bluuebunny/crossref_metadata_2025 huggingface dataset:
Measuring the low-frequency response of an acoustic guitar
Authors: Shepherd Micah R.
Published: May 2016

The low-frequency response of an acoustic guitar is strongly influenced by the combined behavior of the air cavity and the top plate. The sound hole–air cavity resonance (often referred to as the Helmholtz resonance) interacts with the first elastic mode of the top plate creating a coupled oscillator with two resonance frequencies that are shifted away from the frequencies of the two original, uncoupled oscillators. This effect has been modeled using finite elements for the top plate and boundary elements for the air cavity with rigid sides and back and no strings. The natural frequencies of both the individual and combined oscillators will be compared to measurements. The modes and natural frequencies of the coupled and uncoupled top plate were obtained using experimental modal analysis. An array of electret microphones was used above the sound hole to compare the phase of acoustic pressure with that of the top plate vibration. The uncoupled acoustic mode was then measured after placing sandbags on the top plate to restrict its motion. The measurements confirm the mode shapes, natural frequencies, and damping of the model.
Driver placement for selective modal excitation in flat-panel loudspeakers
Authors: Heilemann Michael C. , Anderson David , Bocko Mark F.
Published: May 2016

The audio quality of flat panel loudspeakers may be improved dramatically by selectively driving specific panel modes using an array of force actuators distributed on the panel surface. An optimal actuator array layout will maximize the energy coupling between the actuators and selected panel modes, while simultaneously eliminating modal crosstalk within the operating frequency band. We present results for optimal force actuator array layouts for both dynamic force actuators and piezoelectric patch actuators. Experiments on panel loudspeakers using optimized force actuator array layouts show that each of the panel modes within the tuning bandwidth may be independently addressed, and that the optimized array can limit the inter-modal crosstalk between −17 dB and −22 dB within the bandwidth using non-ideal force transducers. The methods described may also be useful in structural vibration control and noise reduction.
Measures of vibrational localization on point-driven flat-panel loudspeakers
Authors: Anderson David , Bocko Mark F. , Heilemann Michael
Published: May 2016

Previous publications on flat-panel, or distributed-mode, loudspeakers generally assume that a localized driving force is able to spread energy evenly across the surface of a panel. However, investigations have shown that panel vibrations remain localized around the driving point at high frequencies, and this paper presents a deeper investigation into this phenomenon. Energy spreading will only occur when the panel is actuated in a frequency region with a low density of modes, as many modes actuated together will combine to form a band-limited delta function at the location of the driving force. A quantitative measure of localization is introduced, based on the ratio of the energy contained in a small region around the driving point to the energy contained in the entire panel. Simulations demonstrate that the frequency cutoff for localized vibrational behavior is dependent on the damping rate of the modes and the location of the driver. Experiments validate this theory by analyzing the vibrational behavior of a plate subject to small inertial drivers at various locations with a laser vibrometer.
Experience-dependent plasticity in the neural weighting of pitch dimensions: A machine learning approach
Authors: Chandrasekaran Bharath , Reetzke Rachel , Yi Han-Gyol , Roeder Jessica , Xie Zilong , Maddox W.Todd
Published: May 2016

We conducted a cross-linguistic study to evaluate the impact of language experience on midbrain encoding of acoustic dimensions. Midbrain electrophysiological responses were recorded to the four Mandarin tones in native Chinese (N = 10) and English (N = 10) listeners, through a counter-balanced block design. English participants were trained over multiple days to achieve tone categorization accuracy and reaction time equal to that of the Chinese participants. We assessed the extent to which the four Mandarin tones could be discerned from the electrophysiological responses, using a data-driven machine learning approach. The machine learning output was used to generate dissimilarity matrices that were subjected to a multidimensional scaling (MDS) model. A two dimensional MDS solution emerged that corresponded to “pitch height” and “pitch direction” of the Mandarin tones. Findings derived from the individual differences scaling (INDSCAL) method revealed that, initially, pitch direction was weighted more by the Chinese participants relative to the English participants. However, following training, relative weighting on pitch direction was found to increase in the English participants, comparable to the Chinese participants. These results suggest that long-term language experience and short-term training with Mandarin tones selectively enhances weighting of pitch dimensions at the level of the auditory midbrain.
Perceptual compensation in nasal and nasalized vowels in Brazilian Portuguese
Authors: Marques Luciana
Published: May 2016

