The Journal Of The Acoustical Society Of America[JOURNAL]
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200 papers
Complex upper ocean sound-speed structure measured by gliders in the Canada Basin.
Pomales Velázquez LO, Webster SE, Van Uffelen LJ
J Acoust Soc Am · 2026 Apr · PMID 42023943 · Publisher ↗
The Canada Basin's upper ocean structure is undergoing swift changes with the intrusion of warmer Pacific waters and sea ice loss with profound implications for sound propagation in this region. Two Seagliders measured h...
The Canada Basin's upper ocean structure is undergoing swift changes with the intrusion of warmer Pacific waters and sea ice loss with profound implications for sound propagation in this region. Two Seagliders measured high-resolution transects of temperature, salinity, and pressure across fronts and eddies, providing estimates of the spatial sound-speed variability in the upper ocean along transmission paths of an acoustic tomography array during the summer months of 2016 and 2017. The spatial analysis highlights the importance of placing the results within the context of the structure of the Beaufort Gyre. The measured profiles are used to quantify the depth-dependent contributions of internal waves, halocline eddies, and spice on sound-speed fluctuations. Results support and complement measurements from a sub-surface distributed vertical line array mooring and extend observations into the mixed layer. In the upper 100 m, spice is found to be the major driver of sound-speed fluctuations with a maximum of 3 m/s rms at 23 m, which corresponds to the bottom of the mixed layer. Fluctuations from the vertical displacement of isopycnals driven by halocline eddies and internal waves have three distinct peaks at 25, 60, and 270 m, with values of 0.73, 0.43, and 0.2 m/s rms respectively.
Urban impulse responses measured via the synchronous addition of fireworks to access speech intelligibility in evacuation broadcast.
Shimokura R
J Acoust Soc Am · 2026 Apr · PMID 42017611 · Publisher ↗
Although an evacuation broadcast via an outdoor public address at the time of disaster is an important source of information, the long-path echoes from buildings and mountains reduce speech intelligibility. In assessing...
Although an evacuation broadcast via an outdoor public address at the time of disaster is an important source of information, the long-path echoes from buildings and mountains reduce speech intelligibility. In assessing speech intelligibility, it is proposed to use firework explosions to simulate urban impulse responses and calculate speech transmission indices (STIs). In this study, investigators were stationed across cities during firework festivals to record explosions, and the individual events were synchronously combined to improve signal-to-noise ratios, yielding urban impulse responses. The urban impulse responses had distinct echo-time patterns originating from buildings and surrounding mountains, and the estimated source power of the fireworks had a nearly flat energy spectrum. The sound exposure level (LE) of the impulse responses was consistent with values calculated using outdoor sound propagation models (ISO 9613-2). In addition, the STIs of the impulse responses had a highly accurate linear relationship with sound exposure levels, LE's. Therefore, it is important to estimate LE more precisely using geometrical data of cities in assessing the speech intelligibility of evacuation broadcasts.
A 2.5-dimensional acoustic wave solver: Modeling simplified vocal tract geometries with reduced computational load.
Mohapatra DR, Zappi V, Fels S
J Acoust Soc Am · 2026 Apr · PMID 42017610 · Publisher ↗
High-fidelity three-dimensional (3D) wave solvers accurately simulate acoustic wave propagation in complex vocal tract geometries but are computationally demanding, limiting their usage in real-time applications. In cont...
High-fidelity three-dimensional (3D) wave solvers accurately simulate acoustic wave propagation in complex vocal tract geometries but are computationally demanding, limiting their usage in real-time applications. In contrast, low-dimensional models are efficient but limited to cylindrical tracts, neglecting higher-order modes in their frequency responses. This paper introduces a lightweight lumped two-dimensional (2.5D) solver that combines the efficiency of low-dimensional models with the accuracy of 3D approaches to model straight tracts constrained to mid-sagittal symmetry. Like 3D, the 2.5D model captures transverse wave propagation and accounts for higher-order modes. We validate the model by comparing its transfer functions and pressure distributions against those of a conventional two-dimensional (2D) solver and a high-fidelity 3D finite element model for six straight tract geometries of varying complexity. This analysis demonstrates the abilities and limitations of the proposed method. The results show that the 2.5D solver closely matches the 3D model's transfer functions up to 12 kHz, with correlation coefficients exceeding 0.8 for symmetric tracts. For asymmetric geometries, it still performs significantly better than the 2D model. Additionally, the 2.5D solver achieves over two orders of magnitude computational speed-up compared to the 3D model, offering a better trade-off between accuracy and efficiency for vocal tract acoustic modeling.
