bluuebunny/crossref_metadata_2025 huggingface dataset:
A coupled isogeometric finite element and boundary element method with subdivision surfaces for structural -acoustic analysis of shell structures
Authors: Liu Zhaowei , Simpson Robert , Cirak Fehmi , Majeed Musabbir
Published: June 2017

We demonstrate a method for simulating medium-wave acoustic scattering over elastic thin shell structures. We propose a coupled approach whereby the finite element formulation is used to describe the dynamic structural response of the shell and the boundary element method models the acoustic pressure within the infinite acoustic domain. The two methods are coupled through the relationship between acoustic velocities on the structural-fluid interface. In our approach, a conforming subdivision discretization is generated in Computer Aided Design (CAD) software which can be used directly for analysis in keeping with the idea of isogeometric analysis whereby a common geometry and analysis model is adopted. The subdivision discretization provides C1 surface continuity which satisfies the challenging continuity requirements of Kirchhoff-Love shell theory. The new method can significantly reduce the number of elements required per wavelength to gain same accuracy as an equivalent Lagrangian discretization, but the main benefit of our approach is the ability to handle arbitrarily complex geometries with smooth limit surfaces directly from CAD software. Our implementation make use of H-matrices to accelerate dense matrix computations and through this approach, we demonstrate the ability of our method to handle high-fidelity models with smooth surfaces for structural-acoustic analysis.
Computing head related impulse responses and transfer functions using time domain equivalent sources
Authors: Fahnline John B.
Published: June 2017

In the past, head-related impulse responses (HRIR) and head-related transfer functions (HRTF) have primarily been computed using frequency domain boundary element methods or finite-difference time domain methods. The possibility of computing HRIRs and HRTFs using transient equivalent sources is examined using a lumped parameter technique for enforcing the specified boundary condition. It is demonstrated that performing the computations in the time domain is advantageous because only a few thousand time steps are need to fully define the HRIRs and nonuniform meshes can be used to reduce the number of acoustic variables drastically without significantly degrading the solution accuracy. It is also shown that the computations adapt well to parallel processing environments and the times associated with the equivalent source calculations are proportional to the number of processors.
A new infinite element paradigm in computational structural acoustics?
Authors: Burnett David S. , Wigdor Les H.
Published: June 2017

In the 1990s, the lead author developed a radical new formulation for infinite elements for modeling scattering and radiation from structures in unbounded domains. It was shown to be faster than the popular boundary element method (BEM), for the same physics to the same accuracy, by several orders of magnitude; the speedup is unbounded as problem size increases. Academia and industry called it a “revolution” in computational acoustics that would probably bring an end to the BEM. But then Bell Labs patented and licensed the elements, effectively ending the “revolution” and removing the technology from the public domain for the next 20 years. Now, in 2017, some patents have expired and the rest will expire soon, thus restoring the technology to the public domain. The talk will review the original technology and then describe new R&D since 2015: (i) speeding up a commercial acoustic scattering code by 1400x and (ii) extending the technology by developing a new hybrid version that computes the external field over 12,000x faster than the traditional, expensive Helmholtz integral. Now that this “revolutionary” technology is back in the public domain, the market place can finally decide whether it constitutes a new paradigm in computational structural acoustics.
Acoustic radiation modes and normal modes in exterior acoustic problems
Authors: Moheit Lennart , Marburg Steffen
Published: June 2017

Acoustic radiation modes are eigenvectors of the real and symmetric acoustic impedance matrix Z, which is usually computed by the boundary element method (BEM) matrices G and H at the surfaces of inner obstacles in an unbounded fluid-filled domain. Application of the finite element method (FEM) and the infinite element method (IFEM) allows the computation of the acoustic radiation modes as well, but also normal modes can be computed as right eigenvectors of a state-space eigenvalue problem. Modal superposition of both radiation modes and normal modes leads to accurate results of the radiated sound power. However, normal modes additionally provide modal sound pressure distributions in the whole computational domain and can therefore be used to calculate frequency response functions. In this work, modal superposition and reduction in exterior acoustics are presented and discussed.
How runners deal with the shock induced vibrations propagating through their body?
Authors: Chadefaux Delphine , Berton Eric , Rao Guillaume
Published: June 2017

