bluuebunny/crossref_metadata_2025 huggingface dataset:
Robust power spectral density estimation via a performance-weighted blend of order statistics
Authors: Anchieta David C. , Buck John
Published: November 2023

Loud transients introduce bias to background spectrum estimates based on the sample mean, like the Welch Overlapped Segment Averaging (WOSA) [1967]. The Schwock and Abadi [2021] Welch Percentile (SAWP) estimator avoids the loud transient bias by replacing the averaging of the WOSA with a scaled order statistic (OS) of the sample power spectrum. While the scaling ensures the SAWP is unbiased regardless of which OS is chosen, the 80th percentile minimizes the variance of the SAWP estimator in a scenario without loud transients. However, the 80th percentile SAWP is still vulnerable to bias and increased variance from more frequent loud transients. Also, the rate of occurrence of loud transients may change with time, requiring the SAWP to adapt which percentile is used. To approach this challenge, this talk proposes a Universal SAWP estimator which is a weighted sum across different fixed percentile SAWP estimators. At each iteration, the Universal SAWP updates the blend weights to promote the percentile with the lowest sample variance over recent observations. In computer simulations, the Universal SAWP quickly assigns higher weights to the lowest variance estimators as the occurrence of loud transients increases. Overall, the Universal SAWP achieves equal or lower variance as the best fixed percentile SAWP estimator. [Work supported by ONR Code 321US.]
Narrowband underwater vector sensor using michroelectromechanical systems
Authors: Alves Fabio , Ivancic Justin
Published: November 2023

A narrowband underwater vector sensor using microelectromechanical (MEMS) systems is demonstrated. A combination of two pressure gradient MEMS sensors and an omnidirectional hydrophone allows for determination of the direction of arrival (DOA) of income sound over 360 degrees azimuth. The MEMS sensors, inspired on the hearing system of the Ormia-ochracea fly, consist of two wings connected by a bridge and anchored to the substrate by a tortional bean. They are operated with open back to allow a cosine dependence with the angle of incidence in the predominant bending vibrational mode. In the vector sensor, two MEMS sensors are orthogonally arranged to provide a cosine and sine directional pattern. When combined with an omnidirectional hydrophone that shares the same phase center, an arctangent estimator is used for DOA determination. The MEMS sensors are operated near resonance to filter out noise coming from undesired bands and improve the signal-to-noise ratio. A correction algorithm is applied to compensate frequency response differences. Two configurations have been demonstrated with the MEMS sensors enclosed in silicone oil and in air. The standard deviation of the measured DOA error has been computed for both configurations to be less than 5 degrees for various types of sound stimuli.
Evolution of performance metrics in the sonar world
Authors: Felgate Nicholas J. , Panesar Haym S. , Holden Andrew P.
Published: November 2023

The performance of sonar technology, including passive acoustic monitoring, environmental impact assessment and the monitoring of man-made noise sources, is typically characterized by the through-water detection range. This single performance measure depends on the characteristics of the sound sources, the underwater environment and sensing capabilities. Measuring sonar performance is considered critical in the deployment, operation and development of sonar systems, otherwise one is danger of investing in the wrong technology and/or operating this technology incorrectly. It is however already known that using detection range by itself may be insufficient, for instance where the detection and classification tasks are inseparable and counter-detection needs to be taken into account. This talk will describe the different aspects of sonar performance that need to be considered and correctly assessed. In particular, ongoing work to identify better metrics for detection, classification and localization will be outlined. This work is looking initially to borrow metrics from other domains and applications, both for speed of development and retention of cross-domain compatibility, in particular from the area of Machine Learning. Finally, this talk will identify the major areas where further work may still be required.
Joint channel and Doppler shift estimation for underwater acoustic single carrier system
Authors: Cao Xinran , Xu Lijun
Published: November 2023

Underwater acoustic channel is a time-varying, strong multipath channel. Due to the low velocity of sound (about 1500 m/s), Doppler shift, occurred by the relative motion between transmitter and receiver, restricts the performance of communication. In order to transmit data in underwater acoustic single carrier communication system with strong stability and high reliability, especially in high mobility scenarios, a joint channel and Doppler shift estimation method is proposed in this paper. We propose a novel data frame structure based on a Gold sequence, so the channel and Doppler shift can be estimated. For different underwater scenarios, we design Gold sequence templates with different Doppler factors. When the Gold sequence from the received data frame is extracted, with good correlation characteristics of the Gold sequence, the maximum correlation value can be calculated, and the Doppler shift corresponding to the maximum is the estimation of Doppler shift. Then Doppler shift will be compensated by resampling the communication data. For channel estimation, the Gold sequence is also used to estimate the time-domain channel impulse response (CIR). Simulation results show that the proposed method can effectively estimate channel and Doppler shift in an environment with multipath and Doppler.
Acoustical characterization of historical performance structure of Taramati Baradari at Hyderabad, India
Authors: Akhtar Syed A. , Manohare Manish , Kawathekar Vishakha
Published: November 2023

