bluuebunny/crossref_metadata_2025 huggingface dataset:
Designing reverberation chambers for improved low frequency response
Authors: Lilly Jerry G.
Published: October 2018

Most reverberation rooms are able to provide good diffusion and spatial uniformity in the reverberant field of the source at mid- and high-frequencies. Meeting the broadband spatial uniformity requirements for ANSI S12.51 (ISO 3741) is relatively easy to achieve in the frequency bands 100 Hz to 10,000 Hz provided that the recommended room volumes are used. However, meeting the spatial uniformity requirements becomes much more difficult when the desired frequency range includes frequency bands below 100 Hz and the room must meet the pure tone qualification criteria (as is required in AHRI 220). There are two techniques that can be used to improve the low frequency room response of a reverberation room. One is to select room dimensions that will provide near uniform spacing of the room modes in the low frequency bands. The second technique is to install low frequency absorbers to the surfaces of the reverberation room to lower the “Q” of the individual modes. This paper will re-introduce a new quantity (first introduced at Internoise 2012 in New York City) called the modal distribution factor, D2, and illustrate how it can be used to optimize the dimensions of any reverberation room within the space constraints of the proposed location. In addition, the paper will illustrate the design and acoustical performance of two very successful reverberation rooms that were conceived and constructed using these techniques.
Prosody as a window into speaker attitudes and interpersonal stance
Authors: Pell Marc D. , Vergis Nikos , Caballero Jonathan , Mauchand Maël , Jiang Xiaoming
Published: October 2018

New research is exploring ways that prosody fulfils different social-pragmatic functions in spoken language by revealing the mental or affective state of the speaker, thereby contributing to an understanding of speaker’s meaning. Prosody is often pivotal in signaling speaker attitudes or stance in the interpersonal context of the speaker-hearer; in contrast to the vocal communication of emotions, the significance of prosodic cues serving an emotive or interpersonal function is often dependent on the type of speech act being made and other contextual-situational parameters. Here, we discuss recent acoustic-perceptual studies in our lab demonstrating how prosody marks the interpersonal stance of a speaker and how this information is used by listeners to uncover social intentions that may be non-literal or covert. Focusing on how prosody operates in the communication of politeness, ironic attitudes, and sincerity, our data add to research on the acoustic-perceptual characteristics of prosody in these interpersonal contexts. Future implications for how acoustic cues underlying “prosodic attitudes” affect the interpretive process during on-line speech processing and how they influence behavioral outcomes are considered.
Adult learners’ use of lexical cues in the acquisition of L2 allophones
Authors: Barrios Shannon L. , Rodriguez Joselyn
Published: October 2018

Adult second language (L2) learners gain knowledge of L2 allophones with experience. However, it is not well understood how this knowledge is acquired. We investigated whether naïve subjects use lexical cues in the form of visual referents to acquire L2 allophones. We exposed native English speakers to one of two artificial languages where novel words containing two acoustically similar sounds, [b] and [β], occurred in an overlapping distribution. The words were paired with an image that either did or did not reinforce the contrast (i.e., [bati]—"apple" and [βati]—"penguin" (DiffImagegroup) or [bati]—"penguin" and [βati]—"penguin" (SameImagegroup)). Participants completed three tests to determine whether the exposure phase impacted their ability to perceive and lexically encode the [b]-[β] contrast in trained and untrained words. If subjects use lexical cues in the form of visual referents to infer the phonological status of [b] and [β], then participants in the DiffImagegroup, but not the SameImagegroup, should discriminate and lexically encode the distinction. Data from 40 participants suggest that lexical cues may impact participants’ lexical encoding, but not perceptual sensitivity, to the [b]-[β] contrast. We discuss our findings in relation to proposed learning mechanisms.
Production and perception of cross-linguistic categories by Polish-English heritage bilingual children
Authors: Topps Maximilian
Published: October 2018

The acquisition of phonetic contrasts relies on acoustic cues such as duration and formant frequencies, which are used to distinguish acoustic signals. For second language (L2) learners, acquiring native-like L2 contrasts can be challenging when they are similar to first language contrasts, or are distinguished by different acoustic cues. Models such as the L2 Language Perception model (L2LP; Escudero, 2005), Perceptual Assimilation Model (PAM-L2; Best & Tyler, 2007), and the Speech Learning Model (SLM; Flege, 1995) make predictions regarding the outcomes of cross-linguistically similar contrast acquisition based on phonetic perception. This study investigates the similar contrasts of English /i/-/ɪ/ and Polish /i/-/ɨ/ acquired by English-Polish heritage bilingual children. The objective is to observe the models’ predictions in the context of young bilinguals, and to compare strategies across perception and production. Data were collected from 18 children aged 5;11 to 10;7. For each contrast, a picture-naming task elicited production data, and a categorisation task, involving a phonetic continuum across a lexical minimal pair, elicited perception data. Results suggest that unique strategies are used in production for each language. However, this was not reflected in perception data, which suggests perceptual assimilation. Results are discussed within a framework of relative acoustic cue weighting.
A study on the vowel duration difference followed by phonetic contexts
Authors: Lee Sooyoung , Lee Seok-Chae
Published: October 2018