Runners experience numerous shocks leading to vibrations propagating from the foot toward the entire body. These repetitive shocks are related to musculoskeletal injuries. Consequently, runners tend to adapt their running patterns according to the ground surface to cushion the impact. However, the way runners manage precisely the three-dimensional components of the vibrations, especially in the frequency domain, is not well understood. The present study investigated which biomechanical parameters runners adapt to tune the shock induced vibrations according to different running conditions. A specific experimental procedure was designed, based on simultaneously collecting kinematic, dynamic, vibration, and electromyographic data during running barefoot or shod and at various velocities. Using 10 non-specialist runners, energetic and spectral analyses of the three-dimensional foot impact induced vibrations occurring at the third metatarsal bone, the tibial plateau, the knee joint, the hip joint, and the 7th cervical were performed. Results outlined the transfer function of each investigated segment. A significant outcome is the strategy set up by the neuro-musculoskeletal system to protect upper areas of the human body. This contribution opens up new perspectives in running analyses by underlining the significance of the three-dimensional and the spectral contents in the shock induced vibrations.
Hitting the ball on the meat—Finding the sweet spot of a hurley
Authors: King Eoin A. , Celmer Robert
Published: June 2017

Hurling is one of Ireland’s most popular indigenous sports. It is a Gaelic stick-and-ball sport that combines elements of field hockey, lacrosse, and handball. Key to the game is a player’s mastery of the hurling stick (hurley) which is used to the strike the ball (sliotar). The hurley is similar to a hockey stick but with a shorter, wider and more circular head (bás), which is the area of the hurley that strikes the sliotar. The sweet spot is a general term used amongst players to indicate the correct position on the hurley to strike the ball (“hitting the ball on the meat”). By measuring the moment of inertia and center of percussion of a hurley, combined with experimental modal analysis, this paper attempts to define the location of a sweet spot on a 34 inch ash huley. Measurements are based on the ASTM standard test method for measuring the moment of inertia and center of percussion of a baseball bat; however, we also propose an alternative method to this standard for determining the center of percussion of a bat.
Applying pattern recognition to formant trajectories: A useful tool for understanding African American English (AAE) dialect variation
Authors: Arjmandi Meisam K. , Dilley Laura , Ireland Zachary
Published: June 2017

Few studies have focused on the acoustic-phonetic characteristics of African American English (AAE) which distinguish this dialect from Standard American English (SAE), particularly for vowels and sonorant consonants. This study investigated whether formant dynamics from short, sonorant portions of speech are sufficient to distinguish AAE and SAE dialects. Seven female speakers, four SAE and three AAE, from the Lansing, Michigan area, were selected from a corpus of 30-45 minute sociolinguistic interviews. Target portions of speech consisting of a V or VC sequence (C = /n/, /m/, /l/, /r/) were identified from contexts selected to control for coarticulation. First (F1) and second (F2) formant values were extracted from randomly selected tokens at points 19%, 56%, and 81% of the duration through the demarcated speech portions. Pattern recognition techniques were examined to differentiate tokens of the two dialects based on formant trajectories as feature vectors. The results revealed that formant dynamics of the selected contexts are acoustically informative enough to differentiate groups of SAE from AAE speakers. A near-perfect classification of some contexts was also achieved by applying support vector machines to the formant trajectories. These findings highlight the usefulness of incorporating pattern recognition techniques for understanding acoustic variation due to dialect.
Articulatory kinematics during connected speech across dialects and dysarthria
Authors: Berry Jeffrey J. , Kim Yunjung , Schroeder James
Published: June 2017

The current work presents an analysis of articulatory kinematics during connected speech in typical talkers and talkers with dysarthria from two different dialects of American English. Instrumental methods for obtaining articulatory kinematic data during speech (particularly electromagnetic articulography) are becoming increasingly viable within the clinical setting. Yet almost no existing clinical standards for collecting and interpreting articulatory kinematic data have been established. Moreover, there is little basis for differentiating the impact of dialect from dysarthria on articulatory kinematics. We examine articulatory kinematics obtained via electromagnetic articulography during a standard connected speech passage read by typical talkers (n = 30) and talkers with dysarthria (n = 15). Participants are divided among upper Midwestern and Southern American English dialects. Analyses focus on kinematic measures of articulatory movement (range-of-motion, speed, acceleration, and jerk) within and across dialect groups and between typical talkers and individuals with dysarthria. The goal of the current work is to provide a preliminary evaluation of whether different kinematic measures of articulatory movement during connected speech may be differentially sensitive to the impact of dialect and dysarthria. The results of this work are germane to establishing clinically relevant measures of articulatory kinematics to improve the clinical assessment of dysarthria.
Dialect classification reveals mismatch between speech processing and dialect perception
Authors: Dailey Megan , Clopper Cynthia G.
Published: June 2017