Listeners of a language can factor out effects of coarticulation in the signal, consistent with the language’s phonological patterns. Such perceptual compensation should not occur for phonetic features that are inherent to a segment, not effects of coarticulation. Vowel nasality is an interesting feature in this case, because it can be coarticulatory or inherent depending on the language. In Brazilian Portuguese (BP), it can be both. Contrastive nasal vowels may be followed by a brief nasal resonance (nasal appendix), while coarticulatory nasalized vowels must be followed by a nasal consonant. This study explores whether or not BP listeners perceptually compensate for vowel nasality in the perception of both phonemic nasal and coarticulatory nasalized vowels in BP, examining. If BP listeners process nasal vowels as phonemes, they should not attribute nasality in the vowel to coarticulation effects in any context, except nasal consonant. Randomized stimuli pairs containing oral, nasal and nasalized vowels in three different contexts (zero, nasal appendix, nasal consonant) were presented to BP listeners who had to judge which vowel was more nasal. Preliminary results demonstrate that BP listeners do not perceptually compensate for vowel nasality in any context, be it nasal appendix or nasal consonant.
Optimal time- and frequency-domain feature characterization for emotion recognition using electromyographic speech
Authors: Udhan Tejal , Bernadin Shonda
Published: May 2016

In this paper, different approaches for characterizing emotions using pattern recognition in EMG signal are proposed. In time domain features, the variability index (VI) of EMG signal is used as a feature for pattern recognition. Fuzzy clustering toolbox in MATLAB is proposed as the tool for pattern recognition. VI is proven to be a good criterion for characterizing EMG signals, having a good account for amplitude and signal value at the given instant of time. However, there has been very little research on its contribution in identifying closely related emotions. Frequency domain features involving modified mean frequency (MMNF) and modified median frequency (MMDF) are used for different spectral characteristics of speech and kinematic signals. Since emotion recognition using EMG is a relatively new and innovative concept, the extraction of optimal features in the time-domain and frequency-domain that can be used for pattern recognition is a necessary initial step. This paper will explore the use of clustering techniques for optimal feature characterization of emotions.
Attentive tracking in human audition
Authors: Woods Kevin , McDermott Josh
Published: May 2016

Auditory scenes often contain multiple sound sources, but typically one is of particular interest and must be selected for further processing. This “cocktail party problem” is especially difficult when sources are similar and change over time (e.g., speakers of the same gender). To study this situation, we introduced a paradigm in which listeners attempt to follow one of two synthetic voices that vary randomly in several feature dimensions (e.g., f0, f1, and f2). Psychophysical results from this task suggest that human listeners employ a movable focus of attention that follows sources of interest through their feature space. Here we utilize this paradigm to probe the performance limits of attentive tracking. We also report that listeners can learn the abstract “shape” of an attentively tracked trajectory if it occurs repeatedly in a stimulus set. Knowledge of trajectory regularities appears to benefit the tracking of target sources, and demonstrates another way in which attentive tracking might aid our ability to hear under challenging listening conditions.
Selective adaptation of crossmodal speech information is not the result of higher-level stimulus associations
Authors: Dias James W. , Rosenblum Lawrence D.
Published: May 2016

Selective adaptation of speech information can change perceptual categorization of ambiguous phonetic information (e.g., Vroomen and Baart, 2012). The results of a number of studies suggest that selective adaptation may depend on sensory-specific information shared between the adaptor and test stimuli (e.g., Roberts and Summerfield, 1981; Saldaña and Rosenblum, 1994). For example, adaptation to heard syllables can change perception of heard syllables, but adaptation to lipread syllables has not been found to significantly change perception of heard syllables. The lack of crossmodal influence in selective speech adaptation is inconsistent with other phenomena suggesting crossmodal influences occur early in the speech process (e.g., Rosenblum, 2008). For the current investigation, a replication of an attempt to induce crossmodal speech adaptation again failed to yield significant effects. However, a meta-analysis including these results with past attempts (Roberts and Summerfield, 1981) revealed that small crossmodal speech adaptation effects are significant across studies. Next, an attempt to induce crossmodal speech adaptation employing a much larger group of new participants did yield significant crossmodal adaptation effects. A replication using text-stimuli failed to induce similar adaptation effects, suggesting that the observed crossmodal effects are not the result of higher-level stimulus associations. The theoretical implications of these smaller crossmodal effects will be discussed.
Orthogonal interference of indexical information occurs even when phonetic contrasts are unambiguous across talkers
Authors: Choi Ja Young , Hu Elly R. , Perrachione Tyler K.
Published: May 2016

During speech perception, listeners maintain perceptual constancy despite highly variable associations between speech acoustics and abstract phonemic representations. Talker normalization facilitates speech processing by reducing the degrees of freedom for mapping encountered speech to phonemic representations. In our study, we investigated (i) whether talker normalization is obligatory even when there is no potential acoustic ambiguity between speech sounds and (ii) how the magnitude of potential ambiguity in acoustic-phonemic mappings affects talker normalization. We parametrically manipulated the phonetic similarity of words across three levels of difficulty for both vowels (easy: /bit/-/bot/, medium: /bɛt/-/bʌt/, hard: /bot/-/but/) and consonants (easy: /baɪ/-/saɪ/, medium: /baɪ/-/taɪ/, hard: /baɪ/-/paɪ/). In a series of speeded classification paradigms, listeners identified these words spoken by single or mixed talkers. Auditory word categorization was always slower in mixed-talker conditions than in single-talker conditions, even when there was no potential acoustic ambiguity between the target sounds. Furthermore, the processing cost imposed by mixed talkers was greatest when words were most similar phonetically. The results demonstrate (i) that talker normalization is obligatory in speech processing even when sounds are wholly acoustically distinct and (ii) that the facilitatory effect of talker normalization varies as a function of the potential ambiguity of acoustic-phonemic mapping.
Extrinsic talker normalization via rapid accumulation of talker-specific phonetic detail
Authors: Perrachione Tyler , Choi Ja Young
Published: May 2016