The Baradari, a square-shaped building with twelve open doors, was designed for entertainment purposes and featured outstanding acoustic qualities, making it ideal for music and dance performances by noble courtesans. This article focuses on the acoustic characterization of Taramati Baradari, a historical performance structure in Hyderabad, India, built during the 17th century by the Qutb Shahi (Golconda) dynasty. The research aimed to assess the space's acoustical behaviour through in-situ measurements in an unoccupied state. Virtual acoustical models were developed and validated using the in situ measurements in the same condition, later simulated and analysed for both occupied and unoccupied scenarios, revealing an average reverberation time (RT60) of 4.1 s at mid-frequencies for unoccupied conditions. After analysing various acoustical parameters, it became evident that Taramati Baradari's acoustics possessed a distinctive character, giving the structure a unique identity. Unfortunately, in the Indian context, conservation and restoration efforts have primarily focused on visual aspects, often neglecting considerations of the space's acoustics. Therefore, characterizing the acoustic nature of performance structures becomes vital to inform sustainable interventions for conservation architects.
Harmony behind bars: Unravelling the impacts of acoustic design in prisons
Authors: Boland James , Farley Helen
Published: November 2023

Design briefs for prisons often include criteria regulating noise from mechanical plant or break-in, partition performance and reverberation times. However, criteria are often taken (without modification) from non-carceral related standards used for design of residential or health-care projects. While carceral spaces are used for sleeping and health care, this paper will show the acoustic needs (and wants) of people inside prisons can be significantly different from those of people outside. This paper provides insights from the emergent field of sensory criminology into what makes the acoustic environment inside prisons different from other residential or health care contexts. It highlights the role that acoustics plays in the lived experience for the occupants, including incarcerated individuals and corrections staff. In prisons, acoustics facilitates communication and information gathering for incarcerated individuals, and corrections staff rely on auditory cues to gauge the prevailing tension, or “heat.” The intricate dynamics of prison cultures and insights from sensory criminology can readily evade the purview of acousticians. By leveraging complementary disciplines, acousticians can more effectively design acoustics which foster pro-social communication while mitigating risks associated with undesirable social dynamics. This approach ensures the acoustic design becomes well-informed and purposeful, yielding benefits for incarcerated individuals and corrections staff.
A comparative assessment of NSW educational acoustic design guideline
Authors: Reverter Garcia Jorge
Published: November 2023

School Infrastructure New South Wales (SINSW) Educational Facilities Standard Guidelines (EFSG) Design Guideline 11 (DG-11) is the acoustic guideline for public educational premises in NSW. A comparative assessment of the guideline against alternative guidelines is conducted in this paper. The analysis identifies shortcomings stemming from an outdated guideline, as well as gaps in design clarity. Design solutions to meet the mandatory EFSG GD-11 acoustic requirements have been proposed to address the identified shortcomings. Additionally, we thoroughly examined DG-11's design specifications to resolve ambiguity and reduce the potential costly and over design. Our findings revealed weaknesses in SINSW DG-11 when compared to other nationally and internationally recognized standards. To create innovate and sustainable educational spaces that cater the users’ needs and enhance the functionality of educational spaces, fostering of a constructive dialog is required between stakeholders and acoustic designers while preparing acoustic guidelines.
Localizing multiple sound sources using an array of autonomous, asynchronous receivers
Authors: Nosal Eva-Marie , Fedenczuk Tom
Published: November 2023

Time-difference-of arrival (TDOA) methods are well established and commonly used to localize sound sources in bioacoustic applications. TDOA methods generally require/assume that the receivers of the acoustic array are time-synchronized. Time synchronization can be achieved in various ways, including by connecting the receivers to a common data aquisition device, or by using control/calibration sources of known location to time synchronize. In this talk, we introduce a TDOA approach that can be used with an array of autonomous, asynchronous receivers. The approach requires that multiple sources are detected over the array, and localizes the sources simultaneously in addition to solving for clock offsets between the receivers. This approach offers potential cost and time savings since it can be used with autonomous receivers and since it does not require control/calibrationsources. Simulation results will be used to illustrate and explore the approach, and results from an experiment using asynchronous microphones will be presented. [Work supported in part by ONR.]
Innovative AI techniques to aid clinical diagnosis of complex lung pathologies in ultrasound images
Authors: Antico Maria , Moafa Khalid , Vukovic Damjan , Edwards Christopher , Dowling Jason , Canty David , Steffens Marian , Fontanarosa Davide
Published: November 2023