Vowel duration in English serves as a cue to the voicing distinction of the following consonant, in a way that vowels preceding voiced consonants are longer than those preceding voiceless consonants. This study will examine the differences in vowel duration in various phonetic contexts produced by 8 Korean-speaking L2 learners of English in comparison with the differences produced by 4 native English speakers from the K-SEC corpus (Korean-Spoken English corpus). Specifically, we will test for the durational evidence of the influence of different manners of articulation and syllable positions on preceding or following vowels and investigate how the influence is phonetically realized by two language groups. Our hypotheses and predictions are as follows: 1) vowels are longer when preceded by consonants in coda position than when followed by the same consonants in onset position, 2) vowels are longer when preceded by nasals than when preceded by non-nasal consonants, and 3) Korean speakers produce the differences in vowel length to a smaller extent than native English speakers.
Difference of articulatory movement between native and non-native consonant clusters
Authors: Funatsu Seiya , Fujimoto Masako
Published: October 2018

We investigated the difference of articulatory movement between native and non-native consonant clusters. English has consonant clusters, but Japanese does not. Therefore, speakers chosen for comparison were native English and native Japanese speakers. Speech samples consisted of 4 words, “blat,” “bnat,” “plat,” “pnat.” In these words, /bl/ and /pl/ are English clusters, but /bn/ and /pn/ are not. We measured the movement of the tongue tip, the mandible and the lower lip by WAVE system (NDI corp.). There were remarkable differences in the mandible and the lower lip movement between native (/bl/, /pl/) and non-native (/bn/, /pn/) clusters in English speakers. Namely, with the non-native clusters the difference of the articulatory movement in the mandible and the lower lip of every utterance was quite large; however, in native clusters, the difference was quite small. For Japanese speakers it was large for all clusters. Thus, it was revealed that the articulatory movement of the mandible and the lower lip in non-native clusters was not stable in native English speakers, even though English has consonant clusters. (This study was supported by KAKENHI 15K02524, 17K02707.)
Ultrasonic through-tissue communication with video capabilities
Authors: Tabak Gizem , Oelze Michael , Singer Andrew
Published: October 2018

The use of wireless implanted medical devices (IMDs), which communicate data wirelessly from sensors within the body to a receiver outside of the body, are gaining significant momentum in medical diagnosis and treatment procedures. Currently, radio frequency (RF) electromagnetic waves are the most frequently used communication method for wireless IMDs. However, there are various drawbacks of using RF electromagnetic waves in such applications such as high attenuation in the body and strictly regulated frequency spectrum, which consequently limit data rates of these devices. An alternative approach to RF transmission for IMDs uses ultrasonic waves, which experience lower attenuation in the body and have higher available bandwidth. In our work, we aim to demonstrate high data rate (>1.2 Mbps) through tissue communication using ultrasonic waves that will allow us to stream video with an IMD while actively controlling it inside the body. Initial experiments performed with 2mm biocompatible sonomicrometry transducers, where 2.4 Mbps data rate BER less than 5E-5 through water is achieved, demonstrating the capability of this method for high speed data transmission. Additional targets of this work include data transmission through beef liver and a live rabbit abdominal wall with high enough data rates capable of video streaming.
High-resolution ultrasound camera with sparse array for machine diagnosis
Authors: Park Choon-Su
Published: October 2018

Fault detection plays a crucial role in maintenance of mechanical systems. Ultrasound is known to be a prevalent feature of incipient fault. Industrial ultrasound detectors are widely used to find out initial malfunction of machinery. Especially, ultrasound camera can shows the fault by overlapping beamforming power distribution with optical camera image. One can easily observe where the faults are. Ultrasound transducer usually has larger size than the wavelength of its resonance frequency for highly sensitive measurement. It must be a useful approach to design sparse array that gets over the spatial aliasing problem. The aliased beamforming distribution contains high grating-lobe level due to rough spatial sampling beyond half wavenumber spacing. The sparse array is optimized to minimize maximum side-lobe level by genetic algorithm under given constraints. Being with the constraints of size and number of elements, a limitation on spatial resolution is inevitable. Functional beamforming is simple but effective to dramatically increase spatial resolution as well as reduce side-lobes. A high resolution imaging scheme based on functional beamforming with an optimal sparse array is introduced.
SoilComm: A miniaturized through-soil wireless data transmission system
Authors: Yang Sijung , Baltaji Omar , Hashash Youssef M. , Singer Andrew
Published: October 2018

