Journal of the Acoustical Society of America最新文献

This paper postulated and tested the possibility that the mouth rhythm functions as a "packaging mechanism" of information in speech. Cross-spectral analysis between two time series of mouth aperture size [parameterized as sample-by-sample interlip distances, i.e., o(t)] and information variations in speech [parameterized as frame-by-frame spectral entropy values, i.e., h(t)] was employed to reveal their underlying spectro-temporal relationship. Using a corpus containing more than 1000 utterances produced by a typical British English speaker, it was observed that both signals share slow recurring rates corresponding to the stress and syllable, with a slight phase lag of h(t) behind o(t) in the vicinity of 5 Hz.

Listeners can adapt to noise-vocoded speech under divided attention using a dual task design [Wang, Chen, Yan, McGettigan, Rosen, and Adank, Trends Hear. 27, 23312165231192297 (2023)]. Adaptation to noise-vocoded speech, an artificial degradation, was largely unaffected for domain-general (visuomotor) and domain-specific (semantic or phonological) dual tasks. The study by Wang et al. was replicated in an online between-subject experiment with 4 conditions (N = 192) using 40 dysarthric sentences, a natural, real-world variation of the speech signal listeners can adapt to, to provide a closer test of the role of attention in adaptation. Participants completed a speech-only task (control) or a dual task, aiming to recruit domain-specific (phonological or lexical) or domain-general (visual) attentional processes. The results showed initial suppression of adaptation in the phonological condition during the first ten trials in addition to poorer overall speech comprehension compared to the speech-only, lexical, and visuomotor conditions. Yet, as there was no difference in the rate of adaptation across the 40 trials for the 4 conditions, it was concluded that perceptual adaptation to dysarthric speech could occur under divided attention, and it seems likely that adaptation is an automatic cognitive process that can occur under load.

Breast cancer is a leading cause of death for women. Quantitative ultrasound (QUS) and ultrasound computed tomography (USCT) are quantitative imaging techniques that have been investigated for management of breast cancer. QUS and USCT can generate ultrasound attenuation images. In QUS, the spectral log difference (SLD) is a technique that can provide estimates of the attenuation coefficient slope. Full angular spatial compounding (FASC) can be used with SLD to generate attenuation maps with better spatial resolution and lower estimate variance. In USCT, high quality speed of sound (SOS) images can be generated using full wave inversion (FWI) method, but attenuation images created using FWI are often of inferior quality. With the QTI Breast Acoustic CTTM Scanner (QT Imaging, Inc., Novato, CA), raw in-phase and quadrature data were used to implement SLD combined with FASC. The capabilities of SLD were compared with FWI through simulations, phantom experiments, and in vivo breast experiments. Results show the SLD resulted in improved accuracy in estimating lesion sizes compared to FWI. Further, SLD images had lower variance and mean absolute error (MAE) compared to FWI of the same samples with respect to the attenuation values (reducing MAE by three times) in the tissue mimicking phantoms.

We describe and implement a numerical method for modelling the frequency-dependent power-law absorption of ultrasound in tissue, as governed by the first order linear wave equations with a loss taking the form of a fractional time derivative. The (Caputo) fractional time derivative requires the full problem history, which is contained within an iterative procedure. The resulting numerical method requires a fixed (static) memory cost, irrespective of the number of time steps. The spatial domain is treated by the Fourier spectral method. Numerically. comparisons are made against a model for the same power-law absorption with loss described by the fractional-Laplacian operator. One advantage of the fractional time derivative over the fractional-Laplacian operator is the local treatment of the power-law, allowing for a spatially varying frequency power-law.

In this paper, we propose an approximate Bayesian soft symbol detector (SSD) for multi-input multi-output (MIMO) orthogonal frequency division multiplexing underwater acoustic communications over fast time-varying channels. It is enabled by vector approximate message passing (VAMP) algorithm coupled with expectation-maximization (EM) technique, thus is named the EM-VAMP-SSD. The EM-VAMP-SSD consists of an inner soft slicer and an inner soft estimator (ISE), iteratively exchanging extrinsic information to improve symbol estimation performance. The operation of the ISE requires knowledge about channel and noise power, which are estimated via the orthogonal matching pursuit algorithm and the EM algorithm, respectively. To reduce the complexity of ISE, the frequency-domain MIMO channel is approximated by a block diagonal matrix without sacrificing performance much. The proposed EM-VAMP-SSD has been verified by both numerical and experimental data, showing consistent performance advantage over a conventional linear minimum mean square error SSD.