Blind source separation in complex marine soundscapes: An unsupervised two-stage clustering based on non-negative matrix factorization.
Huang B, Li Z, Wang X
J Acoust Soc Am · 2026 Apr · PMID 42017609 · Publisher ↗
Soundscape monitoring assesses biodiversity by analyzing environmental acoustic signals, but overlapping sound sources in complex environments limit the performance of traditional methods. We propose an unsupervised blin...
Soundscape monitoring assesses biodiversity by analyzing environmental acoustic signals, but overlapping sound sources in complex environments limit the performance of traditional methods. We propose an unsupervised blind source separation algorithm using nonnegative matrix factorization (NMF) and a two-stage coarse-to-fine clustering strategy. First, NMF decomposes the mixed spectrogram into spectral bases and temporal activations. In the clustering stage, coarse clustering is first performed via a second NMF with sparsity constraints using the spectral bases, temporal activations, or their derived features. Subsequently, fine clustering is performed using hierarchical clustering-guided K-means, which leverages complementary feature dimensions to refine the initial groups. Performance was evaluated on both simulated data and real-world recordings using separation quality metrics, detection metrics, and a composite score. Robustness was further examined under different mixture complexities. Results demonstrate that the proposed method achieves superior separation performance compared to one-stage clustering on both simulated and real-world data, particularly in successfully recovering a greater number of source components. This work provides a practical approach for fine-grained source separation in complex soundscapes and supports quantitative ecoacoustic analysis.
Active acoustic enhancement systems: A review.
Cassidy WJ, De Bortoli GM, Prawda K … +5 more , Coleman P, Mason R, Lokki T, Schlecht SJ, De Sena E
J Acoust Soc Am · 2026 Apr · PMID 42017608 · Publisher ↗
Active acoustic enhancement systems (AAESs) use microphones, loudspeakers and electronic processing to modify the reverberation of a space, offering flexible and cost-effective alternatives to passive variable acoustics....
Active acoustic enhancement systems (AAESs) use microphones, loudspeakers and electronic processing to modify the reverberation of a space, offering flexible and cost-effective alternatives to passive variable acoustics. These systems can extend the reverberation time of a space and modify perceived characteristics such as wall distance, diffuseness and intimacy. In this article, the current literature is discussed, and common conditions of AAESs are demonstrated using simulations to help researchers to establish a comprehensive understanding of the field. A general model is first defined to approximate any AAES as a linear, time-invariant system of transfer functions. This is used to analyse the general stability condition, which is valuable for system tuning and prediction. The three main topologies of AAESs are presented, namely, in-line, regenerative and hybrid systems, describing the fundamental differences as well as the nuances of commercial implementations with a focus on signal processing techniques. Articles investigating AAESs have been summarised to allow readers to gauge the coverage of experimental research to date. The simulated contribution serves as an exploratory environment to compare AAES conditions, where code and audio examples are available online. Promising future trajectories are identified involving machine learning, artefact perception and expressive performance.
Voice clones are easier to understand in noise than their human originals: The voice cloning intelligibility benefit.
Adank P, Wang H
J Acoust Soc Am · 2026 Apr · PMID 42012867 · Publisher ↗
Voice cloning technology has developed rapidly and can currently produce high-quality humanlike voices from as little as 10 s of speech. It is unclear whether cloned voices are as intelligible as their human originals. W...
Voice cloning technology has developed rapidly and can currently produce high-quality humanlike voices from as little as 10 s of speech. It is unclear whether cloned voices are as intelligible as their human originals. We compared the intelligibility of ten human voices with their ten voice clones in background noise. Eighty participants listened to 80 sentences (40 human, 40 cloned), presented in four signal-to-noise ratios (+3, 0, -3, and -6 dB) in an online experiment. Cloned voices were up to 13.4% more intelligible than their human counterparts across all noise levels. Principal component analysis with linear discriminant analysis classified human and cloned voices correctly in 79.4% of cases based on an extensive set of acoustic measurements, confirming systematic acoustic differences between the two voice types. Human listeners identified human voices with 70.4% accuracy. Elastic net regression analyses indicated that intelligibility in cloned voices was driven mainly by pitch and harmonic measures, whereas formant- and vowel-space measures were more influential for human voices. Our findings have implications for applications of voice cloning, including voice restoration, speech synthesis for non-verbal individuals, and accessibility for people with hearing loss.