Lung ultrasound (LUS) has recently gained increasing interest as a reliable method for point of care (POC) diagnostics and management of lung diseases. LUS is easily accessible, has no radiation-related risks and is portable, so it does not require relocating the patient, minimising the risk of further infection. This research aims to develop efficient and automated methods for lung pathology diagnosis using AI to support clinicians. We have introduced a binary classifier based on a state-of-the-art Swin Transformer to discriminate between LUS clips of healthy patients and patients with interstitial lung disease (ILD). Differently from previous approaches, this is better aligned with the current clinical assessments of ILD since it evaluates LUS clips instead of single frames; and does not require any additional annotations from clinicians, since its training is based only on the already available medical report. Furthermore, we propose an unsupervised deep learning approach (based on a generative adversarial network) to convert in real-time US volumes to MRI-like volumes of the thoracic region. This approach can be used to spatially localise the US probe with respect to the MRI in real-time and gather anatomical contextual information of the imaged region, providing thus guidance to clinicians.
Auditory brainstem response to different chirp (based on cochlear partition) stimulation rates
Authors: Mora Camargo Sergio , Ibarra-Zarate David I. , Alonso-Valerdi Luz Maria
Published: November 2023

All through the last decades, hearing thresholds have been estimated by audiologists by different means, such as pure-tone assessments, auditory steady state response, auditory brainstem response, etc. Objective EEG-based measurements have been applied, such as the auditory brainstem response (ABR), which measures neural synchrony along the brainstem auditory pathway. Through ABR waves’ amplitude and latency due to a stimulus level, hearing thresholds can be estimated. Considered a short-latencyauditory evoked potential (1–20 ms), very short transient stimulus (0.1–10 ms) is needed to evoke ABR. Chirp is a stimulus that has demonstrated to evoke better response, compared with clicks (aprox twice responses amplitude). Chirp is a transient auditory stimulus designed that considers delay in basilar membrane tonotopic gradient inside the cochlea. Chirp auditory response has been suggested to represent higher amplitudes, lower latency and lower time to identify thresholds. Characterization of ABR morphology due to chirp stimulus is something that has still been studied since multiple chirp designs have been proposed to identify changes in response due to chirp parameters differences. In the present work a description of Chirp design and detailed ABR extraction is presented, looking for a comparison between 4 Chirp stimulation rates as described in literature.
A mathematical model for maximizing drug uptake in low-EPR tumors using different methods of ultrasound-mediated drug delivery
Authors: Ghaderi Mohammadaref , Sirsi Shashank
Published: November 2023

Cancer remains a significant health burden and research efforts aim to improve drug delivery options. Focused ultrasound has emerged as a potential mechanism for enhanced targeted drug delivery in tumors with a low Enhanced Permeability and Retention (EPR) effect by “sonoporation”. Traditional sonoporation has shown promise, but it can still benefit from improvements in drug uptake. The current study investigates the use of a new non-portative technique, called “ultrasonic tumor painting,” to enhance vascular retention of nanoparticles in the tumor space to achieve targeted drug delivery and increase therapeutic efficacy. An existing mathematical model was adapted to simulate the concentration of free and liposome encapsulated doxorubicin in circulation. The simulations were performed in MATLAB and specifically evaluate doxorubicin uptake in low EPR tumors. To increase drug uptake, sonoporation was employed where efficiency of intratumoral liposomal doxorubicin uptake was increased. An alternative non-permeabilizing strategy, called “tumor painting,” was utilized to improve vascular retention instead of increasing permeability. After retention, the small molecule doxorubicin is allowed to leak into the tumors space via concentration mediated Fickian Diffusion. This study demonstrates that ultrasound-based drug delivery can be improved with tumor painting. This method is a valuable tool to overcome the limitations of traditional sonoporation and has exciting potential for enhancing the therapeutic efficacy of chemotherapy.
An application of dereverberation and multiplicative beamforming techniques for imaging noise sources generated by boat engines
Authors: Singh Rohit , Mimani Akhilesh
Published: November 2023

This paper demonstrates the application of a dereverberation algorithm for localizing noise sources emitted from two-stroke boat engines in three dimensions domain. The engines considered have a maximum RPM of 1500 and 1200 cc. It is important to note that these engines are operated with both petrol and CNG. The investigation primarily focuses on the noise generated under various loading conditions and aims to differentiate the noise characteristics between petrol and CNG engine operations. The dereverberation algorithm is applied to enhance the accuracy of noise source localization in this study, it involves filtering an approximate optimal window width to retain data related to the direct field while removing reflections through multiplication with an appropriate Hanning window. The analysis showed that the dereverberation algorithm effectively improves the imaging source map compared to conventional beamforming maps in the 3-D domain. To compute the beamforming output, data has been recorded from five different sides of the engine, and the multiplicative beamforming technique is employed. The findings highlight the significance of dereverberation and multiplicative beamforming techniques in accurately localizing noise sources in boat engine applications.
A comparison of background noise reduction algorithms for improving acoustic beamforming output
Authors: Singh Rohit , Mimani Akhilesh
Published: November 2023