Familiar dialects can facilitate speech processing. However, recent investigations of speech processing of the Northern and Midland dialects of American English reveal a different pattern: in noise-masked speech, listeners from both dialect regions identify Midland words and phrases with higher accuracy than Northern words and phrases. This preference may be explained by inconsistencies between Northern talkers’ production and perception of their own dialect. The goal of the current study was to determine whether cross-dialect processing differences between the Northern and Midland dialects reflect listeners’ explicit dialect identification ability. Participants completed a speech intelligibility in noise task followed by a forced-choice dialect categorization task. Speech stimuli in both tasks were short phrases taken from passages read by eight Northern and eight Midland talkers. Responses in both tasks were scored for accuracy. Results revealed higher accuracy in intelligibility for Midland phrases than Northern phrases, as in previous work, but poor dialect categorization performance across all listeners. The inability of listeners to explicitly categorize talkers by dialect while showing an intelligibility benefit for Midland forms indicates that the observed cross-dialect processing differences emerge even in the absence of explicit dialect categorization, revealing a perceptual mismatch between speech processing and dialect perception.
Sociophonetic variation in Mississippi: Gender, ethnicity, and prevoiced plosives
Authors: Herd Wendy
Published: June 2017

While native English speakers are traditionally reported to produce word-initial voiced plosives with short positive VOTs, recent studies suggest sociophonetic variation exists in the production of these sounds. In separate studies, more prevoicing has been reported for men than women, for African American speakers than Caucasian American speakers, and for southern American English speakers than speakers from other regions. The current study investigates the effects of gender, ethnicity, and context on voicing variation in Mississippi by analyzing word-initial /b, d, g/ as read in sentences by forty native speakers of English grouped according to self-reported gender and ethnicity. A significant effect of ethnicity and an interaction between gender and ethnicity were found. African American speakers produced voiced stops with a larger proportion of closure voicing and produced more fully voiced closures than Caucasian American speakers. While African American men and women produced similarly voiced closures, Caucasian American men voiced closures more than women. Similarly, Caucasian American speakers’ closure voicing was affected by context (e.g., following an approximant vs. following a plosive), but African American speakers consistently produced voiced closures regardless of context. These findings strongly suggest that dialectal differences play a role in the voicing variation of word-initial voiced stops.
“He maybe did” or “He may be dead”? The use of acoustic and social cues in applying perceptual learning of a new dialect
Authors: Tatman Rachael
Published: June 2017

When learning to recognize words a new dialect, listeners rely on both acoustics (Norris, McQueen, & Cutler 2003) and extra-linguistic social cues (Kraljic, Brennan, & Samuel 2008). This study investigates how listeners use both acoustic and social information after exposure to a new dialect. American English (AmE) speaking listeners were trained to correctly identify the front vowels of New Zealand English (NZE). To an AmE speaker, these are highly confusable: “head” is often heard as “hid”. Listeners were then played 500 ms vowels produced by both AmE and NZE speakers. Half of the listeners were given correct information on the speakers' dialect, and half incorrect. Listeners' classifications of vowels were affected by what they were told about the speakers' dialect. Vowels labeled as a given dialect, correctly or not, were more likely to be classified as if they were from that dialect. There was also an effect of speakers' actual dialect. Overall, the AmE-speaking listeners were more accurate when identifying vowels from AmE than those from NZE; even with the very limited acoustic information available listeners are still sensitive to inter-dialectal differences. Any model of cross-dialect perception, then, must account for listener's use of both social and acoustic cues.
Acoustic features and gender differences in clear and conversational speech produced in simulated environments
Authors: Morgan Shae D. , Ferguson Sarah H. , Hunter Eric J.
Published: June 2017

In adverse listening environments or when barriers to communication are present (such as hearing loss), talkers often modify their speech to facilitate communication. Such environments and demands for effective communication are often present in professions that require extensive use of the voice (e.g., teachers, call center workers, etc.). Women are known to suffer a higher incidence of voice disorders among those in these professions, possibly due to their accommodation strategies they employ when in adverse environments. The present study assessed gender differences in speech acoustic changes made in simulated environments (quiet, low-level noise, high-level noise, and reverberation) for two different speaking style instructions (clear and conversational). Ten talkers (five male, five female) performed three speech production tasks (a passage, a list of sentences, and a picture description) in each simulated environment. The two speaking styles were recorded in separate test sessions. Several acoustic features relevant to clear speech will be compared between simulated environments, speaking styles, and talker genders to identify differences that may begin to address the higher incidence of voice disorders among women in professions with a high vocal use.
Perceptual learning and gender normalization in fricative perception
Authors: Munson Benjamin , Logerquist Mara
Published: June 2017

Listeners identify fricatives ambiguous between /s/ and /ʃ/ differently depending on whether they believe that the talker is a woman or a man (Strand & Johnson, 1996). The current experiment examined how robust this effect is across experimental manipulations in a relatively large group (n = 99) of listeners. Listeners identified a sack-shack continuum created by pairing a fricative continuum with a VC token with gender neutral acoustic characteristics. Listeners who participated in an experiment in which the stimuli were paired with a male face did not label fricatives differently from listeners for whom the stimuli were paired with a female face. Listeners who participated in an experiment in which the stimuli were paired with both male and female faces identified fricatives differently in the two conditions, but only for participants who were flexible in their assignment of talker gender from speech stimuli alone in a second experiment. Most strikingly, fricative identification for listeners in all three experiments changed significantly over the course of the experiment, as compared to listeners in a baseline experiment in which the stimuli were not paired with a face. This suggests that the presence of a face delays listeners' perceptual learning of ambiguous sounds.
Phonetic properties of the non-modal phonation in Shanghainese register contrast
Authors: Tian Jia , Kuang Jianjing
Published: June 2017

