bluuebunny/crossref_metadata_2025 huggingface dataset:
The speakers in the room corpus
Authors: Lawson Aaron , Ni Karl , Richey Colleen , Armstrong Zeb , Graciarena Martin , Stavish Todd , Stephenson Cory , Hetherly Jeff , Gamble Paul , Barrios Maria
Published: April 2018

The speakers in the room (SITR) corpus is a collaboration between Lab41 and SRI International, designed to be a freely available data set for speech and acoustics research in noisy room conditions. The main focus of the corpus is on distant microphone collection in a series of four rooms of different sizes and configurations. There are both foreground speech and background adversarial sounds, played through high-quality speakers in each room to create multiple, realistic acoustic environments. The foreground speech is played from a randomly rotating speaker to emulate head motion. Foreground speech consists of files from LibriVox audio collections and the background distractor sounds will consist of babble, music, HVAC, TV/radio, dogs, vehicles, and weather sounds drawn from the MUSAN collection. Each room has multiple sessions to exhaustively cover the background foreground combinations, and the audio is collected with twelve different microphones (omnidirectional lavalier, studio cardioid, and piezoelectric) placed strategically around the room. The resulting data set was designed to enable acoustic research on event detection, background detection, source separation, speech enhancement, source distance, sound localization, as well as speech research on speaker recognition, speech activity detection, speech recognition, and language recognition.
Simulation of computerized screening for phonological errors in spoken utterances
Authors: Biswas Amitava , Thames Anita , Cloud Steven
Published: April 2018

Fast and reliable diagnosis of phonological errors is often necessary. This presentation describes a simulation of computerized screening for phonological errors in spoken utterances. A set of common words will be included in a database. Every word in the database will be linked to several other words according to minimal pairs. For example, CAT will be linked to KATE, CUT, CAB, CASH, KITE, BAT, FAT, HAT, MAT, RAT, etc, whereas, RAT will be linked to CAT, RATE, WRITE, RASH, RAM, RAN, RACK, RAG, RAP, etc., and RAN will be linked to RAT, PAN, RAM, BAN, CAN, RUN, RAIN, etc. A set of hypothetical patterns of phonological errors in articulation will also be included in the database. The simulation will explore efficacy of navigating through the data base with the help of a custom software in predicting and correctly identifying the particular patterns of phonological errors.
Automated formant tracking using reassigned spectrograms
Authors: Fulop Sean A. , Shadle Christine H.
Published: April 2018

The measurement and tracking of formant frequencies is usually accomplished with a linear prediction spectrum estimate, combined with a heuristic-based tracking algorithm. Despite the broad acceptance of this procedure over decades, there is a known bias towards the nearest harmonic, and formant estimates are less accurate for speech with higher fundamental frequency. Previous research has established the superior accuracy of formant measurement using the reassigned spectrogram; however, the measurements must be done by hand, severely limiting its usefulness. Here we present a technique that can locate formants automatically in a reassigned spectrogram. First, the reassigned spectrogram of the signal must be heavily “pruned” so that only a few points remain which highlight the detected frequency components, which are the presumed formants. Second, we apply a simple ridge-finding routine to the pruned spectrogram, to determine the formant tracks automatically. This process was tested on speech samples—/hVd/-words—by four speakers previously used in a comparison of 5 automatic algorithms and manually-measured reassigned spectrograms, and on breathy and falling tones in a speaker of Hmong. The results of this process are generally as good or better than the manual measurement when the optimal parameters for ridge-finding are used.
A method for quantifying vowel change directionality
Authors: Morris Miranda R.
Published: April 2018

Vowels are a common variable used in studies on phonetic accommodation and imitation. The standard practice for measuring changes in vowels provides an absolute value of the distance between a baseline vowel production and a target relative to the distance between shifted/imitated vowels and that same target. While distance alone is useful, a vector based approach would form a more complete picture as capturing distance alone can mask phenomena such as exaggeration. Vowels are a two-dimensional variable; thus, a directionality measure for changes in vowels relative to some target is crucial for fully capturing phenomena in imitation. Proposed is a trigonometric approach to quantifying the directionality of vowel changes. While this method was created to aid in studies on phonetic imitation, it is also potentially useful for any work involving vowel change with a known acoustic target.
Passive acoustic mapping and B-mode ultrasound imaging utilizing compressed sensing for real-time monitoring of cavitation-enhanced drug delivery
Authors: Crake Calum , Finn Seán , Marsac Laurent , Gray Michael , Carlisle Robert , Coussios Constantin , Coviello Christian
Published: April 2018