Wireless underground sensor networks (WUSN), are emerging Internet-of-things (IoT) technologies, that can benefit numerous applications including geotechnical data acquisition for online infrastructural health monitoring and automated agricultural systems. State-of-the-art approaches to wireless underground sensor communications have employed, as their data carrier, electromagnetic waves, characterized by: their extreme signal path losses in soil due to absorption and scattering, the challenges they impose on transceiver miniaturization, and the power/energy efficiency constraints they suffer from during battery-based operation. Acoustic waves can resolve many of these challenges, benefiting from lower levels of absorption and scattering losses at target frequencies, and from looser size limitations, resulting in smaller form factor, power-efficient transceivers. In this work, biologically inspired, and afterwards motivated by the physical properties of acoustic propagation in soil, a scheme for wireless data communication through soil employing acoustic waves is presented. Experimental results illustrate the system capability of sending application-specific data, ranging from sensor readings to low-resolution images, over distances exceeding 30 m through soil.
Implications of Riemannian geometry in underwater acoustics
Authors: Finette Steven I.
Published: October 2018

Signal processing involving source localization and detection in underwater acoustics often depends on comparing covariance or cross-spectral density matrices (CSDMs) estimated for data, replica or noise source data from passively sensed acoustic fields incident on sensor arrays. Such comparisons often depend on a measure of similarity between matrix pairs and this typically involves a Euclidean metric. While Euclidean-based metrics are ubiquitous, convenient and useful, they are not fundamental to this task. By exploiting the facts that a CSDM is Hermitian and positive semi-definite, one can interpret such matrices as points constrained to a Riemannian, rather than a Euclidean manifold, with the implication that similarity between matrix pairs should be measured using a metric consistent with the manifold’s intrinsic structure. This geometric interpretation leads to alternative matched-field processors for source localization involving the Riemannian distance as a measure of similarity and can be used to solve this inverse problem. Some implications of this non-Euclidean approach in underwater acoustics, as well as a possible extension to source detection are discussed. [Work supported by the Office of Naval Research.]
Representing sonar target features using Braid geometry
Authors: Sen Gupta Ananya , Kirsteins Ivars P.
Published: October 2018

Acoustic color features that specify a sonar target in the time-frequency spectral domain may be broadly classified into two groups: (i) elastic wave features that depend on the material composition of the target, and (ii) wave features that depend on the specific geometry of the sonar target. Typically these two types of target features overlap intricately in the acoustic color domain that renders them difficult to separate using traditional signal processing techniques. Furthermore, field experimental conditions such as target orientation, proud, buried or semi-buried state of the target as well as sediment characteristics, influence the relative overlap and morphology of these acoustic color features. A key bottleneck to robust target feature representation and subsequent classification is how these potentially overlapping features may change in the field from what is measured under laboratory conditions or observed through model-based simulations. We will present some relevant results from our ongoing work in disentangling sonar features using a combination of geometric signal processing, graph-based informatics techniques and well-known physical models. Specifically, we will employ geometry of braid manifolds as a possible basis for representing the physical phenomena influencing each group of target features. We will also present recent results from graph-based informatics to show how braid encoding across multiple experimental conditions can be employed to determine and disentangle overlapped target features.
Harnessing geometric techniques for robust real-time estimation of shallow water acoustic channels
Authors: Sen Gupta Ananya , McCarthy Ryan
Published: October 2018

The shallow water acoustic channel is well-known to exhibit rapid fluctuations in delay spread, which poses a fundamental roadblock to real-time channel estimation, particularly under moderate to rough sea conditions. This talk will summarize recent research on multiple representations of the shallow water acoustic channel and present the relative merits of each channel representation in localizing different channel effects. In particular, we will examine the efficacy of employing geometric signal processing techniques to adaptively track the shallow water acoustic channel in each representation. Results based on experimental field data as well as channel simulations under varying environmental conditions will be presented.
Self-calibration of dual, closed surface microphone arrays
Authors: Williams Earl G. , Kuperman William A.
Published: October 2018

With the recent availability of very inexpensive, miniature digital MEMS microphones, the construction of large count microphone arrays is financially within the reach of the basic researcher. Using various mathematical techniques such as the Helmholtz-Kirchhoff integral and the equivalent source method (ESM), dual surface microphone arrays provide the ability to separate incident and scattered fields instantaneously in time. When these arrays form a closed surface, this separation is extremely accurate. However, one of the concerns with any microphone array is the calibration of the individual elements, especially near the resonance frequency of the diaphragm. In this paper, a self-calibration technique is proposed, based on ESM, that uses external loudspeakers in a simple laboratory facility not required to be anechoic. The approach is tested numerically on a dual surface, 256 element pair spherical microphone array, and uses an algorithm for the minimization of a cost function based on the extinction of the scattered field when the unknown individual calibration coefficients of the array elements are varied. Use of a Burton-Miller approach to model the ESM sources of the scattered field improves dramatically the accuracy at specific frequencies. The problematic interior Dirichlet eigenfrequencies are addressed. [Work supported by the Office of Naval Research.]
Model-based direction of arrival estimations for sound sources using a spherical microphone array
Authors: Landschoot Christopher R. , Mathews Jonathan , Braasch Jonas , Xiang Ning
Published: October 2018