Earlier literature documented that the tonal registers of Shanghainese were distinguished by both pitch and breathy voice, although recent studies suggested that young speakers might have lost the non-modal phonation. The goal of our present study is two-fold: (1) with the development of analytic tools, we extensively investigated the acoustic and articulatory properties of the breathy voice in Shanghainese; (2) with the ongoing sound change in mind, we recruited a large pool of speakers, born from 1930s to 2000s, so that we are able to discuss the different strategies in producing the register contrast among speakers. Simultaneous audio and EGG signals were collected. Statistical models suggested that speakers born after 1980 use different strategies in producing the register contrast. For older speakers, overall, the lower register has a breathy phonation type: periodicity and noise measures (HNR and CPP) are the most important acoustic correlates, followed by spectral tilts (H1-A1, H1-A2, and H1-A3); articulatorily, Contact Quotient and Speed Quotient reliably distinguish the two registers. For younger speakers, spectral measures are generally no longer a reliable cue, but noise measures still differ significantly between the two registers, since some younger speakers produce the lower register tones with creaky voice.
The nearfield acoustic holography using a concentric rigid microphone array to identify cylindrical sources disturbed by environment noise
Authors: Jiang Weikang , Xiang Shang
Published: June 2017

Many nearfield acoustic holography (NAH) techniques were proposed to identify acoustical noise sources, which were usually available in the free sound field. To extend NAH to more applications in noisy environments such as factories, an NAH based on a rigid rectangle microphone array was proposed to reconstruct normal velocities of vibrating plates in the environment with disturbing noise sources recently. In this presentation, an NAH using rigid microphone array with the profiles matching the vibrating surface is proposed to identify the cylindrical surface sources. The microphone array is flush-mounted at the bottom plane, and two concentric side surfaces and two rectangular side surfaces are designed to satisfy the Neumann's boundary condition. The transfer matrix between the normal velocities of sources and measured pressure is expressed as the summation of acoustic modes of the volume within the rigid array. The disturbing waves from environments can be decomposed by the inverse patch transfer functions method. The normal velocities of a cylindrical surface source are correctly reconstructed in the numerical simulation as well as a motor operating in a workshop. The results indicate that the proposed procedure is valid for reconstructing velocities of the cylindrical sound source in a noisy environment.
Deconvolving the conventional beamformed outputs
Authors: Yang Tsih C.
Published: June 2017

Horizontal line arrays are often used to detect/separate a weak signal and estimate its direction of arrival among many loud interfering sources and ambient noise. Conventional beamforming (CBF) is robust but suffers from fat beams and high level sidelobes. High resolution beamforming such as minimum-variance distortionless-response (MVDR) yields narrow beam widths and low sidelobe levels but is sensitive to signal mismatch and requires many snapshots of data. This paper applies deconvolution algorithm used in image de-blurring to the conventional beam power of a uniform line array (spaced at half-wavelength) to avoid the instability problems of common deconvolution methods. The deconvolved beam power yields narrow beams, and low sidelobe levels similar to, or better than MVDR and at the same time retains the robustness of CBF. It yields a higher output signal-to-noise ratio than MVDR for isotropic noise. Performance is evaluated with simulated and real data. Deconvolution is also applied to a circular array to compare with that obtained using superdirective beamforming (SDB) at small ka, where a is the radius and k is the wavenumber. The deconvolved beam output achieves similar performance as the SDF, and offers the same robustness as CBF at small ka.
Separating silent sources
Authors: Goldhor Richard , Gilbert Keith , MacAuslan Joel
Published: June 2017

In the presence of multiple continuously active acoustic sources, microphone response signals are additive mixtures of acoustic images: that is, filtered, delayed, and scaled versions of the source signals. By adaptively minimizing a contrast function such as the mutual information between outputs, Blind Source Separation (BSS) algorithms can “demix” sets of such microphone responses into output signals, each one of which is coherent with a single hidden source signal. The time required to converge on such a “separation solution” is typically at least several tens of seconds. However, important real-world acoustic signals of interest (such as speech) are intermittent, not continuous. As a result, the response mixtures resulting from such signals include “mute episodes” during which one or more of the sources are silent. These episodes can be employed to advantageously partition the adaptive separation process, and even to generate permanently valid separation solutions for the momentarily silent sources. We present a method of rapid optimal isolation (ROI) of intermittent sources, and explore the utility of enhancing an adaptive BSS algorithm using this new method.
On the influence of continuous subject rotation during HRTF measurements
Authors: Richter Jan-Gerrit , Fels Janina
Published: June 2017