Ultrasound imaging presents a high-speed, low-cost approach for monitoring of focused ultrasound (FUS) therapy, and includes both conventional (B-mode) sonography and passive acoustic mapping (PAM) of acoustic emissions (Gyöngy et al., 2010, Salgaonkar et al., 2009). Incorporation of novel algorithms and other signal processing techniques have improved the resolution and processing speed of PAM. However, while hardware developments such as increasing channel counts provide unprecedented data capture ability, real-time processing of the growing data stream presents an evolving challenge. Previous work has employed sparse array processing techniques for PAM including matching and basis pursuit (Gyöngy & Coviello, 2011) and co-array processing (Coviello et al, 2012). Here we propose to extend PAM utilizing compressed sensing (CS). Acoustic emissions from FUS may be sparse in several domains, e.g. due to limited regions of space and time in which cavitation is likely from a focused transducer, and correlation between sensors. By exploiting this sparsity, CS allows recovery of signals sampled well below the Nyquist limit. CS-PAM thus facilitates PAM with improved spatial resolution compared to conventional methods, from fewer measurements, allowing improved image quality and reduced computational load. The technique was demonstrated for monitoring of FUS-mediated drug delivery in a murine tumor model.
Identification of fin whale 20-Hz pulses and other vocalizations from passive monitoring with a coherent hydrophone array
Authors: Huang Wei , Garcia Heriberto A. , Ratilal Purnima
Published: April 2018

A large variety of sound sources in the ocean, including biological, geophysical and man-made activities can be simultaneously monitored over instantaneous continental-shelf scale regions via the passive ocean acoustic waveguide remote sensing (POAWRS) technique by employing a large-aperture densely-sampled coherent hydrophone array. Millions of acoustic signals received on the POAWRS system per day can make it challenging to identify individual sound sources. An automated classification system is necessary to enable sound sources to be recognized. Here each detected acoustic signal is represented by an amplitude weighted pitch track which describes its fundamental frequency-time and amplitude variation. Multiple features are extracted from the pitch track including mean, minimum and maximum frequencies, bandwidth, signal duration, frequency-time slope and curvature, as well as several amplitude weighted features. A large training data set of fin whale 20-Hz pulses and other vocalizations are gathered after manual inspection and labeling. Next, multiple classifiers including logistic regression and decision tree are built and tested for identifying the fin whale 20-Hz pulses and other vocalizations from the enormous amounts of acoustic signals detected per day. The performance of the classifiers are evaluated and compared
Shallow water multichannel sub-bottom profiling using Kirchhoff imaging
Authors: Ristow João P. , Bousfield Marina M. , Azevedo Gregório G. , Cordioli Julio A. , Paul Stephan
Published: April 2018

Single-channel sub-bottom profilers are commonly used for shallow water seismic investigation. This work reports the design and construction of a multiple channel profiler that allows more effective seismic imaging techniques. The system is composed of a controlled acoustic source and a 1.6 m long array with eight evenly spaced receiver channels, and operates using CHIRP signals with frequency ranging from 800 Hz to 8 kHz. A survey was carried out to obtain seismic data in the region of Lagoa da Conceição, Santa Catarina, Brazil. The acquired data was processed using a seismic imaging technique based on the pre-stack Kirchhoff time migration. Both the multichannel seismic data acquisition system and the applied imaging technique showed to be robust for acquiring subsurface information. The use of the pre-stack Kirchhoff time migration resulted in a seismic image with reflection events allocated in the real position and without bow-tie effects since it repositions the diffracted energy back to the spreaders that originated it. This procedure results in a more reliable subsurface image of the studied environment, when compared with single-channel sub-bottom profilers.
Scattering of low-frequency sound by infinite fluid and solid cylinders
Authors: Baynes Alexander B. , Godin Oleg A.
Published: April 2018

Wave scattering by obstacles is typically studied assuming plane wave incidence. However, full Green’s functions are necessary for problems where the separation of the scatterer from sources, interfaces, or other scatterers is comparable to the dimensions of the scatterer itself. In this paper, the two-dimensional problem of scattering of monochromatic cylindrical waves due to infinite cylinder embedded in a homogeneous fluid is considered. Fluid and solid cylinders are studied, and soft and hard cylinders are revisited. The exact solutions for the Green’s functions are expressed as an infinite series of cylindrical functions with complex amplitudes determined by the acoustic boundary conditions at the surface of the cylinder. Here, we derive closed-form asymptotics for the scattered field in the regime when the scatterer dimensions are small compared to wavelength, i.e., Rayleigh scattering. The scattered wave approximation is valid for arbitrary source and observation points outside the scatterer and is expressed as a simple sum of fields due to three image sources. Image source solutions were anticipated due to classically studied electrostatic analog problems involving dielectric cylinders. Image source representation allows physical insight into the scattering physics and suggests simple yet accurate analytic solutions to interface and waveguide scattering problems.
A review of acoustic research in physical educational settings
Authors: Ryan Stuart
Published: April 2018