Talker normalization makes speech processing more efficient by reducing the degrees of freedom between the highly variable acoustic realization of speech and its abstract mental representation. However, it is unknown how processing gains afforded by talker normalization depend on the amount of talker-specific information available in the preceding speech context. We explored the timecourse of extrinsic talker normalization in a series of speeded classification tasks that parametrically varied the amount of talker-specific phonetic detail available in the preceding speech context. In single- and mixed-talker conditions, listeners identified a target word (either “boot” or “boat”) preceded by either no speech context, a short carrier phrase (“It's a...”), or a long carrier phrase (“I owe you a...”) spoken by the same talker as the target word. Listeners' word identification was always slower in the mixed-talker condition than the single-talker one. However, the amount of interference decreased as the amount of preceding speech context increased: the interference effect of the mixed-talker condition was greatest with no carrier, less with a short carrier, and least in the long-carrier condition. These results reveal how extrinsic talker normalization facilitates speech processing via rapid accumulation of talker-specific detail from the speech context.
Overview of civilian counter drone systems
Authors: Borel-Donohue Christoph
Published: May 2016

Unmanned aerial vehicles (UAV) commonly known as drones pose a significant threat to civilian and military personnel. This talk will explore several commercially available counter UAV systems developed for the civilian market. The systems discussed make use of electro-optic, acoustic, and radio frequency signatures to detect the presence of UAVs. The talk will also show some examples of acoustic, electro-optic, and radio frequency signatures measured on a few commercial drones.
Localization and separation of acoustic sources by using a 2.5-dimensional circular microphone array
Authors: Bai Mingsian R. , Lai Chang-Sheng
Published: May 2016

Circular microphone arrays (CMA) are preferred over more complex spherical microphone arrays (CMA) in the context of some audio applications because azimuth angles of spatial sound are considered more important than the elevation angles in those scenarios. However, CMA treats only two-dimensional sound fields without considering elevation angles, which can be a limitation in spatial audio rendering. This paper proposes a 2.5-dimensional (2.5D) CMA that consists of a baffled CMA and a vertical linear array on the top. In the localization stage, two minimum-variance-distortionless-response (MVDR) beamformers are applied to the circular array and the linear array, respectively. The product of the identified angular patterns yields the direction of arrival (DOA). In the separation stage, Tikhonov Regularization (TIKR) and Compressive Sensing (CS) are employed to extract the source signal amplitudes from the output signals from two arrays. The extracted signals are further processed by using an adaptive correlator to produce the source signal with improved quality. To validate the 2.5D CMA experimentally, a three-dimensionally printed circular array comprised of a 24-micro-electro-mechanical system (MEMS) microphone circular array and an 8-MEMS microphone linear array is constructed for localization and separation for sound sources.
Localization of plumbing noise in a multistory building using an acoustic camera
Authors: Kovacik Marek , Grythe Jørgen
Published: May 2016

Noise generated by plumbing pipes inside a multi-family dwelling can be a source of great annoyance on its residents. Corrective action of such noise begins with identification of the noise source followed by appropriate noise reduction measures. Source identification becomes more challenging when documentation on the plumbing design is missing or when the noise level is so low that it cannot be localized using practical methods. This paper describes a similar problem where low level noise generated by sanitary pipes is heard inside an apartment building. As a localization tool, an acoustic camera utilizing beam-forming technology is used to identify the noise location inside walls between two adjacent apartment units. Findings are visually presented and discussed.
Passive ocean acoustic tomography using ships as sources of opportunity recorded on an irregularly spaced free-floating array: A feasibility study
Authors: Kubicko Jacquelyn S. , Verlinden Christopher M. , Nichols Brendan V. , Sabra Karim G.
Published: May 2016

This presentation investigates the practical feasibility of using an adaptive volumetric array, such as a series of free-floating buoys with suspended hydrophones, which record ships as acoustic sources of opportunity in coastal waters for performing acoustic thermometry or other environmental inversions in near-shore environments in a totally passive manner. Ships are tracked using the Automatic Identification System (AIS). Numerical simulations using a standard normal mode propagation model were first used to test limitations of the proposed approach with respect to frequency band, drifting receiver configuration, signal to noise ratio, precision, and accuracy of the inversion results, along with sensitivity to environmental and position mismatch. Performance predictions using this model are compared with experimental results using at-sea data collected off the coast of New London, CT, in Long Island Sound during August of 2015.
Ray travel times in the northern Philippine Sea circulation from state estimates
Authors: Cornuelle Bruce , Gopalakrishnan Ganesh , Mazloff Matthew , Worcester Peter F. , Dzieciuch Matthew A.
Published: May 2016

