Volume 134, Issue 5, November 2013
- jasa express letters
- letters to the editor
- general linear acoustics 
- nonlinear acoustics 
- aeroacoustics, atmospheric sound 
- underwater sound 
- ultrasonics, quantum acoustics, and physical effects of sound 
- transduction 
- structural acoustics and vibration 
- noise: its effects and control 
- architectural acoustics 
- acoustical measurements and instrumentation 
- acoustic signal processing 
- physiological acoustics 
- psychological acoustics 
- speech production 
- speech perception 
- music and musical instruments 
- bioacoustics 
- acoustical news
- acoustical standards news
- reviews of acoustical patents
- program abstracts of the 166th meeting of the acoustical society of america
- award encomiums
- program abstracts of the 166th meeting of the acoustical society of america
Index of content:
- JASA EXPRESS LETTERS
Correlations of linear and nonlinear ultrasound parameters with density and microarchitectural parameters in trabecular bone134(2013); http://dx.doi.org/10.1121/1.4822420View Description Hide Description
In the present study, correlations of linear and nonlinear ultrasound parameters (speed of sound, normalized broadband ultrasound attenuation, and nonlinear parameter B/A) with bone mineral density and microarchitectural parameters were investigated in 28 bovine femoral trabecular bone samples in vitro. All three ultrasound parameters exhibited relatively high correlation coefficients with the indexes of bone quantity (bone mineral density and bone volume fraction) and lower correlation coefficients with the remaining microarchitectural parameters. These results suggest that B/A, in addition to speed of sound and attenuation, may have potential as an index for the assessment of bone status and osteoporosis.
134(2013); http://dx.doi.org/10.1121/1.4822320View Description Hide Description
The role of visual cues in native listeners' perception of speech produced by nonnative speakers has not been extensively studied. Native perception of English sentences produced by native English and Korean speakers in audio-only and audiovisual conditions was examined. Korean speakers were rated as more accented in audiovisual than in the audio-only condition. Visual cues enhanced word intelligibility for native English speech but less so for Korean-accented speech. Reduced intelligibility of Korean-accented audiovisual speech was associated with implicit visual biases, suggesting that listener-related factors partially influence the efficiency of audiovisual integration for nonnative speech perception.
134(2013); http://dx.doi.org/10.1121/1.4822421View Description Hide Description
An inversion scheme based on time-warping is presented for estimating seabed sound attenuation from modal dispersion of close-range single-hydrophone data. The dispersion information is extracted directly from the warped signal spectrum. Seabed sound speed and density are inverted from the modal group velocity curves, and the attenuation is inverted from the normalized modal amplitudes. The method is applied to experimental data collected in the Yellow Sea of China during the winter of 2002. The inverted sound speed and density are consistent with the sand-silt-clay sediment at the site, and the attenuation is nonlinear over the frequency band from 125–500 Hz.
Calibrating passive acoustic monitoring: Correcting humpback whale call detections for site-specific and time-dependent environmental characteristics134(2013); http://dx.doi.org/10.1121/1.4822319View Description Hide Description
This paper demonstrates the importance of accounting for environmental effects on passive underwater acoustic monitoring results. The situation considered is the reduction in shipping off the California coast between 2008–2010 due to the recession and environmental legislation. The resulting variations in ocean noise change the probability of detecting marine mammal vocalizations. An acoustic model was used to calculate the time-varying probability of detecting humpback whale vocalizations under best-guess environmental conditions and varying noise. The uncorrected call counts suggest a diel pattern and an increase in calling over a two-year period; the corrected call counts show minimal evidence of these features.
134(2013); http://dx.doi.org/10.1121/1.4824036View Description Hide Description
Breebaart et al. [J. Acoust. Soc. Am. 110, 1089–1104 (2001)] reported that the masker bandwidth dependence of detection thresholds for an out-of-phase signal and an in-phase noise masker (N 0 Sπ ) can be explained by principles of integration of information across critical bands. In this paper, different methods for such across-frequency integration process are evaluated as a function of the bandwidth and notch width of the masker. The results indicate that an “optimal detector” model assuming independent internal noise in each critical band provides a better fit to experimental data than a best filter or a simple across-frequency integrator model. Furthermore, the exponent used to model peripheral compression influences the accuracy of predictions in notched conditions.
134(2013); http://dx.doi.org/10.1121/1.4820461View Description Hide Description
Effects of frequency-shifted feedback are typically examined using Eventide Harmonizer Series processors to shift the fundamental frequency (F 0) of auditory feedback during vocalizations, eliciting compensatory shifts in speaker F 0. Recently, unexpected intensity changes were observed in speakers' ear canals, corresponding with F 0 shifts. An investigation revealed that feedback time delays introduced by the processor resulted in phase shifts between feedback and unprocessed voice signals radiating into the ear canal via bone conduction, producing combination waves with gains as high as 6 dB. Shifts of this magnitude potentially alter the interpretation of previously published results and should be controlled in future studies.