Sensitivity to interaural phase as a function of frequency: Age effects measured with behavioral and electrophysiological tasks.
Grose JH, Buss E
J Acoust Soc Am · 2026 Apr · PMID 42012443 · Publisher ↗
Temporal fine-structure processing, as measured with binaural tasks, declines with increasing age in adults. The purpose of this study was to determine whether an electrophysiological test of binaural temporal processing...
Temporal fine-structure processing, as measured with binaural tasks, declines with increasing age in adults. The purpose of this study was to determine whether an electrophysiological test of binaural temporal processing could be used as a proxy measure for behavioral performance to reliably capture this age dependence. The behavioral measure was the upper frequency limit for differentiating in-phase from out-of-phase tones. This was assessed in both quiet and in background noise. The electrophysiological measure was the acoustic change complex elicited by epochs of interaurally out-of-phase frequency modulation carried by tones of different frequencies. Adults with normal/near-normal hearing were tested on both measures. There were 20 participants in each of the three age groups categorized as young, middle-aged, and older. The upper frequency limit measured behaviorally declined with age. It also declined in the presence of background noise, but the noise effect was equivalent across age groups. The robustness of the acoustic change complex also declined with age and as a function of the carrier frequency. However, correlations between the behavioral and electrophysiological measures were modest at best, suggesting that the electrophysiological test as implemented in this study did not provide a robust proxy for behavioral performance.
Discrimination and suppression of reverberation by leveraging the property disparity between target echoes and rough seabed reverberation.
Zou Y, Li X, Yu G
J Acoust Soc Am · 2026 Apr · PMID 42012442 · Publisher ↗
In shallow water environments, reverberation, which is generated from the rough seabed, remains one of the major sources of background interference in active sonar systems. This paper proposes a reverberation suppression...
In shallow water environments, reverberation, which is generated from the rough seabed, remains one of the major sources of background interference in active sonar systems. This paper proposes a reverberation suppression method that leverages multidimensional property differences, designed for a rough seabed. The approach establishes a link between variations in seabed reflection and scattering coefficients and the spectral property of reverberation signals, providing a physical basis for effective suppression. Using the Wigner-Ville distribution, reverberation suppression is formulated as the identification and utilization of time-frequency feature disparities between reverberation and target echoes. The method consists of two stages. First, seed region growing is enhanced using a multi-feature framework and Rényi entropy to suppress reverberation in the time-frequency plane. The multi-feature framework evaluates the stability and similarity of the instantaneous frequency sequence to screen the initial target echo position, whereas the Rényi entropy of the time-frequency domain adaptively adjusts the growing thresholding. In the second stage, the extracted time-frequency results are combined with a de-chirp procedure to reconstruct the time-domain echoes. Simulation and experimental results under various shallow water conditions demonstrate that the proposed method effectively suppresses reverberation while accurately preserving the target echo.
Binaural sensitivity and processing of envelope-based interaural difference cues by bilateral cochlear implant users with perilingual and postlingual onset of deafness.
Peng ZE, Kan A, Litovsky RY
J Acoust Soc Am · 2026 Apr · PMID 42007672 · Publisher ↗
Cochlear implants deliver binaural cues primarily in the signal envelope through two unsynchronized sound processors, presenting challenges to bilateral cochlear implant (BiCI) users to (re)gain binaural hearing. Using a...
Cochlear implants deliver binaural cues primarily in the signal envelope through two unsynchronized sound processors, presenting challenges to bilateral cochlear implant (BiCI) users to (re)gain binaural hearing. Using a custom-designed acoustic complex to elicit multi-channel stimulation through the cochlear implant sound processors, this study measured sensitivity to interaural time difference (ITD) and interaural level difference (ILD) using a discrimination task while concurrently tracking eye gaze positions to reveal decision-making delays during binaural processing. Two groups of BiCI users were tested with either perilingual or postlingual onset of deafness. Results show all BiCI users exhibited ILD sensitivity and a majority (67% of the perilingual and 78% of the postlingual) demonstrated ITD sensitivity, though those with perilingual onset showed higher (poorer) ITD sensitivity compared to postlingual users. Additionally, ILD processing at suprathreshold magnitudes led to faster decision making as accuracy increased for both groups. However, ITD processing was less salient, with faster ITD processing observed only in the postlingual group. These findings provide insights into the differences in binaural processing in users based on the timing of deafness onset.