The North Pacific Acoustic Laboratory (NPAL) Philippine Sea experiment deployed a variety of instruments in the northern Philippine Sea during April 2010 through April 2011, including six acoustic transceivers for reciprocal measurements of travel times between instruments. Five transceivers were moored in a pentagon approximately 660 km in diameter with the sixth transceiver moored in the center. Observed travel-time time series were compared with travel times computed from ocean state estimates made using an eddy-active regional implementation of the MITgcm that were constrained by satellite sea surface height and sea surface temperature observations and by temperature and salinity profiles from Argo, CTDs, and XBTs but not by the acoustic data. The similarities cross-validate the state estimates, while the differences provide a simple estimate of the novel information present in the travel times. Smoothness in the modeled sound speed field had a large effect on the ability to find eigenrays. The ocean state estimates were then re-computed to fit the acoustic travel times. The state estimate was able to match the travel times within their error bars and did not significantly increase the misfits with the other observations.
The integrated ocean dynamics and acoustics project
Authors: Duda Timothy F. , Lynch James F. , Lin Ying-Tsong , Zhang Weifeng G. , Helfrich Karl R. , Swinney Harry L. , Wilkin John , Lermusiaux Pierre F. , Makris Nicholas C. , Yue Dick Y. , Badiey Mohsen , Siegmann William L. , Collis Jon M. , Colosi John A. , Jachec Steven M. , Newhall Arthur E. , Wan Lin , Liu Yuming , Paoletti Matthew S. , Gong Zheng , Haley Patrick J. , Zhang Likun , Raghukumar Kaustubha , Allshouse Michael R.
Published: May 2016

The goal of timely and accurate acoustics modeling in the ocean depends on accurate environmental input information. Acoustic propagation modeling has improved to the point of possibly being ahead of ocean dynamical modeling from the standpoint that some significant ocean features having strong acoustic effects are not faithfully reproduced in many models, particularly data-driven ocean models. This in part stems from the fact that ocean models have developed with other goals in mind, but computational limitations also play a role. The Integrated Ocean Dynamics and Acoustics (IODA) MURI project has as its goals improving ocean models, and also making continued improvements to acoustic models, for the purpose of advancing ocean acoustic modeling and prediction capabilities. Two major focuses are improved internal tide forecasting and improved nonlinear internal wave forecasting, which require pushing the state of the art in data-constrained mesoscale feature modeling as well as developing specialized high-resolution tools. Results are reported on efforts to evaluate internal-tide accuracy in data-constrained models, to insert typically unresolved nonlinear internal waves with nonhydrostatic pressure dynamics into these models, and to make acoustical condition forecasts within three-dimensional operational volumes filled with internal waves.
Relative contributions of various modes of acoustic radiation force to creating mechanical stress in soft tissues
Authors: Sarvazyan Armen , Ostrovsky Lev
Published: May 2016

There are several physical effects responsible for the generation of acoustic radiation force (ARF) in soft tissue. The first effect, which is widely used in elastographic applications of ARF, is the transfer of momentum from ultrasonic wave in an attenuating medium due to absorption and scattering. Two other effects include reflection of sound wave from the boundaries between tissues differing in the acoustic impedance and second, spatial variations of the energy density of the propagating wave as a result of variations in the sound wave velocity in a medium. The latter mode of ARF can be directed both outward and toward the source of the propagating wave. In this study, relative contributions of various modes of acoustic radiation force are investigated both theoretically and experimentally. Using Green function for the acoustic potential near the boundaries between tissue structures, the expression for the motion of tissue interfaces under the action of short ultrasonic pulses is derived. Theoretical estimates of dynamics of the soft tissue response to various modes ARF induced by the microsecond range ultrasonic pulses are in agreement with experimental data. Findings of this study may serve as the basis of new approaches in soft tissue elastography.
Harmonic modulation of acoustic standing wave fields for tissue engineering
Authors: Doyle Timothy E. , Johnson Blaine , Patchett Brian D. , Sullivan Natalie C. , Sanjinez Dolly A. , Behan Ashley , Cowan Nicole , Butler Audrey P.
Published: May 2016