134(2013); http://dx.doi.org/10.1121/1.4824629View Description Hide Description
A standard proposal for rating airborne sound insulation in buildings [ISO 16717-1 (2012)] defines the reference noise spectra. Since their shapes influence the calculated values of single-number descriptors, reference spectra should approximate well typical noise spectra in buildings. There is, however, very little data in the existing literature on a typical noise spectrum in dwellings. A spectral analysis of common noise sources in dwellings is presented in this paper, as a result of an extensive monitoring of various noisy household activities. Apart from music with strong bass content, the proposed “living” reference spectrum overestimates noise levels at low frequencies.
134(2013); http://dx.doi.org/10.1121/1.4824633View Description Hide Description
This letter develops a Bayesian focalization approach for three-dimensional localization of an unknown number of sources in shallow water with uncertain environmental properties. The algorithm minimizes the Bayesian information criterion using adaptive hybrid optimization for environmental parameters, Metropolis sampling for source bearing, and Gibbs sampling for source ranges and depths. Maximum-likelihood expressions are used for unknown complex source strengths and noise variance, which allows these parameters to be sampled implicitly. An efficient scheme for adding/deleting sources is used during the optimization. A synthetic example considers localizing a quiet source in the presence of multiple interferers using a horizontal line array.
134(2013); http://dx.doi.org/10.1121/1.4824631View Description Hide Description
A lightweight push-pull acoustic transducer using dielectric elastomer films was proposed for use in advanced audio systems in homes. The push-pull structure consists of two dielectric elastomer films developed to serve as an electroactive polymer. The transducer utilizes the change in the surface area of the dielectric elastomer film, induced by an electric-field-induced change in the thickness, for sound generation. The resonance frequency of the transducer was derived from modeling the push-pull configuration to estimate the lower limit of the frequency range. Measurement results presented an advantage of push-pull driving in the suppression of harmonic distortion.
Speech quality estimation of voice over internet protocol codec using a packet loss impairment model134(2013); http://dx.doi.org/10.1121/1.4824628View Description Hide Description
This letter proposes a degradation and cognition model to estimate speech quality impairment because of packet loss concealment (PLC) algorithm implemented in the speech CODEC SILK. By considering the fact that the quality degradation caused by packet loss is highly related to the PLC algorithm, the impact of quality degradation on various types of previous and lost packet classes is analyzed. Then, the PLC effects to the proposed class types are measured by the class conditional expectation of the degradation scores. Finally, the cognition module is derived to estimate the total quality degradation in a mean opinion score (MOS) scale. When assessed for correlation with subject test results, the correlation coefficient of the encoder-based class model is 0.93, and that of the decoder-based model is 0.87.
134(2013); http://dx.doi.org/10.1121/1.4824630View Description Hide Description
Compressive sensing, a newly emerging method from information technology, is applied to array beamforming and associated acoustic applications. A compressive sensing beamforming method (CSB-II) is developed based on sampling covariance matrix, assuming spatially sparse and incoherent signals, and then examined using both simulations and aeroacoustic measurements. The simulation results clearly show that the proposed CSB-II method is robust to sensing noise. In addition, aeroacoustic tests of a landing gear model demonstrate the good performance in terms of resolution and sidelobe rejection.
134(2013); http://dx.doi.org/10.1121/1.4824632View Description Hide Description
In this paper, a computational goal for a monaural speech separation system is proposed. Since this goal is derived by maximizing the signal-to-noise ratio (SNR), it is called the optimal ratio mask (ORM). Under the approximate W-Disjoint Orthogonality assumption which almost always holds due to the sparse nature of speech, theoretical analysis shows that the ORM can improve the SNR about dB over the ideal ratio mask. With three kinds of real-world interference, the speech separation results of SNR gain and objective quality evaluation demonstrate the correctness of the theoretical analysis, and imply that the ORM achieves a better separation performance.
134(2013); http://dx.doi.org/10.1121/1.4826148View Description Hide Description
The Speech Intelligibility Index (SII) estimates speech intelligibility based on the audibility of speech cues across frequency. The frequency importance function gives the relative contribution to the SII of the speech audibility at different frequencies. The frequency importance function is usually estimated from the intelligibility data using a complicated multi-step procedure. This paper presents a new procedure for computing the frequency importance function directly from the intelligibility data based on nonlinear joint optimization of the frequency importance function and the SII curve-fitting parameters. An example of using the new approach is presented for previously published W-22 word list intelligibility data.
134(2013); http://dx.doi.org/10.1121/1.4826152View Description Hide Description
Speech reception thresholds were measured for a voice against two different maskers: Either two concurrent voices with the same fundamental frequency (F0) or a harmonic complex with the same long-term excitation pattern and broadband temporal envelope as the masking sentences (speech-modulated buzz). All sources had steady F0s. A difference in F0 of 2 or 8 semitones provided a 5-dB benefit for buzz maskers, whereas it provided a 3- and 8-dB benefit, respectively, for masking sentences. Whether intelligibility of a voice increases abruptly with small ΔF0s or gradually toward larger ΔF0s seems to depend on the nature of the masker.