Benthic biological processes influence seabed heterogeneity and contribute to variability in geoacoustic properties. To investigate these relationships, measurements were conducted to quantify spatial variability in the upper few decimeters of sediment near the water-seabed interface within a fine-grained sediment deposit on the New England continental shelf. At each measurement location, an acoustic multicorer was deployed to sample the seabed. Acoustic probes were inserted into the sediment to collect direct in situ measurements of sediment compressional wave speed and attenuation (30-100 kHz) under near-ambient conditions, after which cores were collected from the inter-probe propagation paths. Sediment physical properties, organic carbon, infaunal community composition, and ex situ compressional wave speed and attenuation spanning two frequency decades (104-106 Hz) were subsequently measured in the laboratory. The frequency dependence of sound speed ratio and attenuation was analyzed in the context of sediment acoustics models for mud based on the viscous grain shearing and extended Biot models. Sites with greater abundance of larger-bodied infauna (>1 mm) displayed higher variability in sound speed and attenuation. Correlation was found between sediment compressional wave modulus and total organic carbon, suggesting that organic matter in the sediment matrix also affects bulk acoustic properties.

Continuous phase modulation (CPM), which is widely used in aviation telemetry and satellite communications, may help improve the performance of underwater acoustic (UWA) communication systems owing to its high spectral and power efficiency. However, applying conventional frequency-domain equalization (FDE) algorithms to CPM signals over time-varying UWA channels considerably degrades performance. Moreover, time-domain equalization algorithms often rely on excessive approximations for symbol detection, compromising overall reception. This study presents an iterative-detection-based time-domain adaptive decision feedback equalization (ID-TDADFE) algorithm that tracks channel variations through symbol-by-symbol detection. The symbol detection in ID-TDADFE fully considers the inherent coding gain of CPM signals can be cascaded with an adaptive equalizer, and enhances symbol detection performance by utilizing joint probability estimation. Numerical simulations with minimum-shift keying (MSK) and Gaussian MSK signals demonstrated that ID-TDADFE significantly improved communication performance over a time-varying UWA channel within one or two iterations. In a sea trial for experimental verification, ID-TDADFE reduced bit errors by 45.08% and 51.8% in the first and second iterations, respectively, compared to FDE.

In a recent research paper, we introduced the statistical wave field theory, which establishes the statistical laws of waves propagating in a bounded volume. These laws hold after many reflections on the boundary surface and at high frequency. The statistical wave field theory is the first statistical theory of reverberation that provides the closed-form expression of the power distribution and the correlations of the wave field jointly over time, frequency and space, in terms of the geometry and the specific admittance of the boundary surface. In this paper, we refine the theory predictions, by investigating the impact of a curved boundary surface on the wave field statistics. In particular, we provide an improved closed-form expression of the reverberation time in room acoustics that holds at lower frequency.

The exact expressions of the three-dimension acoustic radiation torque (ART) of a viscoelastic sphere arbitrarily positioned in a zero-order Mathieu beam (zMB) are derived in this paper. The effects of the ellipticity parameters, half-cone angles, dimensionless frequency, and particle position on the acoustic radiation torques of the spherical particle are studied. Simulation results show the axial ART is zero for an arbitrarily positioned viscoelastic PE sphere in a zMB, while for the x or y axis ART, it varies significantly with the particle position and beam parameters. For certain combinations of beam offset and parameters, axial and transverse torques alternate between positive and negative values as the half-cone angle varies. When ka is away from the resonance frequency, the value of the torque is approximately 0.001, which means the torque is small and the particle can be rotated in a uniform angular acceleration. Moreover, ART shows symmetrical about beam center when the offset is less than one wavelength. A finite element model was established to verify the theory and the comparative results agreed with each other except for the values of ART at the first resonant frequency, which is related to the absorption of the particles. The study helps to better understand the potential mechanism of the particle rotation manipulation in a zMB.

Traditional underwater acoustic source localization methods based on time differences of arrival (TDOA) in the presence of refraction first estimate the source depth and range to each hydrophone and then estimate the horizontal location of the source. The accuracy of these methods is compromised by errors in range estimation. To address this, we propose a three-dimensional source localization method that utilizes TDOA measurements between direct and surface-reflected arrivals at N(N ≥3) hydrophones, taking into account refraction effects. By utilizing multipath signals reflected off the sea surface, the method considers hydrophone position errors, TDOA measurement inaccuracies, and sound-speed variations to perform a Bayesian maximum a posteriori estimation of source localization. Compared with the traditional two-step source localization methods, the proposed method directly estimates the source depth and horizontal location jointly, eliminating the need to estimate ranges between the source and hydrophones. Simulation studies analyzing and comparing the localization performance of the proposed method with that of a two-step source localization method demonstrate the effectiveness of the proposed method. This could lead to more reliable localization of underwater sources, crucial for various applications, such as marine research, underwater navigation, and environmental monitoring.