In recent years, the measurement time of individual Head-Related Transfer Function (HRTF) measurements has been reduced by the use of loudspeaker arrays. The time reduction is achieved by some kind of parallelization of measurement signals. One such fast system was developed at the Institute of Technical Acoustics, RWTH Aachen University and is evaluated in this paper. When measuring HRTFs, the subject is usually rotated by some angle, and stops and waits for the measurement signal to complete before moving to the next measurement angle. It was shown that with this static approach a comparable results to a traditional measurement using a single speaker could be achieved. To further reduce the measurement time, a slow continuous subject rotation can be used instead. While this rotation will violate LTI (linear, time-invariant) requirements of the commonly used signal processing, the influence is assumed to be negligible. As the subject is rotating during the measurement sweep, different azimuth angles are measured per frequency. This frequency dependent offset in the measurement positions has to be corrected during the post processing. To this end, a spherical harmonic decomposition and reconstruction is applied as an interpolation method. To quantify the influence of the rotation and the subsequent post processing, a subjective and objective comparison between statically and continuously measured objects is shown in this paper.
Power delay profiles and temporal vs. spatial degrees of freedom: Predicting and optimizing performance in underwater acoustic communications systems
Authors: Preisig James
Published: June 2017

The performance of underwater acoustic communications systems is often characterized as a function of source to receiver range or the received in-band SNR. However, these measures are incomplete with respect to predicting performance. There is ample field data for which performance improved with increasing range or for which data sets with similar SNRs exhibited markedly different performance. Using field and simulation data, the Primary to Intersymbol Interference (ISI) and Noise Ratio (PINR) is shown to be a more complete predictor of performance than SNR. The results also demonstrate the significant adverse effect of ISI on system performance. With this in mind, the optimal configuration in terms of mitigating the impact of ISI is considered for the receive array and processor structure of equalizers. The role of total degrees of freedom (DOFs), equalizer filter adaptation averaging interval, and the relative stability of the channel's spatial and temporal structures is evaluated. The performance gains attainable with array spatial aperture vs the gains attainable with filter temporal depth is analyzed using closed form expressions as well as simulation and field data.
Acoustic Communications: Through soils, sands, water, and tissue
Authors: Singer Andrew , Yang Sijung , Oelze Michael
Published: June 2017

This talk will cover several experimental results and the challenges that arise in applying the basic concept of acoustic communications to vastly different media. Specifically, results obtained from state-of-the-art ultrasonic communications over distances of 100 m at data rates in excess of 1Mbps will be described, along with the unique challenges that such high data rates impose of any subsea applications, for which traditional Doppler compensation techniques are wholly inadequate. A sample-by-sample time-scale projection method was developed for streaming video applications for subsea operations. Over much shorter distance scales, even more acoustic bandwidth is available, and data races in excess of 300 MB/s have been achieved. Difficulties in such applications arise due to the nonlinearities excited in both the medium and potentially the transducers owing to the extremely wide bandwidths of operation. For potential biomedical applications, acoustic communication methods have been successfully used to transmit data in excess of 30 Mb/s through animal tissue. Challenges in such applications include no only nonlinearity and Doppler compensation, but also reverberation and scattering off bone and other materials. A final application that will be discussed is the use of acoustic communications methods for through-soils and through-sands experiments with application to various geotechnical projects. Attenuation, coupling, and excitation pose substantive challenges for such applications.
Estimating the acoustic channel to improve underwater communication
Authors: Blair Ballard J.
Published: June 2017

With the increasing number of unmanned vehicles and sensors being deployed to support scientific research, oil and gas exploration, and military operations, research and development in underwater acoustic communication is vital. The costs for testing new ideas for underwater acoustic communication is often prohibitively high due to expensive ship time, personnel, and equipment. It is also challenging to accurately model the physics and dynamics of the underwater communication channel because it is spread in both Doppler and delay. The goal of the presented research is to create a library of channels extracted from available experimental data sets. The extracted channels are compared with physical models to see which characteristics, if any, are predictable from the observed data sets. This library of channels would be used both for comparison of existing underwater communication systems and to aid in the development of communication techniques. [Work supported by the MITRE Innovation Program.]
Combining sparse recovery approaches with underwater acoustic channel models for robust communications in the shallow water paradigm
Authors: Sen Gupta Ananya
Published: June 2017