Deduction of the acoustic impedance of the ground via a simulated three-dimensional microphone array134(2013); http://dx.doi.org/10.1121/1.4826149View Description Hide Description
While commonly used ground impedance deduction methods often utilize pairs of vertically separated microphones, deployed arrays rarely have this configuration, which increases the difficulty in automatically deducing local ground impedance from these arrays. The ability to deduce ground impedance using random sounds incident on a three-dimensional array would increase, for example, the accuracy of estimated elevation angles. The methods described by the American National Standards Institute Method for Determining the Acoustic Impedance of Ground Surfaces are extended to simulate deducing ground impedance by a three-dimensional array. Ground parameters indicative of grassland are successfully determined using a simulated three-dimensional array.
134(2013); http://dx.doi.org/10.1121/1.4826150View Description Hide Description
Vowel space area (VSA) is an attractive metric for the study of speech production deficits and reductions in intelligibility, in addition to the traditional study of vowel distinctiveness. Traditional VSA estimates are not currently sufficiently sensitive to map to production deficits. The present report describes an automated algorithm using healthy, connected speech rather than single syllables and estimates the entire vowel working space rather than corner vowels. Analyses reveal a strong correlation between the traditional VSA and automated estimates. When the two methods diverge, the automated method seems to provide a more accurate area since it accounts for all vowels.
- LETTERS TO THE EDITOR
134(2013); http://dx.doi.org/10.1121/1.4824395View Description Hide Description
The study investigated how listeners used level and direct-to-reverberant ratio (D/R) cues to discriminate distances to virtual sound sources. Sentence pairs were presented at virtual distances in simulated rooms that were either reverberant or anechoic. Performance on the basis of level was generally better than performance based on D/R. Increasing room reverberation time improved performance based on the D/R cue such that the two cues provided equally effective information at further virtual source distances in highly reverberant environments. Orientation of the listener within the virtual room did not affect performance.
Response to “Comment on ‘Resonant acoustic scattering by swimbladder-bearing fish’ ” [J. Acoust. Soc. Am. 64, 571–580 (1978)]134(2013); http://dx.doi.org/10.1121/1.4823805View Description Hide Description
In the 1970s a model of resonant scattering from a swimbladder-bearing fish was developed. The fish was modeled as an air bubble, representing a swimbladder, encased in a viscous spherical shell, representing the fish flesh. This model has been used successfully to correlate acoustic scattering data with fish information in a number of ocean locations. Recently, questions have arisen about viscous damping of the flesh and the thickness of the shell [K. Baik, J. Acoust. Soc. Am. 133, 5–8 (2013)]. This Letter responds to those questions and provides practical insight into the model's use.
- GENERAL LINEAR ACOUSTICS 
134(2013); http://dx.doi.org/10.1121/1.4822478View Description Hide Description
Laboratory measurements of enhanced sound transmission from water to air at low frequencies are presented. The pressure at a monitoring hydrophone is found to decrease for shallow source depths in agreement with the classical theory of a monopole source in proximity to a pressure release interface. On the other hand, for source depths below 1/10 of an acoustic wavelength in water, the radiation pattern in the air measured by two microphones becomes progressively omnidirectional in contrast to the classical geometrical acoustics picture in which sound is contained within a cone of 13.4° half angle. The measured directivities agree with wavenumber integration results for a point source over a range of frequencies and source depths. The wider radiation pattern owes itself to the conversion of evanescent waves in the water into propagating waves in the air that fill the angular space outside the cone. A ratio of pressure measurements made using an on-axis microphone and a near-axis hydrophone are also reported and compared with theory. Collectively, these pressure measurements are consistent with the theory of anomalous transparency of the water-air interface in which a large fraction of acoustic power emitted by a shallow source is radiated into the air.
An axisymmetric boundary element formulation of sound wave propagation in fluids including viscous and thermal losses134(2013); http://dx.doi.org/10.1121/1.4823840View Description Hide Description
The formulation presented in this paper is based on the boundary element method (BEM) and implements Kirchhoff's decomposition into viscous, thermal, and acoustic components, which can be treated independently everywhere in the domain except on the boundaries. The acoustic variables with losses are solved using extended boundary conditions that assume (i) negligible temperature fluctuations at the boundary and (ii) normal and tangential matching of the boundary's particle velocity. The proposed model does not require constructing a special mesh for the viscous and thermal boundary layers as is the case with the existing finite element method (FEM) implementations with losses. The suitability of this approach is demonstrated using an axisymmetrical BEM and two test cases where the numerical results are compared with analytical solutions.