This work presents a comparison of different background noise reduction algorithms available in literature at a range of signal-to-noise ratios (SNRs) for improving the conventional beamforming (CB) source maps. The algorithms include the classical background noise removal (BNR), eigenvalue identification organization and subtraction (EIOS), subspace-based background subtraction (SBS), and ensemble empirical mode decomposition (EEMD), amongst others. To assess the performance of different algorithms, three test-cases were considered: (1) localizing a loudspeaker placed in an anechoic environment in the presence of background white-noise, (2) diagnosing machine-tool noise source(s) during machining, i.e., when the workpiece and tool piece are in contact and (3) localizing airfoil trailing-edge (TE) noise sources in the presence of background tunnel noise. The analysis showed that the suitability of an algorithm depends upon the given situation – for the loudspeaker source, all algorithms deliver comparable results while for machine-tool application, the relatively simple BNR algorithm delivered the desired improvements in CB maps. For the experimental airfoil TE noise test-case, the EIOS and EEMD algorithms demonstrated substantial enhancements as compared to other methods. This investigation, therefore, suggests that for a given application, it is desirable to customize the background noise reduction algorithm to obtain optimal results.
Applications of intrinsic coordinates for simulation of thin shocks in relaxing media
Authors: Willis William A. , Cormack John M. , Hamilton Mark F.
Published: November 2023

The augmented Burgers equation that describes nonlinear propagation in a relaxing fluid can be expressed using intrinsic coordinates [Hammerton and Crighton, JFM 252, 585 (1993)]. A multivalued pressure waveform in physical coordinates can be represented as single-valued with intrinsic coordinates. Shocks can then be inserted into the multivalued pressure waveform using the equal area rule from weak shock theory to obtain a single-valued waveform solution, thus avoiding the high computational cost associated with discretizing thin shocks with conventional algorithms. By first solving the evolution equation in intrinsic coordinates, and then using the solution as the input to a conventional algorithm based on the augmented Burgers equation, a two-stage approach can be developed that achieves both high accuracy and significantly reduced computational cost when compared to solving the Burgers equation with conventional numerical algorithms alone. Approaches based on intrinsic coordinates are applied to realistic problems in relaxing media, which include air and seawater, to allow for comparisons of computational efficiency and accuracy. [WAW is supported by the ARL:UT Chester M. McKinney Graduate Fellowship in Acoustics.]
Acoustics of a pressurized air-filled pipe open at one-end at varying depths
Authors: Howard Carl , Craig Richard , Forrest James
Published: November 2023

A rigid pipe with a loudspeaker at one end and open at the other end, was submerged in a water tank at various depths. Microphones were placed along the length of the pipe to measure the acoustic response of the duct due to the acoustic excitation from an internal loudspeaker. Compressed air was injected into the pipe to equalise the internal pressure of the pipe caused by the water pressure acting on the open end of the pipe. The results were that the acoustic resonance frequencies of the duct did not alter with depth.
Comparative analysis of receiving characteristics of all flextensional transducer classes
Authors: Kim Gihyeon , Wang Wenbo , Kim Donghyun , Shim Hayeong , Roh Yongrae
Published: November 2023

Flextensional transducers have been widely used as low-frequency projectors, and their characteristics can be exploited to develop hydrophones with better sensitivity and wider receiving bandwidth than other existing hydrophones in the low-frequency band. In this study, we compared and analyzed the receiving characteristics of different classes of flextensional transducers to identify the most suitable class for wideband low-frequency hydrophones, particularly for broad applications. To begin, we established an initial model with the same peak receiving-voltage-sensitivity (RVS) frequency for each class of flextensional transducers. Through a thorough analysis of various structural variables, we examined their impact on the receiving characteristics of the transducers. Based on these findings, we selected the design variables with the most significant influence on the bandwidth for each class. Subsequently, we determined the optimal combination of these variables to maximize the receiving bandwidth while maintaining RVS at or above 1 kHz. Comparing the performance of the best-fit models across all classes, we found that class IV flextensional transducers exhibit the widest low-frequency receiving bandwidth, closely followed by the convex class I. The −3 dB receiving bandwidth of the class IV transducer amounts to 87.6% of the peak RVS frequency, while the convex class I transducer achieves 67.1%.
Manipulating the phase and group delay of a unit cell in metasurface via impedance engineering
Authors: Yang Dingcheng , Chiang Yankei , Powell David
Published: November 2023