This presentation will review past and current acoustic research in physical education settings. Research suggests that the acoustic issues in physical education settings are likely detrimental to student learning (Ryan, 2009a; Ryan, Grube, & Mokgwati, 2010; Ryan & Mendell, 2010). The problem is compounded if there are students with special needs including learning disabilities, attention and auditory disorders, and students who are listening and learning in a non-native language. In addition, typical classroom teachers are faced with many voice concerns; however, physical education teachers are faced with poor acoustically designed gymnasiums, covered areas, and loud outdoor teaching environments which are more challenging than a typical indoor classroom (Ryan, 2009a). Ryan (2010) viewed physical education settings as “hostile listening environments” and have the potential to effect student learning and damaging teacher’s voices.
Acoustical analysis of preschool classrooms
Authors: Grieco-Calub Tina M. , Peng Z. Ellen
Published: April 2018

The present study explores acoustical parameters, including unoccupied and occupied noise levels and reverberation time (RT), in typical preschool classrooms located in the northern suburbs of Chicago. The study was motivated by the following observations: (1) preschool classrooms are often established in buildings that were not initially constructed to be learning spaces; (2) poor classroom acoustics interfere with skills related to academic outcomes including speech perception, serial recall, literacy, and cognitive skills; and (3) younger children are at greater risk for impaired performance in acoustically complex environments. The study was designed to determine whether these preschool classrooms meet existing classroom acoustics standards (ANSI S12.60-2010). Measurements were made with microphones positioned in locations of the classrooms where children typically engage in activities during school season to reflect realistic building operation situations. Unoccupied and occupied noises were recorded over long durations, with intensity and spectral analysis of the recordings conducted off-line. RTs were measured and quantified at the same microphone positions. Preliminary results suggested that none of the preschool classrooms met the recommended unoccupied noise levels, while 3/11 classrooms were found to have longer than recommended RTs. Occupied noise in these preschool classrooms, and implications for learning, will be discussed.
Classroom soundscape—Virtual and real
Authors: Chung Andy , To W M
Published: April 2018

The sonic environment which both teachers and students are provided in a classroom is important for effective communication during the learning process. How to design and optimize for a classroom requires a study of its soundscape including the feedback of potential users on their perception of the sonic environment. This paper presents how virtual technology is used to characterize the classroom soundscape and how it is related to the reality.
Noise in and around schools
Authors: Beristain Sergio
Published: April 2018

Incoming noise is usually a major problem for many schools which can drastically affect the teaching process, and often times this is the case due mainly to the traffic and community noise, but also in many cases the schools become themselves large noise sources, affecting the community tranquility. A case study is presented which shows the noise levels and acoustics characteristics of a basic school located in the middle of a middle class suburb, emphasizing the main nuisances that the school receives from the surroundings, and also those produced by the school to the community throughout its normal activities from before opening the school doors in the morning, till the last student has leaved, and during the teaching and leisure activities of young students.
Multifunction speech intelligibility in a renovated historic room
Authors: Bochat Sarah L. , Doing Michael B. , Nelson Peggy B. , Jaramillo Ana M. , Olson Bruce
Published: April 2018

Shevlin Hall, a historic building erected in 1906, is presently home to the department of Speech-Language-Hearing Sciences on the University of Minnesota campus. The primary classroom space is a 20' x 50' room with a high coffered ceiling used for lecture-style classes, discussion groups, and social events. The space underwent renovation in 2013 to address room acoustics and sound design. Post-renovation acoustic measurements and subjective responses reflect favorable outcomes for lecture conditions. Challenges remain for small-group and active discussion classroom arrangements. The current study evaluated predicted and measured speech intelligibility for small-group and active discussions. Preliminary data suggest good intelligibility throughout the room for the condition of lecture-style presentation. Reduced intelligibility is suggested for the condition of talker in back and receiver at instructor or front center position. Implications for sound design will be discussed.
Exploring relationships between measured acoustic, lighting, and indoor air quality metrics logged in K-12 classrooms
Authors: Smith Kieren H. , Wang Lily M.
Published: April 2018

Comfort in classrooms engenders learning and countless factors contribute to the quality of an educational environment. A longitudinal study measuring classroom conditions, including indoor air quality, lighting, and acoustic components has amassed data from 220 schools in the Eastern Nebraska area. Measurements logged at time intervals no longer than 5 minutes were taken for two full days in each classroom, three times in a year. Using this data, the relationships over time that acoustic factors have with indoor air quality, thermal comfort, and lighting factors are explored. Correlations are investigated between average measured values, time and spatial variation of measured values, and spectral composition of the acoustic metrics. Besides correlations between the assorted measured quantities, other relationships between the design of the building mechanical and lighting systems and resulting acoustic conditions are discussed; for example, do heat pumps or VAV boxes contribute to significantly different acoustic conditions? [Work supported by the United States Environmental Protection Agency Grant No. R835633.]
Noise level prediction in a bakery using panel contribution analysis combined with scale modeling
Authors: Cheng Gong , Herrin David
Published: April 2018