Acoustic standing wave fields can be used to produce virtual, scaffold-less templates for tissue patterning and engineering. To date, only simple tissue patterns have been attempted and achieved such as layered microstructures. This is due in large part to the use of single-frequency standing waves transmitted from a single source, producing planar nodes and antinodes in the cell/growth medium suspension. The purpose of this study was to explore with computational models the possibility of generating complex tissue microstructures using multiple sources with multiple frequency capabilities. The models simulated multifrequency standing waves (i.e., modulated by higher harmonics), multiple acoustic sources at various angles to each other, and corresponding acoustic cavities. The simulations demonstrated that interference patterns of standing waves could be produced with a three-dimensional complexity similar to that of natural biological structures. Such patterns included alveolar structures (thin, membrane-like walls encapsulating open cavities), lobular structures (cell clusters interlaced with duct- and vascular-like channels), and radial structures (iris-like patterns). Experiments for rendering these models to actual tissue structures have also begun, including the development of multifrequency transducers, using multiple transducers to generate a square lattice of cell walls and channels, and methods to reproduce natural tissue structures with high fidelity.
A generalized reconstruction framework for transient elastography
Authors: Bayat Mahdi , Ghosh Susanta , Alizad Azra , Aquino Wilkins , Fatemi Mostafa
Published: May 2016

Ultrasound transient elastography has emerged as a promising imaging modality for noninvasive evaluation of tissue mechanical properties. Most of the established techniques, however, rely on a pointwise measurement of the induced shear wave speed based on a localized planar wave propagation assumption. This assumption is not always true as most tissues exhibit heterogamous characteristics. Complex interaction of different geometries, boundary conditions, and tissue composition limit the ability of these techniques such that additional spatial-temporal filtering and signal truncations are required. These signal conditionings might lead to improvement in elasticity estimation for some cases (e.g. certain geometries, material properties) but the fact that they do not stem from a comprehensive framework can lead to unrealistic mechanical properties in more complex scenarios. We present a generalized framework based on inverse solution of the elasto-dynamic equations which can simultaneously solve for material properties and geometries without requiring any prior knowledge of the wave types. The method is based on a modified error in constitutive equation (MECE) without requiring additional filtering. We present the results of standard phantom studies based on acoustic radiation force excitation and fast ultrasound tracking where MECE technique successfully estimated the elastic moduli and geometry. [Work supported by NIH grant CA174723.]
Differentiation of breast lesions based on viscoelasticity response at sub-Hertz frequencies
Authors: Bayat Mahdi , Nabavizadeh Alireza , Kumar Viksit , Gregory Adriana , Alizad Azra , Fatemi Mostafa
Published: May 2016

There is an urgent need for a noninvasive but more accurate method to assess breast masses before referring the patient to biopsy. In this study, we use the slow viscoelastic creep response to evaluate breast masses at low (<1 Hz) frequencies. This method uses an automated ramp-and-hold compression device and a high-frame-rate ultrasound imaging to track tissue strain. The study cohort included 30 pre-biopsy patients with suspicious masses. Sequential high-frame-rate images were recorded during the ramp-and-hold process. The data were used to calculate the creep response at each point. Then, using a classic linear model, the retardation-time map was created within the field of view. The resulting retardation-time maps exhibited clear distinction of the mass margins. The results showed that benign breast lesions appear with an increase in creep retardation-time compared to surrounding breast glandular tissue, and the opposite trend was true for the malignant lesions. Statistical analysis of the viscoelasticity features revealed that a contrast feature based on the retardation-time is an accurate classifier of malignant and benign masses (P<0.0003). It is concluded that the retardation time is a promising biomarker for differentiation of breast masses. [Work supported by NIH Grant CA168575.]
Characterization of porous material and a sound suppression system using a shock tube
Authors: Good Erica K. , Taylor Robert , Luskin James , Conaway Zachary
Published: May 2016

This work describes the design and construction of an experimental apparatus for testing shock waves and a sound suppression system for student learning. The shock tube will be used to characterize both porous media and the suppression system response to a non-linear shockwave. The effects of a plane shock wave will be used to determine acoustic properties, such as transmission, reflection, and absorption coefficients, of rigid frame porous materials. The suppression system is comprised of 20 conical chambers that cause a loss of energy in the shock wave by the expansion and acceleration of the flow. The sound pressure level from the shock is in excess of 140 dB re 20 μPa. Material properties and the performance of the designed suppression system will be presented.
Reverberation characterization inside an anechoic test chamber at the weapon sonar test facility at Naval Undersea Warfare Center Keyport Division
Authors: Eastland Grant C. , Buck William C.
Published: May 2016

The Weapon Sonar Test Facility (WSTF) at NUWC Keyport, WA, is a 34 500 gallon pressure tank currently used for test and evaluation of torpedo sonar arrays. The tank has many more possible uses and complete characterization of the testing environment needs to be performed. One of the methods of characterization being used is the determination of the reverberation time of the anechoic chamber. Applying Sabine-Franklin-Jaeger theory of reverberant rooms, an experimentally determined reverberation time T60 for a “live room” can be used to provide an upper bound for the reverberation time in the chamber. Utilizing the Eyring theory of “dead” rooms will provide a better determination of T60 for the anechoic chamber, hence provide a lower bound characterization. The experimental method involves determining the spatially averaged acoustic energy of an initial tone and the corresponding reflection from the chamber wall seen in a recorded signal. These values determined from the root mean square signal voltage determine the wave decay time, τ. The decay time is directly related to the reverberation time through T60 = 6τ×loge10. The reverberation was determined to be within an acceptable tolerance for testing and evaluation.
Loudspeaker system
Authors: Pisani Gordon , Bartolomeo Nico , Mascolo Joseph , Wolff Gabrielle , Laprade Alexander , Gorczynski Ethan , Turo Diego
Published: May 2016