Shallow water acoustic communication techniques are fundamentally challenged by the rapidly fluctuating multipath effects due to oceanic phenomena such as surface wave focusing, specular reflections from the moving sea surface, Doppler effects due to fluid motion as well as sediment-dependent absorption from the sea bottom. Several signal processing techniques have been recently proposed that specialize in recovering the shallow water acoustic channel components using compressing sampling and mixed norm optimization theory. However, these novel techniques are typically agnostic of underlying underwater acoustic propagation phenomena. Furthermore, state-of-the-art in shallow water channel estimation typically does not account for the three-way uncertainty principles governing the localization of sparsity, time, and frequency for the time-varying shallow water acoustic channel. This talk will focus on recent techniques proposed in this domain, their relative benefits and shortcomings; as well as offer new insights into how we can combine knowledge of basic underwater acoustic propagation physics with state-of-the-art in sparse sensing and related techniques to achieve robust shallow water channel estimation. The talk will also provide an overview of equalization techniques that can be harnessed with channel estimation techniques proposed.
A multiple-input multiple-output orthogonal frequency division multiplexing underwater communication system using vector transducers
Authors: Wang Yuewen , Zhang Erjian , Abdi Ali
Published: June 2017

Vector sensors and transducers are compact multichannel devices that can be used for underwater communication via acoustic particle velocity channels (A. Abdi and H. Guo, “A new compact multichannel receiver for underwater wireless communication networks,” IEEE Transactions on Wireless Communications, vol. 8, pp. 3326-3329, 2009). In this paper, a multiple-input multiple-output (MIMO) underwater acoustic communication system is presented using orthogonal frequency division multiplexing (OFDM) modulation. Upon transmitting multiple independent data streams simultaneously over several channels, this MIMO system can increase the transmission rate, whereas the OFDM modulation mitigates the highly frequency selective underwater channels. Various components of the system including vector transducers and algorithms for synchronization, channel estimation, MIMO detection, channel coding, etc., are designed and implemented. Using this system, experiments are conducted to measure and study acoustic particle velocity channels in the MIMO setup. Additionally, system performance parameters such as bit error rate and spectral efficiency are measured and discussed for various conditions and configurations, to understand the performance of the developed vector MIMO-OFDM system. [The work was supported in part by the National Science Foundation (NSF), Grant IIP-1500123.]
Cyclic-feature based Doppler scale estimation for orthogonal frequency-division multiplexing (OFDM) signals over doubly selective underwater acoustic channels
Authors: Zhang Bingbing , Wang Yiyin , Wang Hongyi , Zheng Liming , Zhuang Zhaowen , Xu Kele
Published: June 2017

Underwater acoustic (UWA) communications enable underwater wireless networks to be applied in various applications, such as oceanographic research, pollution early-warning, disaster prevention, and military systems. However, a major challenge in UWA communications is to combat the Doppler distortion caused by doubly selective (time and frequency selective) channels. Most Doppler scale estimators rely on training data or specially designed packet structures. These methods have fundamental limitations in transmission rate and spectral efficiency. Different from these methods, this paper presents a Doppler scale estimation approach exploiting the redundant information contained within the cyclic prefix (CP) or cyclic suffix (CS) of orthogonal frequency-division multiplexing (OFDM) signals. We analyze the cyclic features of OFDM signals over doubly selective underwater channels in order to demonstrate the relationship between the cyclic features and the Doppler scale. Based on the theoretical analyses, we find that the Doppler scale can be estimated from the extremums of the cyclic autocorrelation function (CAF) of the received signal. Simulations validate our theoretical analyzes and the performance of the proposed Doppler scale estimator. Apart from the high estimation performance, we also highlight the utility of our method when only few OFDM blocks are available.
Over-sampling improvement for acoustic triangulation using Barker code audio signals
Authors: Boucheron Romuald
Published: June 2017

Acoustic triangulation applications are widely used in submarine domain to estimate 3D-coordinates of a free model or buoy localization, for example. Accuracy of positioning is dominated by the “perfect” knowledge of sound celerity in the domain and also by temporal synchronization. This short communication deals with the use of Barker audio code signals in order to improve temporal detection. Indeed, properties of such signals allow performing correlation with over-sampling detection like spatial sub-pixel algorithms in image processing domain. The first part of the communication presents this kind of signal and theoretical expected performances with very low signal to noise ratio. The second part is dedicated to the results obtained with an academic experiment devoted to measure directivity pattern of a sub-marine acoustic source. In order to decrease uncertainty of this measurement, over-sampling algorithm has been used successfully to estimate the position of the source during its displacement. This experimental set up allows in return a good angular resolution and an accurate estimate of the directivity pattern.
Geoacoustic inversion using distributed sensors
Authors: Zhang Jingcheng , Yang Tsih C.
Published: June 2017