Deep learning-based environmental source separation and sound enhancement: Advancements for cochlear implant and normal hearing listeners.
Shekar RCMC, Hansen JHL
J Acoust Soc Am · 2026 Apr · PMID 42007671 · Publisher ↗
Humans perceive non-linguistic sounds (NLSs) by associating auditory events with corresponding physical sources in a complex acoustic environment. However, previous studies have shown that cochlear implant (CI) users, vs...
Humans perceive non-linguistic sounds (NLSs) by associating auditory events with corresponding physical sources in a complex acoustic environment. However, previous studies have shown that cochlear implant (CI) users, vs normal hearing (NH) listeners, can face more severe challenges in identifying and tracking NLS. For CI listeners, this leads to limited autonomy, environmental awareness, safety, contextual navigation and daily engagement with individuals, society, and environmental situations. In earlier work, we studied NLS classification among CI and NH listeners and proposed a NLS enhancement solution to benefit CI/NH listeners. Building on this foundation, we propose here an experimental framework to investigate competing environmental sounds or NLS perception among CI and NH listeners. We introduce a two-source mixture model featuring "target" and "interference" source characteristics and develop an experimental setup for listener evaluation in three conditions: (i) mixed-baseline, (ii) source separation (SS) using the SUccessive DOwnsampling and Resampling of Multi-Resolution Features network, and (iii) source separation with non-linguistic sound enhancement (SSE) achieved by cascading SS output with our previously developed NLS enhancement technique. CI and NH listener evaluations were based on subjective ratings and forced-choice preference test based on perceptual measures: (i) interference, (ii) audio quality, and (iii) distortion. Our study shows a statistically significant improvement in interference reduction, with CI listeners demonstrating reduction for "nature" sounds with "category-matched" interference [F(2,21) = 4.935, p = 0.0175], and NH listeners exhibiting reductions across all NLS categories, with F-values ranging from [F(2,135) = 8.481, p = 0.000 339] to [F(2,135) = 32.37, p = 3.29 × 10-12]. Pairwise forced-choice test revealed preferences for SSE-processed nature and "domestic noises" among both CI and NH listeners. Our proposed experimental framework addresses key challenges in competing environmental sound perception among CI and NH listeners: (1) evaluation of SS for interference-characterized NLS mixture, (2) evaluation of environmental sound or NLS enhancement framework to improve perceptual outcomes with speech-targeted CI processing, and (3) perceptual measures to characterize NH and CI listener experience.
Bayesian machine learning framework for time-domain prediction of multirotor vehicle noisea).
Lee H, Ko J, Seshadri P … +1 more , Rauleder J
J Acoust Soc Am · 2026 Apr · PMID 41995685 · Publisher ↗
This work presents a Bayesian machine learning framework developed to predict aeroacoustic time-series signals generated by a quadrotor vehicle in forward flight at varying velocities. In this effort, a Gaussian process...
This work presents a Bayesian machine learning framework developed to predict aeroacoustic time-series signals generated by a quadrotor vehicle in forward flight at varying velocities. In this effort, a Gaussian process (GP) regression model is trained using a database of simulated signals produced by the Comprehensive Multi-rotor Noise Assessment framework. Unlike traditional frequency-domain models, the GP model directly predicts the time-domain signal, inherently capturing both amplitude and phase information of relevant frequency components. This capability is achieved by partitioning the tonal and broadband components during pre-processing, and capturing each component via a blade passage frequency-informed Fourier kernel and a Gaussian likelihood model, respectively. The resulting model is probabilistic in nature, inherently capturing the associated prediction uncertainty. Quantitative evaluations demonstrate strong agreement with ground truth signals in both time and frequency domains, with mean loudness errors of 1.11% in decibels and 5.55% in sones. The mean psychoacoustic annoyance error is found to be approximately 10%. The model is also computationally efficient compared to traditional physics-based solvers, requiring 0.1803 s to generate a time-series signal sampled at 44 100 Hz on a single NVIDIA A100 GPU (NVIDIA, Santa Clara, CA).
Underwater sound levels of transiting crew transfer vessels.