Panel contribution analysis combined with scale modeling was utilized to determine the sound pressure level at different positions in a bakery. The primary sources of noise were three heating and air conditioning units. Each of the units was discretized into several patches and the volume velocity for each patch was measured. Transfer functions were measured between the patches and two different customer locations for the full-scale baseline case and using a 1/10 scale model. Transfer functions for the scale model were adjusted to account for air attenuation in the full-scale case. Panel contribution analysis was used to predict noise levels assuming both correlated and uncorrelated sources. Predicted results compared well with measurement. The suggested method can be used to assess the sound pressure levels in large interior spaces before they are constructed so long as the source positions are known. In addition, contributions from the individual sources can be assessed and ranked.
Analyzing frequency responses from time-windowed impulse responses across assorted types of room usage
Authors: Boyd Brenna N. , Wang Lily M.
Published: April 2018

Lokki et al. (2015) examined how frequency responses in concert halls change over different time-windowed periods of the impulse response across low, mid, and high frequencies, and connected that with perceived quality in those halls. Time windows were applied to highlight the direct, early, late, or total sound energy, and the resulting frequency responses were overlaid on the same graph to view easily the difference and growth between the time periods across frequencies. In this project, such frequency response analyses across different time-windowed impulse responses are applied to spaces other than concert halls, including a variety of classrooms, worship spaces, and music-centered and speech-centered performance spaces. The objective is to explore the utility of this method of analysis for assorted room usages. In this presentation, interesting cases and trends within individual rooms, within one type of room usage, and between different types of room usage are discussed, as well as this analysis method’s advantages and disadvantages.
First-order approximation of diffraction at a rectangular rigid plate using the physical theory of diffraction
Authors: Rozynova Aleksandra , Xiang Ning
Published: April 2018

Efficient predictions of sound diffraction around objects is vital in room-acoustics simulations. Recent efforts using the Physical Theory of Diffraction (PTD) [Rozynova, A. and Xiang, N., J. Acoust. Soc. Am., 141, pp. 3785 (2017)] have been reported on some canonical objects of infinite size. This research has been further developed based on solving diffraction problems on finite, rectangular plates which are free from grazing singularities. Separate solutions are considered for diffraction from two pairs of the rigid plate edges, the edge waves around the plate corner are ignored due to the lower order of diffraction contributions in the far-field. This paper will discuss numerical implementations of the theoretical predictions to demonstrate the efficiency of the PTD in room-acoustics simulations for finite sized rectangular plates. The implemented numerical results will also be compared with some preliminary experimental results carried out with a goniometer.
Temporal fine structure and zebra finch vocalizations
Authors: Prior Nora H. , Smith Edward , Ball Gregory F. , Dooling Robert
Published: April 2018

The temporal fine structure (TFS) of an acoustic signal refers to the smaller changes within the time waveform, independent of larger envelope changes. Bioacousticians typically have relied on sonographic analyses to describe the primary acoustic features of animal vocalizations and human speech. However, sonographic analyses do not reveal the temporal fine structure in complex vocal signals. Recent research on the perception of human speech by normal-hearing and hearing-impaired listeners has suggested that TFS is important for speech comprehension. Interestingly, birds have much better sensitivity to changes in TFS than do humans, suggesting TFS could also be important for communication in birds. TFS in complex signals is extremely difficult to study in isolation since it occurs along with other acoustic features of complex signals. But, here using a variety of techniques to reduce or eliminate other acoustic features of a complex acoustic signal, we show that zebra finches have particularly good perceptual sensitivity to the natural variation in TFS, and that there are very likely systematic differences in TFS that are unique to various classes of vocal signals. These results suggest that, as with human speech, TFS may be important in avian acoustic communication.
Ultrasonic vocalization production by socially isolated and experienced mice
Authors: Screven Laurel A. , Dent Micheal L.
Published: April 2018

Mice produce ultrasonic vocalizations (USVs) in a wide variety of social situations, including courtship, investigation, and territorial defense. Although production of USVs is believed to be innate, suggesting mice do not need acoustic experience with conspecifics to produce calls, it is possible mice require experience to emit calls in the proper behavioral context. Male mice require social experience with a female to produce USVs in response to olfactory signals present in dirty bedding (e.g., Nyby et al., 1983), and isolated male mice will produce more USVs than socially experienced males (Keesom et al., 2017), however the influence of social experience on vocal production by female mice has yet to be investigated. The present experiment aimed to determine if social isolation or experience with conspecifics influences the vocal repertoire of adult female CBA/CaJ mice. Mouse calls were recorded during exposure to an unknown male or female mouse and analyzed to determine if isolation or experience affected call number, rate, or spectrotemporal complexity. Chronic social isolation did influence the vocal repertoire, suggesting that social experience plays a role in the production of USVs by female mice.
Spatiotemporal clutter filtering methods for in vivo microvessel imaging
Authors: Song Pengfei , Chen Shigao
Published: April 2018