The emergence of metasurfaces that tailor the phase via carefully controlling each unit cell brings promising applications including wavefront manipulation, acoustic lenses and levitations. Although excellent phase engineering has been reported in the literature, most metasurface designs suffer from narrow operating bandwidth. To achieve consistent performance over a broad bandwidth, it is necessary to engineer the group delay, in addition to the phase delay, however, few studies have considered this. The majority of them realized specific group delay by true time delay via tuning the depth of a cavity or peculiar topology-optimized structures. No systematic method of unit cell design has been proposed which is a significant step to broadband design. In this work, we proposed a unit cell design process incorporated with transfer matrix method and equivalent LC-circuit fitting. The transfer matrix method is a semi-analytical model capable of predicting the phase response if parameter lists are given. The LC-fitting interprets targeted impedance profile of unit cell by the equivalent inductance and capacitance, assisting the parametric sweeps for geometry design. A numerical study presented showing the mapping of LC values with two physical parameters, leading to control of phase and group delay simultaneously.
Design of volumetric sound metadiffusers using gradient-based optimization
Authors: Amirkulova Feruza , Gerges Samer , Jagannath Bharadawaj , Tran Matthew
Published: November 2023

Broadband volumetric sound diffusers are designed by using gradient-based optimization (GBO) by adjusting the position and radius of each scatterer in nonuniform configurations. The multiple scattering theory is employed to evaluate the diffusion coefficient by computing the scattered pressure field by nonuniform planar configurations of cylindrical scatterers for a monopole excitation. The analytical formulas for the gradients of the diffusion coefficient with respect to positions and radii of a cluster of cylindrical scatterers are derived which enhance the modeling when integrated with GBO and parallel computing. Sound volume metadiffusers with a high diffusion coefficient are designed by perturbatively rearranging the scatterer configurations. Single frequency and broad-octave-band optimizations are performed to maximize the diffusion coefficient while supplying analytical formulas for its gradients with respect to positions and radii. The GBO is implemented by direct optimization employing Multistart and fmincon solvers with sequential quadratic programming algorithms. The GBO approach is demonstrated providing numerical examples for nonuniform configurations of rigid cylinders embedded in the air environment. The polar plots for scattered pressure are evaluated for a wide range of ⅓ octave bands. The GBO approach can facilitate the automated metamaterial process by combining it with global optimization, generative modeling, and reinforcement learning.
Measuring listeners’ distraction to spatially presented masking noise conditions
Authors: Takeuchi Akira , Shim Hwan , Choi Inyong , Kim Sungyoung
Published: November 2023

This study investigates listeners' biological responses to speech under different masking noise conditions during a spatial selective attention task. Ten participants with normal hearing took part in the experiment which measured both behavioral and biological responses during a speech identification task with varying noise streams. The target speech stream was played through a front-left speaker, while masking noise, consisting of music and non-intelligent speech, was presented through either the front-right or back-center speaker. Behavioral performance was assessed based on the correct answer ratio, while biological responses were analyzed using Event-Related Potentials (ERPs) for four distinct conditions. These conditions included two Signal-to-Noise Ratios (SNRs) and two spatial positions of the masking noise stream. Using an EEG measurement headset with dry electrodes, the study collected neural responses from the participants and calculated the ERPs for each of the four conditions. After excluding one outlier, the ANOVA result revealed a difference (F(1, 3.5), p = 0.0984) in this listener group's ERP response between the two masker positions, with a lower mean peak value observed for the rear masker. This finding suggests that they were less influenced by the rear-located masking noise.
Acoustical verification of binaural features in hearing aids
Authors: Diedesch Anna C.
Published: November 2023

Hearing aid verification systems are designed to verify sound pressure levels in a patient’s ear canal, as well as ensure hearing aid features are performing to company specifications. In addition to verifying sound output, verification systems can analyze frequency specific compression, prescription targets, directional microphones, noise reduction performance, etc. However, current verification systems are unable to verify several advanced digital signal processing (DSP) features that companies have included in their hearing aids over the recent years. Particularly, binaural features which require both hearing aids to be analyzed simultaneously to measure interaural cues. Here, interaural cues were analyzed from recordings made on an acoustic manikin in an anechoic chamber across various noise reduction and directional microphone features using open-fit and occluding coupling compared to an unaided condition. During data collection, settings in the fitting software did not always align to hearing aid coupling (i.e., open versus occluding dome conditions), an error likely to occur clinically. For some noise reduction hearing aid conditions, a large interaural level difference (ILD) was introduced to one hearing aid, distorting the sound source location. ILD cues measured here, across different DSP and coupling conditions, suggest verification protocols should include additional features, such as binaural measurements.
What does bilingual experience do to music processing in the brain?
Authors: Deng Ronald , Yu Yan H.
Published: November 2023