Basan F, de Jong CAF, Krüger C … +1 more , Fischer JG
J Acoust Soc Am · 2026 Apr · PMID 41995288 · Publisher ↗
Underwater sound recordings from Helgoland, Germany, were analyzed to detect passages of crew transfer vessels (CTVs). From these opportunistic observations, the source levels of 13 individual vessels were derived using...
Underwater sound recordings from Helgoland, Germany, were analyzed to detect passages of crew transfer vessels (CTVs). From these opportunistic observations, the source levels of 13 individual vessels were derived using the smoothed semi-coherent image method, including frequency-dependent absorption. Statistical analysis, using both generalized additive models and random forest models, showed that vessel-specific differences are the primary source of variability in source levels. While speed, length, and propulsion type all influence source levels, their effects vary across vessels and frequency bands, with no single factor dominating overall. The results indicate that, despite their relatively small size, CTVs have radiated noise levels similar to larger cargo vessels. The low variability in source levels across vessels suggests that a single source level spectrum for transiting CTVs could be a viable input for future noise modelling efforts.
Explicit finite-difference method with time-step n-tupling and extended CFL stability limit for acoustic wave simulation.
Gao Y, Zhu MH, Zhang H
J Acoust Soc Am · 2026 Apr · PMID 41995287 · Publisher ↗
The explicit finite-difference scheme is widely used in seismic wave simulation. Using large time steps can significantly reduce the number of iterations and thus improve computational efficiency, but this approach faces...
The explicit finite-difference scheme is widely used in seismic wave simulation. Using large time steps can significantly reduce the number of iterations and thus improve computational efficiency, but this approach faces two main challenges: reduced accuracy caused by temporal dispersion and restrictions imposed by the Courant-Friedrichs-Lewy (CFL) stability condition. Based on the wavefield iteration equation in matrix form, an explicit finite-difference method with time step n-tupling and an extended CFL limit is developed for acoustic wave simulation. By combining n successive iteration operators into a single-step operator, two types of n-tupling algorithms are constructed, effectively expanding the CFL stability limit by a factor of n and enabling time steps well beyond conventional thresholds. Both theoretical analysis and numerical experiments demonstrate that simulations with time step n-tupling achieve accuracy equivalent to conventional single-step methods while reducing computation time to approximately 1/n, resulting in a corresponding n-fold increase in overall computational efficiency. Time dispersion is suppressed using two complementary strategies: for relatively small base time steps, n-tupling inherently attains the accuracy of smaller steps; for relatively large base steps, a time-dispersion transform is applied to eliminate errors and maintain high numerical accuracy throughout the simulation.
Distinctive acoustic multipath propagation over a low velocity layer of sediments in deep ocean.
Zhan J, Piao S, Dong Y … +2 more , Gong L, Guo Y
J Acoust Soc Am · 2026 Apr · PMID 41989424 · Publisher ↗
Very-low-frequency (VLF) (in the range 1-100 Hz) sound propagation is significantly influenced by the properties of the seafloor, especially the presence of low velocity bottom (LVB) layers. During a VLF sound propagatio...
Very-low-frequency (VLF) (in the range 1-100 Hz) sound propagation is significantly influenced by the properties of the seafloor, especially the presence of low velocity bottom (LVB) layers. During a VLF sound propagation experiment in the northwestern sub-basin of the South China Sea (about 3800 m), a distinctive acoustic multipath excited by a source near the sea surface was observed for the first time. This phenomenon is characterized by the arrival of low-frequency components preceding high-frequency components with bandpass effects around 30-60 Hz, as recorded by an ocean-bottom-seismometer that received signals from broadband explosive sources deployed at approximately 200 m. Numerical simulations reveal that this multipath is formed by the sediment borne mode in the LVB layer, which corresponds to the distinctive acoustic path that refracts in the water column and interacts with the LVB layer at a small grazing angle. Unlike previous studies in an isovelocity profile, the sediment borne mode oscillates with a certain amplitude distribution rather than exponentially decaying in the water column under the incomplete sound channel, leading to the observation of distinctive multipath even when the source is near the surface. Furthermore, through geoacoustic inversion, the sound velocity of the LVB layer aligns well with previous in situ measurements.
Tunable acoustic logic gates based on superimposed Mie resonator.
Lan J, Zhou Y, Liu C … +1 more , Li Y
J Acoust Soc Am · 2026 Apr · PMID 41989224 · Publisher ↗
Traditional acoustic logic gates are constrained by their fixed structures, thereby impeding the pursuit of integrated acoustic systems. This work presents a symmetric three-port waveguide that achieves tunable acoustic...