In many room acoustics and noise control applications, it is often challenging to identify the directions of arrivals (DoAs) of incoming sound sources. This work seeks to solve this problem reliably by beamforming, or spatially filtering, incoming sound data with a spherical microphone array via a probabilistic method. When estimating the DoA, the signal under consideration may contain one or multiple concurrent sound sources originating from different directions. This leads to a two-tiered challenge of first identifying the correct number of sources, followed by determining the directional information of each source. To this end, a probabilistic method of model-based Bayesian analysis is leveraged. This entails generating analytic models of the experimental data, individually defined by a specific number of sound sources and their locations in physical space, and evaluating each model to fit the measured data. Through this process, the number of sources is first estimated, and then the DoA information of those sources is extracted from the model being the most concise to fit the experimental data. This paper will present the analytic models, the Bayesian formulation, and preliminary results to demonstrate the potential usefulness of this model-based Bayesian analysis for complex noise environments with potentially multiple concurrent sources.
Beamforming loudspeaker array in a multi-layered configuration
Authors: Ih Jeong-Guon , Cho Wan-Ho
Published: October 2018

A planar, broadside acoustic array with uniform loudspeaker spacing is often adopted for generating a highly directional acoustic beam, but the performance in radiated power and spatial bandwidth should always compromise with the transverse size limit. To achieve a high-power, directional sound in a compact size, a multi-layered loudspeaker array configured in both broadside and end-fire types is devised. This design concept is tested with an array of 3 layers with 0.2 m spacing. Each layer contains 7 loudspeakers in a centrally symmetric arrangement on a plane with the longest dimension of 0.6 m. In each layer, the separation distance between loudspeakers is uniformly set as 0.2 m. The optimal beamforming method is employed for calculating the control filter. The compensation for the performance degradation due to self-scattering effect is also included in the control filter, for which the additional phase delay due to scattering is estimated from the simulated response of the array system. The maximum sound level at 1 m distance from the array center is achieved as 145 dB at 1.5 kHz, and, at 30 m position, the level decrease of 26–30 dB with the directivity index of 15 dB is observed for 1–5 kHz.
An experiment of sound effects on shrimps for ecoacoustics' research
Authors: Cao Zhengliang
Published: October 2018

It is important to understand relationships of underwater soundscapes, marine ecology, and acoustic behaviors for refine research questions on ecoacoustics. Benthic crustaceans live in a local range and are normally used to an indication of environmental impacts, especially for the ecosystem in ocean. To get much information of interaction impacts on benthic crustaceans, this paper introduces an experiment of sound effects on shrimps in an aquatic pond. In the experiment, there is only a kind of benthic crustaceans, Penaeus prawns. Environmental and artificial noises are recorded by a passive sound monitor. The preliminary results are obtained through preprocessing data acquisition, analyzing effective phenomena, and comparing sound characteristics. It will be valuable to consider much simple cases for main questions. So data of natural rainfall noise, aerator noise and artificial noise may help to know about the acoustic ecology of shrimps in an aquaculture environment. Since the water quality is very poor in real shrimp pond, it cannot accurately obtain the relationship between the sound of shrimp and its behavioral characteristics. Further research considerations are also be prospected in this paper. [The project is funded by The National Natural Science Foundation of China (41374147) and (14111103900).]
A new model for underwater noise research in larval fishes: Biomedical and ecological implications
Authors: Coffin Allison , Xu Jie , Gritsenko Dmitry , Lawton Kristy , Villalpando Beija , Sisneros Joseph A. , Bhandiwad Ashwin , Uribe Phillip
Published: October 2018

In humans, excessive noise exposure from occupational or recreational sources causes permanent hearing loss. Similarly, exposure to underwater anthropogenic noise can cause hearing loss in aquatic organisms, including fish. While fish can recover from noise-induced hearing loss, underwater noise exposure can cause behavioral changes that reduce organismal fitness. In all vertebrates, acoustic trauma can cause damage to sensory hair cells. To better study the effects of noise on hair cells, we have developed a noise exposure system that uses broadband sound to damage hair cells of the inner ear and lateral line of larval zebrafish. Acoustic over-exposure kills hair cells in an intensity- and time-dependent manner, with maximum hair cell damage observed 72 hours after noise exposure. This time course is consistent with mammalian studies, where hair cell death occurs days to weeks after noise exposure. Other features of acoustic trauma are also conserved between zebrafish and mammals, including activation of apoptotic signaling cascades and changes in hair cell-afferent synapses. These studies demonstrate that larval zebrafish are a tractable new model for studies of noise-induced hair cell death. However, our acoustic trauma system could also be used in other species, allowing for new studies of underwater noise in larval fishes.
Monitoring noise levels and delphinid presence through autonomous acoustic in Rio de Janeiro coastal area
Authors: Bittencourt Lis , Sousa André , Bisi Tatiana , Lailson-Brito José , Azevedo Alexandre d.
Published: October 2018