This presentation will demonstrate the design, construction, testing, and final performance of a student designed loudspeaker system. The purpose of this project was to create a 2.1 channel, high quality sound system that produces a bandwidth just short of the full range of human hearing, approximately 20–20 000 Hz. This system consists of 2 two-way speakers utilizing a woofer and tweeter, with one separate subwoofer. The function of this particular sound system is to reproduce sound faithfully in a 350–400 square foot room. The performance characteristics of this system will be presented through frequency response and linearity testing results. These tests were used to calculate and implement modifications made to the cabinet size and shape while adhering to a budget of $1200. The analysis of every performance will also reveal data for different electrical component values in designing the Linkwitz-Riley crossovers.
Classification and optimization of beam responses synthesized from non-uniformly spaced linear arrays
Authors: Nguon Chrisna , Misiunas Nicholas , Au Jenny , Chandra Kavitha , Thompson Charles
Published: May 2016

A probability distribution for the element positions on a linear array is derived so as to match a target beam pattern that approximates a Dolph-Chebychev array. The random beam responses generated from such a configuration exhibit a mean profile in observation angle that is invariant with element size N but the variance decays as 1/N. The beam responses are classified considering the N-1 inter-element distances as features that control the deviation from the mean beam profile and the variance of the class. The inter-element distance constraints are incorporated in a computational model of the firefly algorithm (FA) designed to optimize the location of the array elements that match the target beam pattern. The FA consists of a set of fireflies, representing arrays, which move through the solution space with directed movement based on each fireflies' fitness and random movement to ensure adequate exploration and avoid local minima. The fitness function is a combination of a target beam-width, side-lobe level, and bounds on the position dependent mean inter-element distances. The variance of the resulting beams are compared with the result from the unconstrained random placement of elements as a function of the element size and side-lobe parameters.
The affect of audio enhancement on vocal timbre
Authors: Edwards Matthew
Published: May 2016

In acoustic singing, performers learn to finely tune their resonance in order to control the spectral output of their voices. However, in amplified singing styles, microphone frequency response, equalization, and other editing tools can affect our perception of resonance. These effects can have positive and negative impacts on the spectral output of a singer’s voice that teachers must be aware of when training Contemporary Commercial Music (CCM) artists. This presentation will include information on how audio equipment can alter the timbre of the voice, affect our perception of registration, and alter the singing power ratio.
Why fry? An exploration of the lowest vocal register in amplified and unamplified singing
Authors: Nix John P.
Published: May 2016

Vocal fry is the lowest register in human vocal production. It can be defined on a psycho-acoustical basis as vocalization at fundamental frequencies below the perceptual threshold for discrete events, which occurs at approximately 70 Hz. Vocal fry has become more commonplace in conversational speech and amplified singing styles such as popular and country, but it is typically unused in non-amplified accompanied performances of most Western Classical music. The author’s presentation will include the results of a survey of listener preferences of performances of popular and country performances with and without vocal fry, and the results of an experiment to examine what acoustical information is being transmitted to listeners during the fry portions of performances.
A comparison of readily available sound level apps with respect to field performance and applications in architectural education
Authors: Woolworth David S. , McGlohn Emily
Published: May 2016

This investigation was driven first by a suggestion that apps be used by residents, sound creators, and police to evaluate sound sources in regard to regulations, and then further in an effort to understand how apps might be useful for architecture students studying active building or environmental systems. To simulate a typical educational or “street” situation, soundwalks are taken in which various sources are measured using apps on different phones as well as a Type 1 sound level meter for reference. The results are compared and a recommendation of appropriate application is made.
Mock-up testing for CrossFit vibration
Authors: Manley David , Bridgewater Ben
Published: May 2016

A new construction mixed-use building was proposed in Crested Butte, CO, to consist of ground floor fitness, second floor commercial office, and third floor residential. It was anticipated that the ground floor fitness space would include CrossFit style workouts which often incorporate Olympic weight lifting and dropping activities. Due to the remote project location, five different mock-ups of impact and vibration isolation products were assembled in a gym in Denver to test the reduction in vibration levels from heavy weight impacts. Acceleration levels were measured on the concrete slab on grade close to the weight drop and in the soil outside the gym area. Mock-up testing methodology and results are presented. Based on the test results, recommendations on products, assemblies, and construction details were provided.
Radiation stress, momentum, angular momentum, and power relationships for axisymmetric objects in acoustic vortex and coaxial wavefields
Authors: Marston Philip L. , Zhang Likun
Published: May 2016