Traditional acoustic logic gates are constrained by their fixed structures, thereby impeding the pursuit of integrated acoustic systems. This work presents a symmetric three-port waveguide that achieves tunable acoustic logic gates. The waveguide consists of one rectangular straight waveguide and two arc-shaped waveguides, all without cladding layers. Each waveguide is filled with superimposed Mie resonators (SMRs) exhibiting anisotropic dispersion characteristics. A wave interference model is established to modulate the phase by adjusting the spacings between the SMRs in the two arc-shaped waveguides. Based on a linear interference mechanism, four basic logic functions of OR, XOR, NOT, and AND and a combinational logic function of O = (A AND B) OR (C AND D) are realized within the waveguide structure. This work overcomes the inherent limitations of fixed-structure designs and establishes a valuable theoretical and technological pathway for integrating acoustic physics with advanced acoustic circuits.
Effect of frequency-to-place mismatch on speech and music sound quality in acoustic cochlear implant simulation.
Villejoubert L, Picinali L, Faulkner K … +1 more , Vickers D
J Acoust Soc Am · 2026 Apr · PMID 41989223 · Publisher ↗
Sound quality perception for cochlear implant (CI) users has become increasingly important. Although many CI users achieve near-normal speech recognition in quiet, they often report poor sound quality, particularly for m...
Sound quality perception for cochlear implant (CI) users has become increasingly important. Although many CI users achieve near-normal speech recognition in quiet, they often report poor sound quality, particularly for music. One factor contributing to this degradation is frequency-to-place mismatch (FTPM), which occurs when electrode positions do not align with the cochlea's characteristic frequency map. This study aimed to better understand the impact of FTPM on sound quality in CI simulations across different signal types and configurations. Twenty-three normal-hearing participants were tested (online or onsite) using an adapted MUlti Stimulus test with Hidden Reference and Anchor (MUSHRA) paradigm. Different FTPM configurations were simulated with a noise vocoder to assess their influence on speech and music sound quality. Results showed that greater FTPM caused noticeable degradation, especially when lower frequencies were affected. Variability in FTPM across electrodes also significantly reduced perceived quality. Furthermore, the impact of FTPM depended on the type of stimulus, with speech and music showing distinct sensitivity patterns. Online assessments closely matched onsite results, confirming the reliability of remote testing. Together, these findings clarify why sound quality perception differs between CI users and contexts and highlight new opportunities to develop strategies for alleviating mismatch effects in CIs.
A Bayesian optimization framework for two-dimensional sparse array design driven by physics-based simulation.
Zeng Z, Hu Q, Wang Z … +3 more , Yu Y, Zhang J, Ren C
J Acoust Soc Am · 2026 Apr · PMID 41989222 · Publisher ↗
Traditional design of underwater two-dimensional arrays focuses on local beam-pattern metrics, with insufficient attention paid to the overall imaging performance in practical acoustic environments. To mitigate this limi...
Traditional design of underwater two-dimensional arrays focuses on local beam-pattern metrics, with insufficient attention paid to the overall imaging performance in practical acoustic environments. To mitigate this limitation, this paper proposes a collaborative array design framework that integrates physical acoustic simulation with Bayesian optimization. Employing a Gaussian process as the surrogate model, the framework incorporates a dual-stage delay optimization strategy and a composite loss function to enable the automatic search for optimal array configurations. Simulation experiment results demonstrate that within a 90° × 90° field of view, the proposed DSDO method reduces the maximum delay mean square error by approximately 66.7% compared with the Fresnel approximation. The segmentation performance metrics (peak signal-to-noise ratio, structural similarity, intersection over union, and accuracy) of the images obtained by the optimized array show improvements of 6.49%, 1.62%, 6.21%, and 2.17%, respectively, compared to the Fermat spiral array. This indicates that the method enhances the clarity and structural fidelity of targets within the images, laying a foundation for subsequent downstream tasks, such as target detection and recognition.
Integrated high-update-rate ultra-short baseline positioning and robust communication using underwater acoustic direct-sequence spread-spectrum signals.
Deng J, Zhang J, Cui H … +3 more , Sun D, Wu J, Zhang Z
J Acoust Soc Am · 2026 Apr · PMID 41984046 · Publisher ↗
Integrated positioning and communication (IPAC) is critical for underwater unmanned platforms with severe constraints on space, power, and bandwidth. Conventional IPAC typically processes positioning and communication se...