In order to investigate the soundscape in a Rio de Janeiro coast site, passive acoustic monitoring was conducted during two non-consecutive weeks in the summer of 2015/2016 by deploying one SM2M + device. The equipment recorded at a 66% duty cycle with sample rate of 96 kHz and 36 dB gain. Third octave levels (TOLs) were calculated for all recordings through PAMGuide software. To search for delphinid presence, a band limited energy detector was employed using Raven 1.5 in a 512 Hann window, 50% overlap. TOLs varied across frequencies and day hours. Light hours were noisier than dark hours in 25 frequency bands (MW, p<0.01), with highest mean level being measured at 794 Hz at 06am (105.4 ± 5.4 dB re 1μPa), and lowest mean level measured at 39.8 kHz at 01am (80.3 ± 3.7 dB re 1μPa). A total of 281 delphinid sound emissions were detected in nine occasions, seven during week 1 and two during week 2. Seven detection events occurred during dark hours. Although more than one species of delphinid is known to occur in Rio de Janeiro coast, these results indicate a frequent nocturnal use of the area, which was only possible to observe through autonomous monitoring.
Do nearby seismic airguns reduce singing effort in humpback whales?
Authors: Noad Michael J. , Dunlop Rebecca
Published: October 2018

The high-level percussive sounds generated for seismic sea floor exploration have the potential to disrupt normal behaviors of whales. This study assesses if there is any reduction in the singing behaviour of migrating humpback whales in response to nearby airguns. Singing whales were acoustically tracked as they migrated along the coastline of south-eastern Queensland. A 20 cubic inch airgun or 140 cubic inch array of airguns was towed through the study area for 1 hour with the airguns firing every 11 sec (active treatments) or without the airguns operating (controls). Singing activity across each day was measured by counting the number of singing whales within the 10km-radius study area every 10 min, from 0700 to 1700, including during experiments. Singing activity during active periods and controls were compared with each other as well as with pseudo-randomly selected 1 hour periods when experiments were not underway (baseline). Changes in singing effort were also recorded by noting the number of instances of a singer stopping or starting during these periods. Preliminary analyses show no significant differences in singing activity, nor changes in singing effort, between active, control, or baseline periods.
Multi-channel cross-correlation used to estimate time delays
Authors: Nosal Eva-Marie
Published: October 2018

Many marine mammal localization techniques rely on estimating the time of arrival delays of a call between the hydrophones of an array. This is commonly achieved by cross-correlating (some version of) received signals between hydrophone pairs. This process is complicated in cases with noise, multiple animals, and multipath arrivals. In this talk, I explore and demonstrate the potential of a multi-channel cross-correlation technique to help establish time delay estimates. Possible extensions to signal enhancement for detection and classification purposes are also discussed. [This work was supported by ONR Award No. N00014-16-1-2598.]
Turning birds into bats—Multi-modal tracking to study collective behaviour
Authors: Koblitz Jens C. , Frokosh Oren , Nagy Mate , Carlson Nora , Naik Hemal , Iain Couzin
Published: October 2018

Acoustic localization has been used to track numerous vocalizing animals, including whales, bats birds by measuring the time of arrival differences of a sound recorded by multiple receivers. This method, however, requires the species of interest to vocalize regularly to achieve decent temporal resolution. In order to track the movements of birds that do not vocalize regularly, a miniature, radio controlled ultrasonic speaker is attached to the birds and constantly emits ultrasound chirps. The chirps from various tagged animals can be discriminated based on information encoded in the signal. A 30 microphone array covering the ceiling of a large aviary (14.8 × 6.6 × 3.9 m) is used to record the chirps and provides the basis for accurate localization of the sources. As the chirps do not overlap with the frequency of the animals vocalizations, these can be localized and assigned individually in a flock of moving birds. In addition, a VICON motion capture system provides a simple and accurate method to ground truth the acoustic localizations as it records very precise movement information of the individuals while line of sight is between the marker on the animal and a number of cameras is established. The acoustic and visual tracking systems working together provides a unique and novel answer to consistent and precise localizations of non-vocalizing individuals (and groups of individuals) as they move through space.
Increasing access to big bioacoustic data through cloud-based systems
Authors: Wall Carrie C. , Anderson Charles , Spring Allyson , Gedamke Jason
Published: October 2018