Starting with P. Westervelt’s formulation of the radiation stress tensor and G. Maidanik’s application to angular momentum in the 1950s, some recent theorems concerning acoustic radiation forces, torques, angular momentum, and power flow will be illustrated. Applications include acoustic vortex wavefields, Bessel beams, and other invariant beams. While derivations are simplified for objects surrounded by inviscid fluids [L. Zhang and P. L. Marston, Phys. Rev. E. 84, 035601 (2011); L. Zhang and P. L. Marston, Phys. Rev. E. 84, 065601 (2011); L. Zhang and P. L. Marston, J. Acoust. Soc. Am. 131, EL329–EL335 (2012)], for spheres in slightly viscous liquids limiting approximations are known at long wavelengths that recover results derivable by other approaches [L. Zhang and P. L. Marston, J. Acoust. Soc. Am. 136, 2917–2921 (2014); P. L. Marston, POMA 19, 045005 (2013)]. Theorems obtained are helpful for relating negative radiation forces to the asymmetry of the scattering pattern for spheres in Bessel beams. The relation with extinction theorems is noted [L. Zhang and P. L. Marston, Bio. Opt. Express 4, 1610–1617 (2013); (E) 4, 2988 (2013)] as well as momentum and angular momentum radiated by sources. [Work supported in part by ONR.]
Angular momentum conservation and symmetry
Authors: Zhang Likun , Li Yong , Jiang Xue , Liang Bin , Cheng Jian-chun
Published: May 2016

The conservation of angular momentum during the interactions of sound fields with objects is associated with the symmetry in the fields and the objects. Simple arguments of symmetry and/or conservation can have efficient applications in analyses of angular momentum transfer during the interactions. Scenarios that will be illustrated include: (a) conserved phase distribution for analysis of torques exerted by vortex beams on axis-symmetric objects for its connection with absorption [L. Zhang and P. L. Marston, JASA 129, 1679–1680 (2011); Phys. Rev. E. 84, 065601 (2011)], (b) symmetric superposition of vortex beams for analysis of torques exerted by orthogonal waves on a small compressible object in a slightly viscous fluid [L. Zhang and P. L. Marston, JASA 136, 2917–2921 (2014)], and (c) broken symmetry in a planar meta-screen for introducing angular momentum into a plane sound field of null angular momentum [Y. Li et al., Phys. Rev. Appl. 4, 024003 (2015); X. Jiang et al., in preparation].
Sonic Screwdrivers and Tractor Beams: what the Acoustics Community often gets wrong about gradient forces versus radiation pressure
Authors: Spalding Gabriel C. , Dahl Patrick , Yang Zhengyi , Glynne-Jones Peter , MacDonald Michael P. , Demore Christine , Cochran Sandy
Published: May 2016

Through discussion of our sonic screwdriver and tractor beam experiments, we aim to highlight the respective advantages of conservative and non-conservative forces. Commonly in acoustic trapping, conservative, gradient-induced mechanisms (e.g., standing waves) are used to manipulate matter. Such situations are reasonably described in terms of potential energy landscapes, an approach also applied to optics, for applications such as cell sorting [MacDonald et al, Nature 426 (2003)]. No such description is possible for radiation pressure, which is non-conservative, a distinction that is sometimes muddled in the literature, although it was made clear even in early work [e.g., King, Proc. R. Soc. Lond. A 147 (1934); Gor’kov, Sov. Phys. Doklady 6, 773 (1962)]. Our “sonic screwdriver” makes use of two non-conservative mechanisms: levitation by radiation pressure and rotation by transfer of azimuthal momentum components [Demore et al., PRL 108 (2012)]. We also note that the term “tractor beam” has often been reserved to describe an effect involving non-conservative forces, and demonstrate an attractive force produced in such an arrangement, even against a net momentum flux [Demore et al., PRL 112 (2014)].
Single-beam acoustical tweezers: Concept, theory, and experimental realization
Authors: Baresch Diego , Marchiano Régis , Thomas Jean-Louis
Published: May 2016

Vortex beams are characterized by wavefronts twisting around their axis of propagation. A three dimensional region of silence tightly bounded by a high intensity ring emerges from the phase's screw dislocation, which can be used for particle manipulation. In this paper, we present the first experimental realization of selective 3D acoustical tweezers with a single beam. It had long been recognized that acoustic radiation pressure pushed objects in the direction of propagation. As we will show, by accurately modeling the radiation pressure exerted on an elastic sphere lying in any arbitrary location, it is possible to identify this axial ejection and to propose a solution to counterbalance this effect. The acoustic puling force of a properly focused vortex beam is explained as a result of the particle's monopolar mode annihilation on the propagation axis. Consequently, these beams act as a stable trap in three dimensions. An experimental realization has been proposed in water using a ultrasonic beam (1 MHz). Elastic particles in the sub-millimetric range have been accurately trapped and manipulated. This setup's dexterity and selectivity is the macroscopic analogous of optical tweezers exerting, however, forces which are 5 orders of magnitude stronger at comparable intensities.
Understanding the speech-understanding problems of older adults
Authors: Humes Larry E.
Published: May 2016