Geoacoustic inversion is a method using sound to invert for the sediment/bottom properties. The method has been researched extensively using sound received on a vertical or horizontal line array (VLA/HLA). It requires usually a dedicated ship to deploy the array and source to collect the data. Conducting such experiments are costly and its’ use is limited in practical applications. It is anticipated in the future that many low-cost acoustic transceivers will be deployed in the ocean in the form of a distributed sensor network. It seems logical to use these distributed sensors as the receivers to conduct geoacoustic inversion using a ship-mounted source, thus covering a wide area wherever the distributed nodes are deployed. The question is how well can one estimate the bottom parameters using distributed sensors, compared with that using the traditional arrays. Note that the VLA/HLA data offers spatial coherence, thus one usually carries out geoacoustic inversion using the so called matched-field inversion. The signal received on the distributed data may not be coherent and one will conduct geoacoustic inversion using frequency-coherent inversion methods. The advantage of distributed receiver is spatial diversity, particularly when the source moves. Simulation results will be presented to compare their performance.
Directional received medium access control protocol for underwater sensor networks
Authors: Zheng Maochun
Published: June 2017

Aiming at handshake protocols in the presence of hidden terminal and exposed terminal problems in underwater sensor networks of single-channel, this paper has presented the communication protocol of directional received slotted floor acquisition multiple access based on single vector sensor orientation. This protocol uses the characteristics of single vector sensor to directional receive, solves the problem of the hidden and exposed terminal problem in the single channel condition through the directional handshake mechanism. For the problem of near-far effect, each node maintains the neighbor node information table in one-hop range, calculates the power of transmitting RTS and CTS packets according to the current neighbor information, and then calculates the transmission power that satisfies the signal-to-noise ratio requirement of the destination node according to the obtained information. Compared with the single channel protocol based on omnidirectional antenna, this protocol can make full use of the resources of idle channel and the degree of network spatial reuse. Simulation experiments show that under the two different business, compared with the classical Slotted FAMA protocol and MACAW, the throughput of network has improved by 40%~60% and 30%~40%, respectively, which proves that the directional protocol can improve the network throughput effectively.
Source localization in shallow waters with horizontal and vertical arrays
Authors: Jia Yuqing , Guo Shengming , Su Lin
Published: June 2017

In order to develop the ability of localization of the underwater target in shallow sea, a method of localization with multi-array is proposed in this paper. This method employ matched field processing (MFP) combined with vertical line arrays (VLA) and horizontal line array (HLA) to sample the sound field in large scale to get the 3-D information of the target. The multi-array is built by take a full account the affect of waveguide in shallow sea comparing with single array, which can overcome the twin-line arrays port/starbord blur problem and achieve the estimation of range and azimuth effectively. The localization is realized by a MFP and the results from simulations and data processing is more reliable in complex shallow sea environment than with single array MFP.
Depth dependence of pile driving noise measured at the research platform FINO3
Authors: Gerdes Frank
Published: June 2017

Impact pile driving is a source of high-amplitude underwater sound that can propagate large distances and may have a negative impact on marine fauna. In Germany, wind farm operators are required to monitor underwater sound levels and to ensure that the levels do not exceed certain values. Mainly because of practical reasons sound measurements are usually performed with hydrophones about 2 to 3 m above the sea-floor. It is of some interest to know whether these sound values are representative for the entire water column. We investigated this with an underwater measurement system that was designed to provide simultaneous sound measurements at up to eight different heights above the sea-floor. It consisted of a bottom mounted tripod to which a vertical chain of five to six hydrophones was attached with the top-most hydrophone being located about four meters below the sea-surface. The system was cable-connected to the research platform FINO3. This paper analyses the depth dependence of pile driving noise that was emitted from piling operations in the nearby offshore wind-farm DanTysk with propagating distances between pile and FINO3 ranging between 3km and 15km. Observed sound exposure levels (SEL) were usually somewhat larger at 2 m above the sea-floor than at hydrophones higher in the water column. Details about the variability and frequency dependence of the observed differences are presented.
Measurements of particle motion near the seafloor during construction and operation of the Block Island Wind Farm
Authors: Potty Gopu R. , Tazawa Makio , Giard Jennifer , Miller James H , Lin Ying-Tsong , Newhall Arthur , Vigness-Raposa Kathleen J.
Published: June 2017

Noise radiation and particle motion from pile driving activities were monitored using multiple sensors during the construction of the first offshore wind farm off Block Island, RI, USA in 2016. The Block Island Wind Farm (BIWF) consists of five turbines in water depths of approximately 30 m. The substructure for these turbines consists of jacket type construction with piles driven to the bottom to pin the structure to the seabed. Pile driving operations generate intense sound, impulsive in nature, which radiates into the surrounding air, water and sediment producing particle motion that may affect marine animals. The particle velocity sensor package consists of a three-axis geophone on the seabed and a tetrahedral array of four low sensitivity hydrophones at 1 m from the bottom. The acoustic pressure acquired by the hydrophones will be processed to calculate particle motion. Data from the BIWF site will be compared with model predictions and published data from other locations. Recent measurements from the same wind farm location during the operational phase also will be presented. [Work supported by Bureau of Ocean Energy Management (BOEM).]
A preliminary numerical model of three-dimensional underwater sound propagation in the Block Island Wind Farm area
Authors: Lin Ying-Tsong , Newhall Arthur , Potty Gopu R. , Miller James H
Published: June 2017