The National Oceanic and Atmospheric Administration’s (NOAA) National Centers for Environmental Information (NCEI) has recently developed an archive for the long-term stewardship of passive acoustic data. Protocols for archiving passive acoustic data are currently being established in support of the NOAA Ocean Noise Reference Station Network project, and monitoring marine mammals and fish. Archives maintain data, but access to these data is a core mission of NCEI that allows users to discover, query, and analyze the data in new and innovative ways. To facilitate global access to what will be 100s of TB of bioacoustic data, NCEI has partnered with Google Cloud through the NOAA Big Data Project. Cloud-based access to large volumes of data creates the ability to listen to sound files, download more easily, and bring processing routines to the data instead of bringing the data to the processing.
Beam coding with orthogonal complementary Golay codes for signal-to-noise ratio improvement in ultrasound mammography
Authors: Kumru Yasin , KÖymen Hayrettin
Published: October 2018

In this paper, the experimental results of using Golay coded signals at 7.5 MHz to detect breast microcalcifications of 50 μm size is presented. For improving the success on detecting the microcalcifications, orthogonal Golay complementary sequences up to 40 chips having cross correlation for minimum interference are used as coded signals and they are compared to tone burst pulse of equal energy in terms of resolution under weak signal conditions. The measurements are conducted using an experimental ultrasound research scanner, DiPhAS having 256 channels, a phased array transducer with 7.5 MHz center frequency, and the results obtained through experiments are validated by Field-II simulation software. In addition, to investigate the superiority of coded signals in terms of resolution, multipurpose tissue equivalent phantom (Model 84-317) containing series of monofilament nylon targets, 240 μm in diameter, and cyst-like objects with attenuation of 0.5 dB/(MHz*cm) is used in the experiments. We obtained ultrasound images of monofilament nylon targets and cyst-like objects for the evaluation of resolution. Experimental results show that it is possible to differentiate closely positioned small targets with increased success by using coded excitation in very weak signal conditions.
Influence of surfactant encapsulation on the acoustic droplet vaporization
Authors: Lacour Thomas , Valier-Brasier Tony , Coulouvrat François
Published: October 2018

Nanodroplets have great, promising medical applications such as contrast imaging, embolotherapy or targeted drug delivery. Their functions can be mechanically activated by means of focused ultrasound inducing a phase change of the inner liquid known as acoustic droplet vaporization (ADV) process. In this context, a four-phases (vapor + liquid + surfactant layer + surrounding environment) model of ADV is proposed. Attention is especially devoted to the nonlinear mechanical properties of the surfactant layer encapsulating the metastable bubble/droplet system and described as a model of nonlinear surface tension with two parameters. Various responses to ultrasound excitation are illustrated, depending on surfactant properties and acoustical excitation parameters. Different classes of ADV outcomes are exhibited, and a relevant threshold ensuring complete vaporization is defined. The dependence of this threshold with acoustical, geometrical, and mechanical parameters is also discussed.
Perceptual thresholds of violin soundpost length variation
Authors: Fu Lei , Fritz Claudia , Scavone Gary
Published: October 2018

The soundpost of a violin is an essential component of the instrument, providing structural support and a mechanical coupling between top and back plates. Subtle changes made by a luthier to the soundpost dimensions or position can result in significant variations in sound and playing qualities. A study was designed to investigate the perception of sound quality associated with changes to the soundpost length, making use of an adjustable carbon fiber soundpost. Violin excerpts were recorded with different soundpost lengths. Violinists and luthiers were invited to listen to the recordings and provide perceptual feedback using a computer interface. Tasks included pairwise comparisons and free sorting. Perceptual thresholds in detecting soundpost length changes were characterized, as well as possible groupings according to soundpost lengths that might be considered too short, too long and optimal. Bridge admittances were also measured and analyzed for different soundpost lengths.
Studying the clarinet with an artificial mouth: Comparison of playing frequencies between model and measurement
Authors: Gabriel Jack D. , Coyle Whitney L.
Published: October 2018

In the field of musical acoustics, instruments, such as the clarinet, are often played with the use of an artificial mouth or playing machine in order to objectively measure the playing characteristics such as sound levels, playing frequency, regime changes, etc. The purpose of this research was to study the tuning tendencies of the clarinet experimentally and compare this to the models found in literature. A clarinet was artificially blown to determine the playing frequencies for varying levels of blowing pressure and reed opening. These measurements could then be compared to computational models of the clarinet. The experimental results will be presented and compared with the theoretical values from past studies that predicted playing frequencies both analytically and computationally. Finally, suggestions for improving the models will be presented and discussed.
Construction and validation of a high resolution finite element model of the Japanese koto
Authors: Coaldrake Angela K.
Published: October 2018

