dx.doi.org/10.1121/1.4915000View Description Hide Description
Coprime linear microphone arrays allow for narrower beams with fewer sensors. A coprime microphone array consists of two staggered uniform linear subarrays with and microphones, where and are coprime with each other. By applying spatial filtering to both subarrays and combining their outputs, microphones yield directional bands. In this work, the coprime sampling theory is implemented in the form of a linear microphone array of 16 elements with coprime numbers of 9 and 8. This coprime microphone array is experimentally tested to validate the coprime array theory. Both predicted and measured results are discussed. Experimental results confirm that narrow beampatterns as predicted by the coprime sampling theory can be obtained by the coprime microphone array.
Extension of a spectral time-stepping domain decomposition method for dispersive and dissipative wave propagation137(2015); http://dx.doi.org/10.1121/1.4915061View Description Hide Description
For time-domain modeling based on the acoustic wave equation, spectral methods have recently demonstrated promise. This letter presents an extension of a spectral domain decomposition approach, previously used to solve the lossless linear wave equation, which accommodates frequency-dependent atmospheric attenuation and assignment of arbitrary dispersion relations. Frequency-dependence is straightforward to assign when time-stepping is done in the spectral domain, so combined losses from molecular relaxation, thermal conductivity, and viscosity can be approximated with little extra computation or storage. A mode update free from numerical dispersion is derived, and the model is confirmed with a numerical experiment.
137(2015); http://dx.doi.org/10.1121/1.4915063View Description Hide Description
Given a geometrical model of a space, the problem of optimally placing absorption in a space to match a desired impulse response is in general nonlinear. This has led some to use costly optimization procedures. This letter reformulates absorption assignment as a constrained linear least-squares problem. Regularized solutions result in direct distribution of absorption in the room and can accommodate multiple frequency bands, multiple sources and receivers, and constraints on geometrical placement of absorption. The method is demonstrated using a beam tracing model, resulting in the optimal absorption placement on the walls and ceiling of a classroom.
137(2015); http://dx.doi.org/10.1121/1.4914999View Description Hide Description
Localization of a 2-ms-click target was previously shown to be influenced by interleaved localization trials in which the target was preceded by an identical distractor [Kopčo, Best, and Shinn-Cunningham (2007). J. Acoust. Soc. Am. 121, 420–432]. Here, two experiments were conducted to explore this contextual effect. Results show that context-related bias is not eliminated (1) when the response method is changed so that vision is available or that no hand-pointing is required; or (2) when the distractor-target order is reversed. Additionally, a keyboard-based localization response method is introduced and shown to be more accurate than traditional pointer-based methods.
137(2015); http://dx.doi.org/10.1121/1.4914998View Description Hide Description
Acoustic signals generated in water by terawatt (TW) laser pulses undergoing filamentation are studied. The acoustic signal has a very broad spectrum, spanning from 0.1 to 10 MHz and is confined in the plane perpendicular to the laser direction. Such a source appears to be promising for the development of remote laser based acoustic applications.
137(2015); http://dx.doi.org/10.1121/1.4915062View Description Hide Description
To expose systematic trends in violin sound accompanying “tuning” only the plates or only the bridge, the first structural acoustics-based model auralizations of violin sound were created by passing a bowed-string driving force measured at the bridge of a solid body violin through the dynamic filter (DF) model radiativity profile “filter” RDF(f) (frequency-dependent pressure per unit driving force, free-free suspension, anechoic chamber). DF model auralizations for the more realistic case of a violin held/played in a reverberant auditorium reveal that holding the violin greatly diminishes its low frequency response, an effect only weakly compensated for by auditorium reverberation.
137(2015); http://dx.doi.org/10.1121/1.4915005View Description Hide Description
Passive time reversal (pTR) is a low complexity receiver scheme that uses multichannel probing for time signal refocusing, thus reducing time spreading and improving inter-symbol interference. Recognizing that signals traveling through different paths are subject to arrival-angle-related Doppler displacements, this letter proposes a further improvement to pTR that applies correcting frequency shifts optimized for beams formed along each specific path arrival angle. The proposed channel equalizer is tested with real data, and the results show that the proposed approach outperforms both pTR and the modified pTR channel combiners providing an MSE gain of 4.9 and 4.2 dB, respectively.
137(2015); http://dx.doi.org/10.1121/1.4916792View Description Hide Description
French listeners' reliance on voicing, manner, and place was tested in a mispronunciation detection task. Mispronounced words were more likely to be recognized when the mispronunciation concerned voicing rather than manner or place. This indicates that listeners rely less on the former than on the latter for the purposes of word recognition. Further, the role of visual cues to phonetic features was explored by the task being conducted in both an audio-only and an audiovisual version, but no effect of modality was found. Discussion focuses on crosslinguistic comparisons and lexical factors that might influence the weight of individual features.
137(2015); http://dx.doi.org/10.1121/1.4916794View Description Hide Description
A methodology is developed to measure ex situ ultrasonic velocity of submerged aquatic vegetation tissue, in particular, macroalgae, in a nondestructive and efficient manner. An entire thallus is submerged in artificial seawater-filled tank through which many ultrasonic pulse-echo measurements are recorded while thallus parts are randomly displaced. Average sound speed of tissue is estimated from normal fit to extracted travel times given measured total volume fraction of tissue and travel time in water alone. For species Ecklonia radiata the resulting values for sound speed 1573.4 ± 4.8 m s−1 and adiabatic compressibility 3.134 ×10−10 ± 1.34 ×10−11 Pa−1 at 18 °C agree with more laborious and destructive methods.