The Block Island Wind Farm, consisting of five 6-MW turbines, is the first U.S. commercial offshore wind farm harvesting wind energy to generate electricity, located 3.8 miles southeast of Block Island, Rhode Island. In-situ underwater and airborne noise measurements were made during the construction and the first two months of the operational period for the purpose of environmental impact assessment. To better interpret the noise measurements and extend the noise propagation prediction beyond the coverage of listening stations, a three-dimensional underwater sound propagation model is created with a high resolution bathymetric map and a data-assimilated ocean dynamic model. The bathymetric map is made using the 3 arc-second U.S. Coastal Relief Model (CRM) with a 100-m horizontal resolution provided by the National Centers for Environmental Information (NCEI). The ocean model is extracted from the Regional Ocean Modeling System (ROMS) ESPreSSO (Experimental System for Predicting Shelf and Slope Optics) model covering the Mid-Atlantic Bight with a 5 km horizontal resolution and 36 terrain-following vertical levels. Temporal and spatial variability of noise propagation conditions is identified in the integrated acoustic and oceanographic model. Future model development incorporating surface wind waves and sub-bottom sediment layer structure will be discussed. [Work supported by BOEM.]
Variations in the acoustic field recorded during pile-driving construction of the Block Island Wind Farm
Authors: Vigness-Raposa Kathleen J. , Giard Jennifer , Frankel Adam S. , Miller James H , Potty Gopu R. , Lin Ying-Tsong , Newhall Arthur , Mason Tim
Published: June 2017

The Block Island Wind Farm, the first offshore wind farm in the United States, consists of five 6-MW turbines three miles southeast of Block Island, Rhode Island in water depths of approximately 30 m. The turbines include a jacket-type substructure with four piles driven at an angle of approximately 13 deg to the vertical to pin the structure to the seabed. The acoustic field was measured during pile driving of two turbines in September 2015 with an 8-element towed horizontal line array. Measurements began at a range of 1 km from the turbine on which piling was occurring and extended to a range of 8 km from the construction. The peak-to-peak received level, sound exposure level, and kurtosis from each pile strike were determined as a function of range from the pile. The ambient noise just prior to each signal was also measured to calculate signal-to-noise ratio values. Results provide insight into the transition from fast-rise-time impulsive signals at close range to slow-rise-time non-impulsive signals at longer ranges. In addition, the variability among signals at the same range is being characterized as a function of pile and hammer strike characteristics. [Work supported by Bureau of Ocean Energy Management (BOEM).]
Multidimensional scaling approach in room acoustic evaluation
Authors: Malecki Pawel , Wiciak Jerzy
Published: June 2017

In the event of natural acoustics recordings, it is essential to decide on the room and, consequently, on the reverb. The decision is very important, as the consequences are not possible to be changed in the process of mixing. Therefore, proper selection of the parameter is often essential to achieve the effect desired. The aim of the study is an attempt to answer the question what is the optimal or preferred by an average listener reverberation time and other room acoustic parameters in choral music. As part of the study, sound samples of choral music that differed only by the reverb were made. The samples were tested using the procedures of multidimensional scaling. Ambisonics, altogether with VBAP, was used in the listening test.
Enhancing home recording spaces through architectural acoustics design
Authors: Otero Sebastian
Published: June 2017

The demand for home recording spaces has increased in the past years. They vary in terms of privacy, comfort, size, audio quality, budget, type of materials, acoustic treatments, types of usage and equipment. Although it is hard to unify the concept of “home studio,” there are certain architectural acoustics criteria that should be considered in order to guarantee the use of the space for creative and technical purposes. This paper analyzes different cases to demonstrate how home recording is enhanced applying these architectural acoustic principals.
When the difference between “good” and “great” is 0.5dB—Ponderings of a “Tuning Conductor.”
Authors: Blair Christopher
Published: June 2017

For many years the author has been privileged to work closely with music directors, their orchestras, and soloists in optimizing the acoustic environment in numerous rooms employing acoustic enhancement. During the process he has learned that even very small changes in energy levels and timing of virtual reflections can make a profound difference in perceived acoustic quality, largely due to masking effects, both in the audience and on the podium. The quick adjustments that can be made utilizing such systems not only make A/B comparisons useful in assessing relative quality, but also as an educational tool informing the designer as to what specific attributes in an impulse response are helpful. This presentation contains a number of illustrative vignettes from his experience as both an acoustician and conductor, including the notion that sometimes the most effective changes to acoustical perception in a concert hall can come from changing how the orchestra plays.