The Japanese koto is recognized for its complex resonances, but few studies have been conducted to understand its acoustic properties. This paper presents a COMSOL Multiphysics finite element model of a hand-crafted, professional-grade Japanese koto. To do this, the complex internal and external geometry was captured on 2400 CAT scan cross-sections and imported into Comsol. The model was validated by reference to literature data, Chladni patterns, frequency response experiments, acoustic camera and laser scanning vibrometer studies. These results were then compared to the sound as played on the actual instrument. Challenges in the model’s development arising from the materials and construction of the instrument are discussed, most notably the anisotropic nature of the paulownia wood including multiple grain orientations used in its construction and the added complexity of modeling the organic shape of the real, hand-crafted 1.83 m long instrument with its major internal variations. These developments were guided by parallel studies in less intractable geometry such as simpler box models or models with idealized geometrical shapes lofted along a spline. The CAT scan derived model is used as a quasi-experimental tool to investigate the effect of internal components of the koto on its resonances and results presented.
Human-technology interfaces with the tactile metronome
Authors: Hesselink Nathan
Published: October 2018

The story of the metronome is the story of humankind coming to terms with evolving conceptions of time and coordination as mediated through technologies of the modern age. What began as a tool for regulating and documenting tempo soon became the temporal yardstick by which aspiring musicians strove to emulate in practice and performance. In the early twentieth century the metronome took on the new role of synchronizing live musicians with moving images on a screen (the so-called “click track”), and as the century progressed the metronome would come to dictate the manner in which musicians related to each other in the recording studio and in live events. This paper focuses on the latest manifestation of this phenomenon, the tactile metronome, looking at how vibrotactile stimulation is being used for temporal synchronization as well as an enhanced sense of embodiment for the performer, including haptic feedback. Four modern products tailored to performing musicians will be introduced and analyzed in the overlapping contexts of synchronization and embodiment. It is my argument that the metronome has now come full circle, returning a sense of human feel to the experience of making music.
From behavior to physiology and back again: The role of auditory cortex in vocal production and control
Authors: Eliades Steven , Tsunada Joji
Published: October 2018

Vocal communication plays an important role in the lives of both humans and many animal species. Ensuring accurate communication, however, requires auditory self-monitoring to control vocal production and rapidly compensate for errors in vocal output. Despite the importance of this process, the underlying neural mechanisms are relatively unknown. Previous work has demonstrated that neurons in the auditory cortex are suppressed during vocal production, while simultaneously maintaining their sensitivity to vocal feedback, suggesting a role in auditory self-monitoring. The behavioral role of auditory cortex in vocal control, however, remains unclear. We investigated the function of auditory cortical activity during vocal self-monitoring and feedback-dependant vocal control in marmoset monkeys. Using real-time frequency-shifted feedback during vocalization, we demonstrate that marmosets exhibit rapid compensatory changes in vocal production, a feedback-dependent behavior that is predicted by the activities of neurons in auditory cortex. We further establish the role of auditory cortex in vocal control using electrical microstimulation to evoke rapid changes in produced vocalizations. These findings suggest a causal role for the auditory cortex in vocal self-monitoring and feedback-dependent vocal control, linking mechanisms of production and perception, and have important implications for understanding human speech motor control.
How much formant variability in linear prediction-based measurements can be explained by F0 variability?
Authors: Chen Wei-Rong , Whalen D. H. , Shadle Christine H.
Published: October 2018

Previous studies have used speech variability as a measure of speech development; for instance, children reduce the variability in their formant frequencies as they grow, indicating increases in speech motor control. However, formant measurements in most of these studies are computed using variants of linear prediction coding (LPC), which is known to exhibit a bias towards the nearest harmonic of F0, especially at high F0 [Shadle et al. (2016). JASA. 139(2), 713–727]. The proportion of the reported formant variabilities that can be attributed to changes in F0 is still unknown; thus, the true variability in formant frequency is also unknown. Therefore, we used published values of formant and F0 variabilities for /a/ produced by children (age range: 4~19 + yrs), and, for each age group, we synthesized 1000 vowels with the same formants (F1~F3) fixed at the reported group means, but randomly varied F0 using a normal distribution (μ = F0 group mean, σ = F0 standard deviation (SD)). We then measured formants in those synthesized vowels using LPC; the SDs of the measured formants were considered to be the F0-biased formant variability (true formant SD = 0). A ratio of F0-biased SD to reported formant SD for each age group indicates how much of the reported formant variability can be explained by F0 changing in children’s speech. The results show, on average, that the purported formant variabilities in previous studies that can be explained by F0 bias are 82.2 % for F1, 31.1% for F2, and 17.4% for F3.
Underwater acoustic localization of sperm whales with a pair of hydrophones
Authors: Skarsoulis Emmanuel , Piperakis George , Kalogerakis Michael , Orfanakis Emmanuel , Papadakis Panagiotis , Dosso Stan E. , Frantzis Alexandros
Published: October 2018