Ultrafast plane wave imaging enables acquisition and accumulation of a large number of Doppler ensembles in a short period of time, which substantially boosts ultrasound Doppler sensitivity by at least an order of magnitude. As a key component, spatiotemporal clutter filtering based on singular value decomposition (SVD) has recently demonstrated superior performance in tissue clutter suppression and microvessel signal extraction over conventional high-pass-filter-based clutter filtering techniques. This presentation introduces several novel SVD clutter filter techniques from our recent work, focusing on addressing the technical challenges (complex tissue motion, high computational cost, and noise) of SVD clutter filter during clinical translations of the ultrafast microvessel imaging technology for human imaging. We present a block-wise adaptive SVD filter method that can robustly threshold singular values on a local level, a real-time feasible SVD clutter filtering method based on partial SVD and randomized spatial downsampling, and a noise equalization and suppression method that can significantly improve microvessel imaging quality at a low cost. In vivo performance of these methods will be demonstrated in different organs.
The design of broadband constant beamwidth and constant beam pattern transducers
Authors: Huang Dehua
Published: April 2018

The broadband constant beamwidth (CBT) and the constant beam pattern (CBP) transducers attracted much attentions, because of their main lobes or even the whole beam patterns (CBP case) are independent of frequencies. For a spherical (or hemispherical) design, the far field angular beam patterns are the same as the normal directional radial particle velocity, or shading, distribution on the surface of the spherical transducer, if the ka product for the wave number and the sphere radius is greater than one, per spherical Hankel function asymptotic approximation to the solution of the Helmholtz wave equation. Any arbitrary velocity shading functions can be expanded by series of Legendre polynomials for optimization beamwidth and sidelobe levels. This paper introduced a new transducer or array design, where the methodology by CBP is employed, while the geometry of CBT spherical cap is applied, such that the shading functions are no longer confined by only one Legendre polynomial P(cos(θ)) of the CBT case. Here, the shading examples by various Legendre polynomials of P(z0*cos(θ)) and the classic Dolph-Chebyshev T(z1*cos(θ)) of different orders have been successfully simulated, where the z0 and the z1 are the design control parameters of the beams.
A review of single crystal underwater transducers
Authors: Robinson Harold
Published: April 2018

The unique material properties of lead magnesium niobate-lead titanate (PMN-PT) single crystals enable compact, broadband, high power sound projectors that cannot be realized using conventional ceramics. This paper shall highlight single crystal transducer prototypes developed at the Naval Undersea Warfare Center, Division Newport, and elsewhere demonstrating that the broadband, high power performance promised by single crystal’s material properties can be realized in practical underwater transducers. Various single crystal transducer designs for underwater applications, including longitudinal vibrators, fully active and active-passive air-backed cylinder transducers, free-flooded ring transducers and bender bar transducers, will be shown. These transducer designs cover a wide span of frequencies, and utilize both 33- and 32- operating modes. Tuned bandwidths of up to 2.5 octaves of frequency, as well as source level increases of up to 15 dB at the band edges, will be demonstrated. t will also be shown that these technologies provide stable performance under high drive and duty cycle conditions. Finally, the impact of using these single crystal sources on the drive electronics and power consumption shall also be discussed.
An overview of the Standards Program of the Acoustical Society of America
Authors: Struck Christopher J.
Published: April 2018

An overview of the standards program of the Acoustical Society of America is presented. The American National Standards Institute accreditation and the U.S. voluntary consensus process are described. The structure of the ASA standards organization is explained and the roles of each standards committee are outlined. The standards development process from new work item proposal through final approval is detailed. The relationship between national and international standards bodies is also discussed. Examples of the practical benefits of acoustical standards for both participating organizations and end users are given. Information about how to participate in the standards process is also provided.
Laboratory accreditation—Building confidence on acoustical testing- and calibration-lab quality
Authors: Peppin Richard J. , Anastasopoulos George , Ramakrishnan Prasanth S.
Published: April 2018

Testing (and calibration) laboratories, to be good, must provide consistent accurate and precise measurements of the test objects based on their well-defined procedures. Lab accreditation is one way to assure the engineers, scientists, and technicians in the lab AND the lab customers and the public have confidence in the skills, facilities, and results of the lab’s work. There is an international standard, ISO/IEC 17025-2005 “General requirements for the competence of testing and calibration laboratories.” This standard is for use by laboratories in developing their management system, for quality, administrative, and technical operations Following this standard will help labs assure quality in their work. To assure the public (and the lab’s customers) that the lab does follow this, an objective agency evaluates the lab’s quality program. This process is called “accreditation.” Accreditation is done under strict quality control by accrediting agencies, of which there are several available for worldwide labs. Most of the presentation will emphasize what is required and involved in the accreditation process This presentation will also discuss (1) why accreditation is useful, (2) who does the accreditation, (3) how are they qualified to do accreditation, (4) how costs are determined, and (5) how fast can it be accomplished.
Testing the limits of reverberation room qualification standard AHRI 220
Authors: Knight Derrick P.
Published: April 2018