The relationship between language and music in the brain has constantly been a topic of discussion. Understanding how and to what extent music and language experiences influence cortical dynamics for brain functions (e.g., music and speech processing) will open a crucial door to treat a myriad of neurological conditions with an auditory or language base. Bilingual experience affects nonverbal cognition, and the neural coding of pitch is shaped by language experience in the adult population. However, it is not clear whether and how different bilingual development affects music processing differently. The purpose of this study was to determine whether different early bilingual language experience modulates cortical sensitivity to music features. Participants were exposed to varying musical phrases. The electroencephalograms (EEGs)/event-related potentials (ERPs) where recorded during changes in musical features. The ERPs were compared among bilingual English-Mandarin and bilingual English-Spanish pre-adolescents and adolescents. Preliminary results suggested that bilingual teenagers with English-Mandarin backgrounds showed larger mismatch negativity responses to some, but not all six types of music changes than the bilingual teenagers with English-Spanish backgrounds. Bilingual experience modulates music process and different bilingual experiences influence music processing differently.
Articulatory working space in first and second languages
Authors: Kim Yunjung , Thompson Austin
Published: November 2023

Articulatory working space (acoustic and kinematic) is often studied to understand the overall size (limits) of a speaker’s articulatory behaviors. For example, prior research has shown that the magnitude of articulators’ movement (e.g., maximum tongue advancement, lip aperture) changes as a function of speech effort (loud, clear, and slow speech). To better understand second language acquisition in adults (i.e., articulatory working space determined by the language or anatomical differences), we compare both acoustic and kinematic working space of adult learners of English between their first language (L1) and second language (L2), which is also compared with that of native speakers of English. Specifically, the articulatory convex hull is measured during passage reading for both acoustic (F1 and F2 trajectories) and kinematic (tongue trajectories on x- and y-dimensions) data. Participants include 11 adult learners of English (four men and seven women) with a Korean language background and 10 adult L1 speakers of English (six men and four women). In the presentation, the findings will be discussed to address whether articulatory space is (1) different between native and nonnative languages (determined by linguistic needs) or (2) rather a constant articulatory characteristic of speakers between the languages, regardless of the speaker’s English proficiency.
Vocal fold contact pattern during phonation and comparison to Hertzian contact theory
Authors: Zhang Zhaoyan
Published: November 2023

The vocal folds are subject to repeated collision during phonation. The resulting contact pressure is often considered to play an important role in vocal fold injury, and has been measured in many experimental studies. In this study, vocal fold contact pattern and contact pressure during phonation were numerically investigated and compared to Hertzian contact theory. The results show that vocal fold contact in general occurs within a horizontal strip on medial surface, first appearing at the inferior medial surface and propagating upward. Because of the localized and traveling nature of vocal fold contact, sensors of a finite size may significantly underestimate the peak vocal fold contact pressure (by about as large as 80% for a sensor diameter of 2 mm), particularly for vocal folds of low transverse stiffness or thick medial surface. In general, the contact pressure increases with the size and curvature of the contact area, as predicted from Hertzian contact theory, indicating the possibility of estimating vocal fold contact pressure from medial surface kinematics. However, deviations from Hertzian theory were also observed, which need to be taken into consideration in estimating vocal fold contact pressure.
Kinematic redundancies in /u/ production with and without dysarthria
Authors: Berry Jeffrey J.
Published: November 2023

The speech sensorimotor control system exploits kinematic redundancies: various articulatory degrees of freedom can exhibit trading relationships in service of a speech goal. These redundancies affect movements within and across articulators, resulting in multiple “motor equivalent” patterns that may lead to perceptually equivalent phonemes. One classic example of this redundancy is the purported trading relationship between the tongue and lips in controlling the frequency of the second formant (F2). For the acoustic goal of achieving a relatively low F2 frequency in the vowel /u/, neurotypical talkers may exhibit trade-offs in tongue dorsum and lip position to allow motor adaptability while achieving the acoustic goal. The current work examines speech kinematic and acoustic data from many talkers with and without dysarthria during connected speech to further evaluate the proposition that tongue-lip movement trade-offs allow relatively adaptable articulatory movement in service of a relatively stable acoustic goal during /u/. It is hypothesized that idiosyncrasies in the kinematic form of tongue-lip trade-offs may reflect speaker-specific articulatory-acoustic mappings and that dysarthria may compromise the use of kinematic redundancies, particularly when articulation is severely impaired. Discussion will focus on how methodological issues, particularly kinematic and acoustic normalization, influence the findings.
Real-time, adaptive microphone subset selection for a 128-elementarray
Authors: Frazier William G. , Knutson Noah
Published: November 2023

Some advantages of high-sensor count microphone arrays include gain over random noise, increased multiple-source resolution, and potential increases in effective frequency bandwidth. But in real-time applications, computational capabilities of embedded computing hardware might limit the number data channels that can be processed without data loss. For example, given a specific processing pipeline and sampling rate, perhaps only 24 of 128 channels can be supported in real-time. Therefore, the question of which 24-element subset of the 128 possible elements should be being used to detect and track multiple moving acoustic sources. In this presentation, principles of genetic search algorithms are used to develop a method to adaptively select optimal subsets of sensors as data is being processed. Discussion concerning optimality criteria, types of genetic algorithms, and algorithm performance will be presented as well as results corresponding to a 3-D 128-element array of digital MEMS microphones when multiple types of moving acoustic sources were nearby.
Acoustic modems and models in constrained marine spaces
Authors: Porter Michael B. , Duda Timothy F. , Traykovski Peter A. , Johnson Jennifer J. , Singh Sandipa , Manganini Kevin
Published: November 2023