137(2015); http://dx.doi.org/10.1121/1.4916796View Description Hide Description
The received Doppler signal of a stationary sensor, as emitted by a transiting acoustic source, is used to estimate source motion parameters, including speed, closest distance, rest frequency, and closest point of approach (CPA) time. First, the instantaneous frequency, amplitude, and CPA time are accurately estimated by the polynomial chirplet transform of the Doppler signal. Thereafter, the three other source motion parameters are obtained with a simplified amplitude-weighted nonlinear least squares method. The proposed scheme is successfully applied to the analysis of the characteristics of a moving truck.
137(2015); http://dx.doi.org/10.1121/1.4916797View Description Hide Description
During a phone conversation, loud vocal emission from the far-end to the near-end space can disturb nearby people. In this paper, the possibility of actively controlling such unwanted sound emission using a control source placed on the mobile device is investigated. Two different approaches are tested: Global control, minimizing the potential energy measured along a volumetric space surface, and local control, minimizing the squared sound pressure at a discrete point on the phone. From the test results, both approaches can reduce the unwanted sound emission by more than 6 dB in the frequency range up to 2 kHz.
137(2015); http://dx.doi.org/10.1121/1.4916791View Description Hide Description
Materials with well-defined microlattice structures are superlight, stable, and thus bear great potential in sound absorption. An integrated transfer matrix method (TMM) is proposed to evaluate the sound absorbing efficiency of these lattice materials, in which a massive number of micropores are densely placed. A comparison between integrated TMM and conventional TMM reveals that the proposed approach offers better predictions on sound absorption of microlattice. This approach is then employed to optimize the microlattice material to determine the best pore and porosity that lead to maximum absorbing efficiency capability and minimum required thickness to attain a target sound absorption.
How broadband speech may avoid neural firing rate saturation at high intensities and maintain intelligibility137(2015); http://dx.doi.org/10.1121/1.4916793View Description Hide Description
Three experiments examined the intelligibility enhancement produced when noise bands flank high intensity rectangular band speech. When white noise flankers were added to the speech individually at a low spectrum level (−30 dB relative to the speech) only the higher frequency flanker produced a significant intelligibility increase (i.e., recovery from intelligibility rollover). However, the lower-frequency flanking noise did produce an equivalent intelligibility increase when its spectrum level was increased by 10 dB. This asymmetrical intensity requirement, and other results, support previous suggestions that intelligibility loss at high intensities is reduced by lateral inhibition in the cochlear nuclei.
137(2015); http://dx.doi.org/10.1121/1.4916795View Description Hide Description
Low-frequency and infrasonic pure-tone monaural hearing threshold data down to 2.5 Hz are presented. These measurements were made by means of a newly developed insert-earphone source. The source is able to generate pure-tone sound pressure levels up to 130 dB between 2 and 250 Hz with very low harmonic distortions. Behavioral hearing thresholds were determined in the frequency range from 2.5 to 125 Hz for 18 otologically normal test persons. The median hearing thresholds are comparable to values given in the literature. They are intended for stimulus calibration in subsequent brain imaging investigations.
- ANIMAL BIOACOUSTICS
Effects of exposure to intermittent and continuous 6–7 kHz sonar sweeps on harbor porpoise (Phocoena phocoena) hearing137(2015); http://dx.doi.org/10.1121/1.4916590View Description Hide Description
Safety criteria for mid-frequency naval sonar sounds are needed to protect harbor porpoise hearing. A porpoise was exposed to sequences of one-second 6–7 kHz sonar down-sweeps, with 10–200 sweeps in a sequence, at an average received sound pressure level (SPL av.re.) of 166 dB re 1 μPa, with duty cycles of 10% (intermittent sounds) and 100% (continuous). Behavioral hearing thresholds at 9.2 kHz were determined before and after exposure to the fatiguing noise, to quantify temporary hearing threshold shifts (TTS 1–4 min) and recovery. Significant TTS 1–4 min occurred after 10–25 sweeps when the duty cycle was 10% (cumulative sound exposure level, SELcum: ∼178 dB re 1 μPa2s). For the same SELcum, the TTS 1–4 min was greater for exposures with 100% duty cycle. The difference in TTS between the two duty cycle exposures increased as the number of sweeps in the exposure sequences increased. Therefore, to predict TTS and permanent threshold shift, not only SELcum needs to be known, but also the duty cycle or equivalent sound pressure level (Leq). It appears that the injury criterion for non-pulses proposed by Southall, Bowles, Ellison, Finneran, Gentry, Greene, Kastak, Ketten, Miller, Nachtigall, Richardson, Thomas, and Tyack [(2007). Aquat. Mamm. 33, 411–521] for cetaceans echolocating at high frequency (SEL 215 dB re 1 μPa2s) is too high for the harbor porpoise.
137(2015); http://dx.doi.org/10.1121/1.4916591View Description Hide Description
To investigate the auditory effects of multiple underwater impulses, hearing thresholds were measured in three bottlenose dolphins before and after exposure to 10 impulses produced by a seismic air gun. Thresholds were measured at multiple frequencies using both psychophysical and electrophysiological (auditory evoked potential) methods. Exposures began at relatively low levels and gradually increased over a period of several months. The highest exposures featured peak sound pressure levels from 196 to 210 dB re 1 μPa, peak-peak sound pressure levels of 200–212 dB re 1 μPa, and cumulative (unweighted) sound exposure levels from 193 to 195 dB re 1 μPa2s. At the cessation of the study, no significant increases were observed in psychophysical thresholds; however, a small (9 dB) shift in mean auditory evoked potential thresholds, accompanied by a suppression of the evoked potential amplitude function, was seen in one subject at 8 kHz. At the highest exposure condition, two of the dolphins also exhibited behavioral reactions indicating that they were capable of anticipating and potentially mitigating the effects of impulsive sounds presented at fixed time intervals.