Trane’s acoustic lab recently upgraded measurement equipment which necessitated re-qualification of two reverberation chambers according to AHRI 220. We took this opportunity to test restrictions listed in the standard in relation to source-to-wall spacing and minimum number of fixed microphone positions. Measurements were simultaneously taken using rotating and fixed microphones. It will be shown that the wall proximity limit is overly restrictive, while the minimum fixed microphone requirement is appropriate.
Art in Acoustics
Authors: Gossweiler Peter Francis C.
Published: April 2018

Since the first time the caveman shouted out loud to the heart of a cave, to listen for a kind of magical relationship, a huge spectrum of possibilities has grown to deal with this phenomenon of nature. In fact, the eco or even the reverberation has been fascinating the human being till nowadays, as we can see on noise control studies, acoustics studies, sound studies, etc. But we propose another approach, maybe not so straight related to science (as a demand centered on results) but with an undeniable interest with its capacities of creation through the study of the acoustics (with a demand focused on the process). From artistic works of John Cage, Alvin Lucier, Nicolas Collins, Charlemagne Palestine, Llorenç Barber, and Lukas Kühne, we engage an interdiciplinary relation between art and acoustic to clarify this type of magic in our life. A historical, theorical, and critical analysis is applied in this task.
Laryngeal height, modal registers, and modes of phonation contribute jointly to vocal timbre
Authors: Elliott Taffeta M. , Popeil Lisa
Published: April 2018

Timbre encompasses every distinguishing quality of a sound other than its pitch, duration, loudness, and sound location. Musical sound makers exploit contrasts in timbre. The voice organ in particular is capable of an impressive array of vibrational patterns with acoustic consequences perceived as timbral changes. As a first step towards determining the dimensionality of perceptual timbre in a singing human voice, we assessed the relation between, on the one hand, production parameters, executed by a well-trained vocalist capable of the articulation control necessary for demonstrations, and on the other hand, the complex spectral and temporal modulations in the acoustic signal of the resulting 50 timbral variants. Production parameters on a high front vowel included 5 laryngeal heights (from the lowest possible to a mostly elevated position), 4 modes of phonation (clean, blowy, breathy, and pressed), and several changes to resonance (nasal, bright, and muted) as well as vocal register differences comparing vocal fry and modal registers. Raised and lowered larynx positions alter formants in a regular manner that overlaps with phonation patterns and other articulatory alterations of resonance.
Computational analysis of the Bundengan, an endangered musical instrument from Indonesia
Authors: Kusumaningtyas Indraswari , Parikesit Gea O.
Published: April 2018

The Bundengan, an endangered Indonesian musical instrument, was first developed and played by duck herders. The dome-shaped resonator, which also protects the herders from adverse weather, is made from a specially woven grid of bamboo splits, covered with layers of bamboo sheaths, held in place with sugar palm fibre straps. Inside the resonator are a set of long, thin bamboo plates and a set of strings equipped with small bamboo clips, which vibrate together with the strings. When playing, the Bundengan player uses one hand to pluck the clipped strings and the other to pluck the bamboo plates. The clipped strings generate metal-like sounds, while the bamboo plates generate drum-like sounds. Hence, the Bundengan can imitate the sound of gamelan, a traditional Indonesian instrumental ensemble. Bundengan players report difficulties to tune this instrument and to keep it in tune for a long time. We develop simulations to help Bundengan musicians better understand this instrument. The first simulation quantitatively predicts how the bamboo clips influence the frequencies of the clipped strings. The second one quantitatively predicts how the shape of the resonator affects the soundscape of the Bundengan. These simulations also allow for a more precise procedure to build the instrument.
Fit-testing systems for hearing protectors: ANSI standards to the rescue
Authors: Fackler Cameron J. , Berger Elliott H.
Published: April 2018

To standardize the performance of fit-testing systems for hearing protection devices (HPDs), an ANSI/ASA standard has been in development since 2008. As of this writing, BSR/ASA S12.71 “Performance criteria for systems that estimate the attenuation of passive hearing protectors for individual users” has been submitted for public comment and ballot. This standard refers to fit-testing systems as field attenuation estimation systems (FAESs). In this talk, we introduce important FAES concepts, as handled in the draft S12.71. There are two types of FAESs: physical systems that take an objective measurement and psychophysical systems that rely on subjective responses from the user being tested. FAESs may produce a quantitative output as a personal attenuation rating (PAR) or simply indicate pass/fail for a given HPD fit. For all types of systems, S12.71 specifies methods to assess the quality of the attenuation estimates, by comparing the FAES results to a laboratory real-ear attenuation at threshold (REAT) measurement. Other important considerations covered in S12.71 include the maximum levels of ambient noise in which FAESs may be operated, the incorporation of uncertainties due to HPD fitting and spectral variability in users’ noise exposures, calibration intervals, and information that must be provided to users.
A field microphone in real ear hearing protector attenuation measurement system
Authors: Michael Kevin
Published: April 2018