Integrated positioning and communication (IPAC) is critical for underwater unmanned platforms with severe constraints on space, power, and bandwidth. Conventional IPAC typically processes positioning and communication separately, leading to a fundamentally low positioning update rate. In this paper, we propose a receiver design for ultra-short baseline (USBL) systems that fully exploits standard direct-sequence spread-spectrum communication signals. To simultaneously support both robust communication and high-update-rate positioning, the core challenge lies in obtaining high-precision and unwrapped phase information from the modulated signal. To address this, a per-survivor processing algorithm is employed to jointly and reliably separate the transmitted bits from the phase induced by the channel. Then, a cascaded phase unwrapping method resolves the phase wrapping for each symbol across all receiving hydrophones. This process generates symbol-by-symbol unwrapped phase that enables high-accuracy and high-update-rate USBL positioning. Simulation and experimental results demonstrate that the proposed receiver achieves simultaneous USBL positioning with a nearly 20 Hz update rate and robust communication. The positioning accuracy remains consistently high and is unaffected by the modulation order, validating the method's efficacy.
Effect of thermal modifications on the anisotropic acoustic properties of spruce and their relevance for stringed instruments.
Quintavalla M, Viala R, Santini M … +1 more , Bonanomi S
J Acoust Soc Am · 2026 Apr · PMID 41984045 · Publisher ↗
Thermal modification alters wood properties and helps improving dimensional stability against humidity changes, making it a promising treatment for wood used in stringed instrument construction. Its effects on the anisot...
Thermal modification alters wood properties and helps improving dimensional stability against humidity changes, making it a promising treatment for wood used in stringed instrument construction. Its effects on the anisotropic mechanical and acoustical properties, however, remain incompletely understood. In this study, spruce samples were analyzed to quantify changes in physical, mechanical, and acoustical properties following thermal modification at 160 °C. Density, orthotropic viscoelastic constants, and damping were measured using non-destructive techniques, while microstructural effects were examined via x-ray microtomography. Finite element analysis of a guitar soundboard assessed impacts on eigenfrequencies, mode shapes, and acoustic radiation. Results indicate that thermal treatment causes a slight reduction in density, a modest increase in longitudinal and radial stiffness, and a significant decrease in damping, leading to enhanced radiation ratio and acoustic conversion efficiency. Microstructural observations suggest that removal of resins and volatile extractives may underlie these changes. Finite element analysis shows that eigenmodes shape remain largely unchanged, with only minor shifts in eigenfrequencies. The combination of improved radiation efficiency and reduced damping could influence sound radiation and string-to-soundboard coupling.
Characterizing tendon microstructure using metrics associated with the angular dependence of ultrasound backscatter.
Wayson-Madejski SE, Helguera M, Raeman CH … +4 more , Jackson T, Chwalek J, Hocking DC, Dalecki D
J Acoust Soc Am · 2026 Apr · PMID 41984044 · Publisher ↗
A high-frequency, quantitative ultrasound technique, based on the angular dependence of the ultrasound backscatter (AIB), was developed to characterize tendon microstructural alignment and detect alterations in alignment...
A high-frequency, quantitative ultrasound technique, based on the angular dependence of the ultrasound backscatter (AIB), was developed to characterize tendon microstructural alignment and detect alterations in alignment with diabetes and aging. Backscattered echoes (58-MHz, bandwidth 31-86 MHz) were obtained from murine tail tendon, skin, and liver at 31 insonation angles. AIB parametric images and the average AIB for a region-of-interest were computed at each angle. AIB measurements were angular-dependent in tendons having aligned collagen fibers and were angular-independent in skin and liver. Quantitative ultrasound metrics characterizing AIB as a function of insonation angle included (1) AIB¯N,max, the maximum AIB; (2) ΔAIB¯N, the change in AIB within ten degrees of its maximum; and (3) M, the linear rate of change of the derivative of the AIB as a function of insonation angle. All three metrics were significantly different in murine aged, wild-type tendons and young, diabetic tendons compared to young, wild-type tendons, indicating that scattering strength was reduced, and tendon disorganization was increased in aged, wild-type tendons and young, diabetic tendons compared to young, wild-type tendons. This work demonstrates the utility of an ultrasound technique employing metrics associated with the angular dependence of ultrasound backscatter to characterize tendon microstructure non-invasively.
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