This presentation will provide an overview of the nature of the difficulties experienced by older adults with impaired hearing when listening to speech in noise. Many older adults may experience a “triple whammy” of difficulties: (1) peripheral pathology in the cochlea or auditory nerve; (2) age-related deficits in central-auditory function; and (3) age-related changes in cognition that impact speech perception. Each of these difficulties may appear in isolation or in various combinations for a given older adult. This presentation will review the evidence surrounding each of these potential sources of difficulty and the relative contribution of each across various types of “noise” backgrounds. In addition, the relative importance of each source of difficulty for aided and unaided listening conditions will be considered. [Work supported, in part, by NIA R01 AG008293.]
Interactions between spectral and temporal processing on speech understanding in cochlear-implant users
Authors: Oxenham Andrew J.
Published: May 2016

Understanding speech in noise remains one of the greatest challenges facing people with cochlear implants (CIs). For normal-hearing listeners, speech understanding in noise seems limited to a large extent by the modulation energy (temporal fluctuations) present in the noise, rather than the noise energy itself. This finding does not seem to apply in CI users, for whom no difference is observed between maskers with or without inherent temporal fluctuations. The current explanation for this qualitative difference between the results from normal-hearing listeners and CI users is based on the lack of spectral resolution in CIs, which results in an overlap of stimulation from adjacent channels, and a resultant smoothing of the temporal envelopes produced by noise. This study reviews evidence for this hypothesis, and tests some arising predictions, including the effects of narrower stimulation patterns (tripolar versus monopolar) and increased spacing between active electrodes in both CI users and normal-hearing listeners using envelope-vocoder simulations of CI processing. [Work supported by NIH grant R01 DC012262.]
Efficacy and viability of an algorithm to improve speech understanding in noise for the hearing impaired
Authors: Healy Eric W.
Published: May 2016

A primary complaint of hearing-impaired individuals involves poor speech understanding when background noise is present. Hearing aids and cochlear implants often allow good speech understanding in quiet backgrounds. However, the listeners’ noise intolerance and the devices’ inability to effectively combat background noise often conspire to produce poor performance in noise. Perhaps surprisingly, effective solutions to this problem have remained elusive despite considerable effort. One promising solution involves a single-microphone algorithm to extract speech from background noise. The algorithm is based on the concept of the ideal binary or ratio mask, and employs standard machine-learning techniques to train a deep neural network to estimate the mask, given only the speech-plus-noise mixture. Existing data indicate that large intelligibility increases by hearing-impaired listeners may be obtained across a variety of noisy conditions. In this talk, an overview of this approach will be provided, and the potential for implementation into hearing aids and cochlear implants will be discussed. [Work supported by NIH.]
Metrics for evaluating noise suppression in hearing aids
Authors: Kates James
Published: May 2016

Noise suppression in audio systems such as hearing aids typically involves modifying the speech envelope. Noisy speech segments having poor signal-to-noise ratios are attenuated while those considered to be primarily speech are left at or near full intensity. Both spectral subtraction and ideal binary mask noise suppression work on this principle. Recent work in the development of speech intelligibility and speech quality metrics indicates that the envelope fidelity, computed as the correlation coefficient between the processed noisy speech envelope and the original noise-free speech envelope, is an important factor in determining noise-suppression benefit. This result implies that the best noise suppression system is one in which the envelope of the processed noisy speech is restored to match as closely as possible that of the original noise-free speech. The potential benefits and limitations of envelope restoration for noisy speech are analyzed using the HASPI intelligibility and HASQI quality metrics, and the performance of existing noise suppression processing is compared to that of envelope restoration for both normal-hearing and hearing-impaired listeners.
Neuro-steered noise suppression for auditory prostheses
Authors: Francart Tom , Das Neetha , Van Eyndhoven Simon , Biesmans Wouter , Bertrand Alexander
Published: May 2016

In a multi-speaker scenario, a major challenge for noise suppression systems in hearing instruments is to determine which sound source the listener is attending to. It has been shown that a linear decoder can extract a neural signal from EEG recordings that is better correlated with the envelope of the attended speech signal than with the envelopes of the other signals. This can be exploited to perform auditory attention detection (AAD), which can then steer a noise suppression algorithm. The speech signal is passed through a model of the auditory periphery before extracting its envelope. We compared 7 different periphery models and found that best AAD performance was obtained with a gamma-tone filter bank followed by power-law compression. Most AAD studies so far have employed a dichotic paradigm, wherein each ear receives a separate speech stream. We compared this to a more realistic setup where speech was simulated to originate form two different spatial locations, and found that although listening conditions were harder, AAD performance was better than for the dichotic setup. Finally, we designed a neuro-steered denoising algorithm that informs the voice activity detection stage of a multi-channel Wiener filter based on AAD, and found a large signal-to-noise-ratio improvement at the output.