An iterative Bayesian approach using a pair of hydrophones was applied for the localization of vocalizing (click producing) sperm whales off the southern coast of Crete in the Eastern Mediterranean Sea in June 2015. The localization method relies on the time difference between direct and surface reflected arrivals at each hydrophone as well as between the direct arrivals at the two hydrophones, and on the knowledge of the hydrophones depths, whereas it accounts for refraction due to a depth-dependent sound-speed profile. The method provides range and depth estimates as well as estimates for the localization uncertainty reflecting measurement errors and environmental uncertainties. Further, by assuming a simple model for the array geometry bearing estimates were obtained. The inherent left-right ambiguity in bearing estimation can be resolved by changing the array orientation through maneuvering. The localization results were verified through encounters with the sperm whales when they ascended to the surface. [Work supported by EU-Greece through the Aristeia-II program and the NSRF 2014-2020/PERAN project.]
Sound quality metrics applied to road noise evaluation
Authors: Marciniuk Karolina , Kostek Bozena
Published: October 2018

Road noise monitoring systems typically measure sound levels in specific time periods. The more insightful approach suggests to measure also the nature of noise. Sound quality of sounds such as car noise can be objectively evaluated by several parameters. One of them is psychoacoustic annoyance, described by loudness, tone color, and the temporal structure of sound. In this paper the assessment of several sound quality parameters, such as: loudness of time-varying noise, sharpness, fluctuation strength, and roughness, as well some additional parameters borrowed from the Music Information Retrieval area are presented. Then a comparison between parameter values obtained by means of the professional measurement system and a tool working in the Matlab environment is performed. The conducted investigations are carried out using recorded samples of an individual vehicle pass-by in close proximity of the road, organized as a database of road traffic noise recordings. [This research was partly supported by the Polish National Centre for Research and Development within the Grant No. OT4-4B/AGH-PG-WSTKT.]
Extraction of shaft frequency based on the DEMON line spectrum
Authors: Song Mingyu , Li JiangQiao , Hui Juan
Published: October 2018

Target-identification is an important technique in modern sea war. As the core device to identify targets, negative sonar analyzes noise underwater and abstracts their properties and determines its source. Shaft frequency which barely lies in the properties of the propeller differs among different kinds of ships, so this property can be regarded as a key to identify targets. This paper studied the theory and math model of the noise produced by ships. And Detection of Envelope Modulation on Noise (DEMON) is an effective technique to abstract low-frequency properties from high-frequency signals via demodulation. There are two methods to demodulate the noise, namely absolute low pass demodulation and square low pass demodulation. To improve the performance of DEMON spectrum estimation, adaptive line spectrum enhancement technology is introduced. An adaptive threshold is then set to eliminate the continuous spectrum. Then the largest common divisor algorithm is presented to calculate to shaft frequency out of the array consist of the centre frequency of these line spectrum. Simulation is conducted as described above. Two methods of demodulation is compared in this part. Brief sea experiment data processing shows that this algorithm is able to abstract the shaft frequency among the ship noise.
Estimating the velocity of a moving acoustic source based on chirplet transform
Authors: Liang Ningning , Yang Yixin , Guo Xijing , Zhang Boxuan
Published: October 2018

The instantaneous frequency (IF) of a pure tone emitted by a transiting acoustical source varies due to the acoustical Doppler effect, which is usually used to estimate the velocity of the source. Hence, the common practice is first to estimate the IFs by the time-frequency analysis (TFA) of the signals and then to derive the velocity from the IFs. In this paper, a TFA method based on the chirplet transform is proposed where the IFs of the chirplets are formulated by the nonlinear kinetic function with respect to the source velocity. By iteratively fitting the chirplets to the signals it directly renders the velocity of the source. The efficiency of the proposed method is validated by the microphone recorded noise data due to a flying helicopter during an experiment carried out in May, 2018, northeast to the coast of Sanya, Hainan province, China.
Statistical inference approach applied to simultaneous vertical particle velocity and acoustic pressure measurements from SUS explosive sources made in the New England Mudpatch
Authors: Knobles David P. , Dahl Peter H. , Dall'Osto David R. , Wilson Preston S.
Published: October 2018

Pressure and particle velocity time series were measured in the New England “Mudpatch” in the Spring of 2017. The time series were generated from Signal Underwater Sound (SUS) MK-64 explosive sources and recorded by a bottom-mounted L-shaped array of hydrophones and a vector sensor system that recorded both pressure and particle velocity (1.3 m off the bottom) in a separate location. A maximum entropy approach was utilized to infer the marginal probability distributions of geo-acoustic parameter values of the seabed using both the particle velocity and the pressure. The seabed consists of a surface layer of mud over sand whose thicknesses are range- and azimuth dependent. The modeled vertical component of particle velocity is computed by a three-point derivative of the pressure fields generated by RAM-PE along the vertical axes centered at the depth of the vector sensor. The two-way travel time CHIRP data for the multiple range and azimuth-dependent sediment layers provide a constraint for the sediment thickness for each sound speed hypothesis of the model parameterization space. Results of the statistical inference of the geo-acoustic structure derived from the particle velocity measurements are compared to those derived from pressure field measurements. [Work supported by ONR.]