Constrained underwater spaces such as harbors and ports present many interesting acoustic features and challenges. Quays, wharves, seawalls, ships, floats, bridge piers, jetties, are often part of the “scenery.” Passive and active acoustic security systems are often used in these settings. Our focus is on how acoustic communications between system modems are affected by these reflectors. The evidence is mainly anecdotal, but there is a concern that the complexity of the environment can be bad for communications, just as a convention center room can have lousy acoustics. One might think that standard ocean acoustic models would allow one to study this. However, ocean acoustic modeling has been shaped by the needs of open ocean applications. There the standard process has been to assume that the sound launched along a specific azimuth, stays in that vertical plane. This is the so-called Nx2D approximation and not appropriate to model the above mentioned features that knock the sound out of its launch plane. We have been developing a 3D beam tracing model for feature-laden applications and testing it in various sites. Here we discuss the modeling and the measurements in terms of both the fundamental echo structure and the effect on acoustic modems.
Coordinated target localization and tracking with node scheduling for a network of autonomous underwater vehicles
Authors: Xu Wen , Jiang Cheng , Li Jianlong
Published: November 2023

Underwater mobile sensor networks (UMSNs) have the advantages of large area coverage, flexible network structure, and good local resolution. Particularly, spatial diversity can be exploited through node mobility to get more target information. To take those advantages in the context of regional cooperative target localization and tracking, a joint node scheduling and acoustic model fusion method is proposed for an UMSN consisting of multiple autonomous underwater vehicles (AUVs). First, a target tracking algorithm framework based on the random finite set theory is established. To take the acoustic environment into account, the a priori acoustic field and target distribution information are combined to evaluate the detection probability. A path-planning scheme with node scheduling is then developed based on the information gain to optimize the detection performance. To achieve effective node scheduling, three reward functions with different information gains respectively derived from the Rényi divergence and Fisher information matrix are proposed and compared. The rationale and the performance of the developed method are verified via simulations. A simplified version of the approach is also tested in field experiments with three AUVs, verifying the effectiveness of node scheduling in a realistic scenario.
Orthogonal time-frequency space (OTFS) modulation for underwater mobile acoustic communications
Authors: Xue Yukang , Zheng Yahong R.
Published: November 2023

This paper investigates the Orthogonal Time-Frequency Space (OTFS) Modulation for Underwater Mobile Acoustic Communications where the communication channel suffers from severe multipath and Doppler effects simultaneously. Practical OTFS modulation schemes with different parameters are designed for acoustic transmission at a center frequency of 115 kHz and a bandwidth of 11.5 kHz. The schemes are tested in lake experiments where the transmitter was anchored and the receiver was towed by a boat at a speed of approximately 1 m/s or 3.6 km/h. The receiver utilizes low-complexity channel estimation and equalization algorithms, such as NLMS (Normalized Least Mean Squares) and IPNLMS (improved proportionate NLMS) algorithms. The results show some insights of the OTFS scheme for acoustic communications. First, the mobile acoustic channel characteristics are different in the 2D delay-Doppler domain than those in the 1D time and frequency domains for the Single-Carrier Coherent Modulation (SCCM) and the Orthogonal frequency division modulation (OFDM), respectively. Second, in mobile scenarios, the OTFS scheme receiver effectively and significantly reduces the accuracy requirements of the Doppler compensation algorithm and provides better frequency-selective fading suppression and Doppler effect robustness compared to SCCM and OFDM schemes. Third, the OTFS scheme has better anti-multipath performance and reduces multipath interference by effectively differentiating signals on different paths in the 2D delay-Doppler domain.
Downlink Non-Orthogonal multiple access underwater acoustic communication receiver design Deep neural network
Authors: Zuberi Habib H. , Liu Songzuo , Bilal Mohammad
Published: November 2023