A Field-Microphone in Real Ear (F-MIRE) hearing protector Field Attenuation Estimation System (FAES) was developed for individual or group testing of any type or style of hearing protector device (HPD). Both occluded and unoccluded ear canal stimulus level measurements are made with a tiny Micro Electro-Mechanical System (MEMS) microphone mounted on a 2-mil thick flat flexible cable. The microphone is mounted inside of a conventional foam ear dam to protect the microphone and the ear canal and to facilitate insertion into the ear canal. The presence of the cable in the ear canal was demonstrated to have a negligible effect on attenuation measurements for both muff and insert-type HPDs. The system measures HPD attenuation in 1/1 or 1/3 octave bands and it is capable of simultaneously measuring HPD attenuation on up to eight human test subjects. Stimuli can be presented via headphones or loudspeakers. Human subject testing using both the laboratory based ANSI S12.6-2016 method and the F-MIRE system was performed using muff- and insert- type HPDs. Regression equations were generated from these data. These equations are incorporated in the F-MIRE software to allow valid comparisons between F-MIRE data and labeled HPD attenuation data.
Fit testing of a hearing protection device with integrated in-ear noise exposure monitoring
Authors: Smalt Christopher J. , Davis Shakti K. , Murphy William J. , Kardous Chucri A. , Lacirignola Joe , Calamia Paul
Published: April 2018

In-the-ear hearing protectors are often used in high-noise environments, such as in military operations and during weapons training. The fit of such a hearing protection device in the ear canal can cause significant variability in the noise dose experienced, an important factor in characterizing the auditory health risk. Our approach for improved dose estimates under these conditions involves a portable noise recorder used to capture in-the-ear noise behind a hearing protector, and on-body noise, while assessing hearing protection fit throughout recording. In this presentation we describe evaluation of this system using ANSI S12.42 testing using a shock tube and an acoustic test fixture, to evaluate impulse peak reduction and for measurement validation. Also explored were angle dependent effects on the peak insertion loss and measurement accuracy of the on-body recorder. An exploratory study was conducted with a small sample of experimenters during a recent Navy-sponsored noise survey conducted at Marine Corps Base Quantico. Results will be presented from a shooter’s ear and a nominal instructor’s position as well as from bystanders observing at a firing range. Our results show good correspondence between behavioral fit-testing of insertion loss and estimated protection from the in-ear and on-body microphones. [Work supported by ONR.]
Acoustical corrections to be used for improved in-ear noise dosimetry measurements
Authors: Bonnet Fabien , Nélisse Hugues , Voix Jeremie
Published: April 2018

Although much effort has been made to reduce sound at the source, hearing protection devices (HPDs) remain currently the most widely used solution against occupational noise exposure in an attempt to prevent noise-induced hearing loss. Such devices, though, often suffer from uncertainties when it comes to ensuring ideal protection for a given individual. These uncertainties stem mainly from three unknown variables: i) the actual attenuation provided by the HPD to the individual; ii) the ambient noise levels; iii) the wear time of the given HPD. In-ear dosimetry (IED) could overcome these issues, as it makes it possible to establish precise personal noise exposures through real-time in-ear noise monitoring. Furthermore, IEDs can be worn in the open, occluded (under earplugs), or cushioned (under earmuffs) ear. Nevertheless, current IEDs are weakened by two inaccuracies: the non-discrimination of self-induced noise, and the lack of acoustical corrections to convert the measured in-ear sound pressure levels to the eardrum and to the free-field. The latter issue is treated in this presentation, which describes methods to identify the appropriate acoustical corrections to be used for improved IED. Such methods aim at personal calibration procedures, so that the unique ear geometry of each individual is accounted for. Preliminary results will be presented as a comparison between theoretical and experimentally measured corrections on human test-subjects.
Application of a registration method on magnetic resonance images to evaluate the displacement field of a human subject ear canal due to various earplug insertions
Authors: Benacchio Simon , Doutres Olivier , Varoquaux Arthur , Wagnac Eric , Le Troter Arnaud , Callot Virginie , Sgard Franck C.
Published: April 2018

Earplugs are a usual way to protect workers subjected to noise exposure. However, the efficiency of these hearing protection devices is often affected by induced discomforts. A factor suspected to impact both acoustical and physiological comfort attributes of earplugs is the deformation they apply on the ear canal walls. As the geometry of both open and occluded ear canal is difficult to obtain, the ear canal deformation due to earplug insertion is not trivial to evaluate. Current medical imaging techniques and image post-processing methods are promising tools to investigate this deformation. In a previous study of the authors, an approach using registration methods on medical images had been proposed to estimate the ear canal displacement field induced by earplug insertion. This approach had been validated in the case of computed tomography scans of a human-like artificial ear occluded by a controlled-shape custom molded earplug. In the present study, this approach is used to evaluate the ear canal displacement of a human subject for various earplug insertions (foam, pre-molded and custom made) in the case of magnetic resonance images. The computed displacement field shows noteworthy differences between each earplug and gives information on how and where the occluded ear canal deforms.
Experimentally derived turbulence filters applied to low amplitude sonic booms
Authors: Klos Jacob
Published: April 2018