Downlink non-orthogonal multiple access (NOMA) is necessary to handle the underwater acoustic channel's limitations on bandwidth. NOMA downlink underwater acoustic (UWA) communication transmits data symbols from a source station to many users. Superimposed coding with variable power levels allows successive interference cancelation (SIC) receivers to decode data. However, SIC receivers require knowledge of channel conditions and channel state information (CSI). This is difficult to acquire, particularly in UWA communication. To address this problem, this paper proposes downlink underwater acoustic using 1D Convolution neural network (CNN). Two users with different power levels and distances from the transmitter employ BPSK modulation to support multiuser communication. Users far from the base station receive the most power. The base station uses superimposed coding. BELLHOP algorithm generates the training dataset with user depth modifications. For training the model, a composite signal passes through the samples of the UWA channel and is fed to the model along with labels. DNN receiver learns the characteristic of the UWA channel and does not depend on CSI. The testing CIR is used to evaluate the trained model. The results are compared to the traditional SIC receiver. The DNN-based DL NOMA underwater acoustic receiver outperformed the SIC receiver in simulations.
Characterization and modeling of high-speed underwater acoustic communication channels
Authors: Yan Honglu , Liu Songzuo , Ma Lu , Qiao Gang , Han Xue
Published: November 2023

High-speed mobile underwater acoustic communication (HSM-UAC) is a relatively unexplored terrain, but its importance is expected to increase with the widespread application of flexible underwater mobile platforms such as AUVs and UUVs. Knowledge of HSM-UAC channels is a prerequisite to achieving reliable communication. This work conducted two HSM-UAC channel measurement experiments with a maximum speed of 20knots. The analysis of channel time-varying characteristics demonstrates the shortcomings of traditional consistent Doppler frequency offset channel model. The experimental findings reveal the presence of two non-stationary modes in the HSM-UAC channel. For short moving distances, the changes in channel geometry result in each path experiencing a different Doppler scale, thus observing the multiscale-multilag (MSML) channels. For long moving distances, the alteration in sound ray propagation tracks entirely reshapes the channel impulse response (CIR) structure, leading to a rapid decline in channel time correlation. In further analysis, the correlation coefficient of the power delay profile (PDP) is used to measure channel stationarity. Subsequently, the stationary time and corresponding stationary distance are calculated as characterization parameters for the time-varying channel model. The quantitative evaluation results are helpful for the communication signal design and determining the update frequency of channel state information.
Three-dimensional sound recording using directional beams
Authors: Poletti Mark
Published: November 2023

The recording of three-dimensional spatial audio is typically carried out using microphone arrays that produce higher-order B-format responses, which are defined by spherical harmonics. Spherical harmonics provide a compact, orthonormal representation of the sound field, but can be intuitively challenging for general users because they do not have a direction, as opposed to typical microphones used in the audio industry. Furthermore, spherical microphone arrays typically use microphone angles based on Platonic solids which do not have intuitive directions such as front, rear, left, right, up, and down. This paper considers the use of sets of directional beams that have orthonormal properties in a similar manner to spherical harmonics. Sets of angles that have intuitive directions are used, and a method of approximating orthonormal beams is developed. The approach can also be applied to angles obtained from Platonic solids. Finally, a prototype microphone array that implements the directional recording format is described.
On the acoustics of “Argentina” and “Costanzi” Opera Houses in Rome by means of 3-D maps representation
Authors: Tronchin Lamberto , Farina Angelo , van Tonder Cobi , Bevilacqua Antonella , Yan Ruoran
Published: November 2023

The acoustics of Italian-style Opera Houses has been widely studied in the last 30 years, especially after the burning of the Teatro La Fenice in Venice in 1996. Starting from that disaster, the acoustics of these important buildings became a priority, especially for storing as much information about their acoustics as possible. The recent war events in Ukraine have underlined the importance of these acoustics for a proper reconstruction of theaters, and especially Oper Houses. This paper analyzes the acoustics of the two most important opera houses located in Rome, i.e., the “Teatro Argentina” and the “Teatro Costanzi” (also called “Teatro dell'Opera"). Both the theaters are located in the very city center of Rome and have a strong artistic activities. In both the cases, no info about their acoustics are available among the scientific community. This paper describes the most important results obtained measuring their acoustics using monoaural, binaural b-format microphones. Moreover, in order to determine 3-D acoustic maps, an EM32 Microphone from MHAcoustics was used for capturing sparial RIRs and mapping early reflection in a 360 pictures.
Enhancing acoustic contrast while tracking acoustic transfer functions: A simultaneous control approach
Authors: Lu Jing , Hu Meiling , Zou Haishan
Published: November 2023

In the context of sound zone control (SZC) systems, the ability to effectively track variations in acoustic transfer functions is essential due to the inevitable variation of acoustic environments. This paper addresses the non-uniqueness problem encountered in time domain SZC, arising from the coupling observed in multichannel transfer function modeling. To address this issue, the utilization of time-varying control filters is proposed, which offers a viable solution. In order to ensure a stable acoustic contrast performance regardless of environmental changes, as well as to mitigate extreme distortion resulting from maximal acoustic contrast (AC) in the time domain, the paper focuses on the objective of maintaining a desired level of AC while tracking the acoustic transfer functions. Simulations are carried out to evaluate the proposed method, demonstrating its capability to effectively track environmental changes and continually update the control filters, consequently providing a stable acoustic contrast performance.