Turbulence effects on low amplitude sonic booms are not currently well understood. In 2011, NASA funded an experiment to collect data using a large linear microphone array, consisting of 81 ground placed microphones spaced 125-feet apart, as a supersonic F-18B aircraft flew over the array. Most of the measurements were made during aircraft maneuvers designed to place a focus boom on the microphones. However, a few flights were straight and level passes of the aircraft over the array. Data from two of these level passes were used to estimate filter functions that describe the variation at each microphone relative to the mean response across the array. Two approaches were implemented, one is frequency domain and the second is time domain. The resulting filters can be applied to other non-turbulent boom waveforms to estimate the changes in either the spectrum or time domain signature due to the variations observed during these flights. Variation in loudness level both outdoors and indoors when these filters are applied to predicted signatures is presented. Both outdoors and indoors, the standard deviation of Perceived Level is only slightly smaller for low boom waveforms in comparison to N-waves.
Time-domain modeling of the wind effects on the nonlinearity and absorption of sonic booms
Authors: Lonzaga Joel B.
Published: April 2018

This paper discusses the derivation and solution, in the time domain, of a Burgers’ equation for sonic boom propagation in a windy atmosphere. As the wind effects become important with increasing wind speed, the proper treatment of wind is necessary for sonic booms propagating through a jet stream as well as for those propagating to the stratosphere or thermosphere where stronger winds typically exist. The effects of wind are quantified in terms of the Doppler shift and of a parameter which measures the strength of the convection. The latter, referred to as the convective index, is a ratio of the local sound speed to the magnitude of the ray velocity and is unity in the absence of wind. Both Doppler shift and convective index are larger than unity if there is a component of the wind velocity opposite to the direction of the acoustic propagation. Consequently, the effective nonlinearity coefficient and effective absorption coefficients are larger for rays propagating opposite the wind direction than for rays propagating in the same direction. To demonstrate the significance of the derived Burgers’ equation, the effect of a jet stream with a typical strength and profile on nominal sonic booms will be described.
Temporal integration and multiple looks, revisited
Authors: Wakefield Gregory H.
Published: April 2018

Recent advances in Kullback-Leibler divergence and its generalizations have led to a renewed interest in the modeling of various detection and classification problems. Building from the concept of multiple looks, this paper considers adaptive detector and classifier structures that support stable, multi-stream perception. With respect to these structures, the phenomena of temporal integration and the distinctions between energetic and informational masking are discussed. [Work supported by ONR N00014-16-1-2560.]
Amplitude modulation detection and modulation masking in school-age children and adults
Authors: Buss Emily , Grose John , Leibold Lori
Published: April 2018

Temporal resolution is often evaluated by measuring sensitivity to amplitude modulation (AM) as a function of modulation rate. Published data for AM detection on a noise carrier indicate that school-age children perform more poorly than adults, but thresholds increase with increasing AM rates above 50 Hz in a parallel fashion for both age groups. This result has been interpreted as reflecting adult-like temporal resolution in school-age children. However, this interpretation is complicated by the observation that inherent AM of the noise carrier can elevate AM detection thresholds in adults via modulation masking. Two studies with 5- to 11-year-olds and adults were carried out to better understand the development of AM detection with and without modulation masking. The first estimated thresholds for detecting 16-, 64- or 256-Hz sinusoidal AM on a 4.3-kHz carrier, a condition without modulation masking. Age effects were larger than previously observed with a noise carrier. The second study manipulated modulation masking by measuring detection of 16-Hz sinusoidal AM on a bandpass noise carrier (1.5—2.5 kHz), with and without additional masker AM at nominal rates of 6.3, 16, or 40.3 Hz. All listeners exhibited on-frequency modulation masking, with similar tuning to modulation rate for children and adults.
Auditory enhancement and other context effects in normal, impaired, and electric hearing
Authors: Oxenham Andrew J. , Kreft Heather A. , Feng Lei
Published: April 2018

Neal Viemeister’s contributions to our understanding of auditory enhancement stood out because they combined novel and intriguing empirical data with a testable theoretical framework to explain the results. Thirty-five years later, the resultant “adaptation of inhibition” hypothesis remains the default explanation for auditory enhancement effects, and it can be used to account for both basic auditory context effects as well as speech context effects. Recent work in our lab has focused on comparing speech and non-speech context effects in people with normal and impaired hearing, as well as cochlear implants, in an attempt to elucidate the underlying neural mechanisms and to work towards compensating for any loss of context effects in clinical populations via signal processing. This presentation will provide a survey of recent progress in this area with examples from simultaneous masking, forward masking, and speech discrimination experiments. [Work supported by NIH grant R01DC012262.]