bluuebunny/crossref_metadata_2025 huggingface dataset:
Quantifying kinematic aspects of reduction in a contrasting rate production task
Authors: Tiede Mark , Espy-Wilson Carol Y. , Goldenberg Dolly , Mitra Vikramjit , Nam Hosung , Sivaraman Ganesh
Published: June 2017

Electromagnetic articulometry (EMA) was used to record the 720 phonetically balanced Harvard sentences (IEEE, 1969) from multiple speakers at normal and fast production rates. Participants produced each sentence twice, first at their preferred “normal” speaking rate followed by a “fast” production (for a subset of the sentences two normal rate productions were elicited). They were instructed to produce the “fast” repetition as quickly as possible without making errors. EMA trajectories were obtained at 100 Hz from sensors placed on the tongue, lips, and mandible, corrected for head movement and aligned to the occlusal plane. Synchronized audio was recorded at 22050 Hz. Comparison of normal to fast acoustic durations for paired utterances showed a mean 67% length reduction, and assessed using Mermelstein's method (1975), two fewer syllables on average. A comparison of inflections in vertical jaw movement between paired utterances showed an average of 2.3 fewer syllables. Cross-recurrence analysis of distance maps computed on paired sensor trajectories comparing corresponding normal:normal to normal:fast utterances showed systematically lower determinism and entropy for the cross-rate comparisons, indicating that rate effects on articulator trajectories are not uniform. Examples of rate-related differences in gestural overlap that might account for these differences in predictability will be presented. [Work supported by NSF.]
Physiological correlates of loud speech: Respiratory and intraoral pressure data
Authors: Koenig Laura L. , Fuchs Susanne
Published: June 2017

Many previous studies have investigated how increased loudness affects speech production behavior, but authors have varied widely in the measures they have used, and few studies have systematically assessed relationships among respiratory, laryngeal, and acoustic measures. In this work, we present respiratory and aerodynamic (intraoral pressure) data on eleven German-speaking women who produced speech in regular and loud conditions in three different tasks: Reading short sentences, responding to questions, and producing spontaneous speech. Loudness variation was assessed naturalistically by varying speaker-experimenter distance. Respiratory behavior was assessed using inductance plethysmography, and intraoral pressure was obtained via a pressure transducer affixed to the hard palate. In the respiratory data, we measured inspiratory magnitude as well as the slope of the inspiratory and expiratory phases. In the intraoral pressure data, we searched automatically for the peak pressure value during plosives anterior to the transducer (viz., bilabials and alveolars). These physiological data will be related to previously-presented data on speech acoustics to begin to disentangle respiratory and supraglottal contributions to the characteristics of loud speech.
Phonation threshold pressure and the properties of the vocal tract
Authors: Fulcher Lewis , Lodemeyer Alexander , Kniesburges Stefan , Kaehler George , Doellinger Michael , Becker Stefan
Published: June 2017

In a classic 1988 paper, Titze presented arguments based on the dynamics of the motion of the air through the glottis and its relation to the pressures there to describe how the presence of the vocal tract should affect the phonation threshold pressure. He argued that the action of the intraglottal pressures due to the vocal tract and the motion of the vocal folds would be in phase, and thus the presence of the vocal tract should lower the threshold pressure by an amount that depends upon its inertence. Since the inertence of the vocal tract depends directly upon its length and inversely upon its cross sectional area, these arguments set the stage for quantitative studies of the connection of the geometry of the vocal tract with threshold pressure in both mathematical and physical models. To this end, two sets of experiments were carried out in Erlangen with a physical model of the vocal folds and a vocal tract whose dimensions could be varied. One set of experiments focused on the relationship of the threshold pressure and its frequency with the cross sectional area (areas varied from about 2 cm2 to 12 cm2) and the other addressed the relationship of threshold pressure and its frequency with the length of the vocal tract (lengths varied in increments of 5 cm from about 4 cm to 54 cm). These measurements are compared with calculations done with the surface wave model and those done with a two-mass model.
Sensorimotor adaptation to auditory perturbation of speech is facilitated by noninvasive brain stimulation
Authors: Haenchen Laura , Daliri Ayoub , Dougherty Sara C. , Thurston Emily J. , Chartrove Julia , Perrachione Tyler K. , Guenther Frank H.
Published: June 2017

Repeated exposure to disparity between the motor plan and auditory feedback during speech production results in a proportionate change in the motor system’s response called auditory-motor adaptation. Artificially raising F1 in auditory feedback results in a concomitant decrease in F1 during speech production. Transcranial direct current stimulation (tDCS) can be used to alter neuronal excitability in focal areas of the brain. The present experiment explored the effect of noninvasive brain stimulation applied to the speech premotor cortex on the timing and magnitude of adaptation responses to artificially raised F1 in auditory feedback. Participants (N = 18) completed a speaking task in which they read target words aloud. Participants' speech was processed to raise F1 by 30% and played back to them over headphones in real time. A within-subjects design compared acoustics of participants’ speech while receiving anodal (active) tDCS stimulation versus sham (control) stimulation. Participants' speech showed an increasing magnitude of adaptation of F1 over time during anodal stimulation compared to sham. These results indicate that tDCS can affect behavioral response during auditory-motor adaptation, which may have translational implications for sensorimotor training in speech disorders.
Empirical eigenfunctions as a function of glottal adduction in excised hemilarynx experiments
Authors: Berry David
Published: June 2017

For three excised human male hemilarynxes, vocal fleshpoints and empirical eigenfunctions were computed along the vocal fold surface. For two larynges, an increase in adduction resulted in an increase in lateral and vertical oscillation amplitudes and an improved energy transfer from the airflow to the vocal fold tissues. In contrast, the third larynx exhibited a decrease in oscillation amplitudes. By evaluating the empirical eigenfunctions, this decrease in oscillation amplitudes was associated with an unbalanced oscillation pattern with predominantly lateral amplitudes. These results suggest that adduction facilitates the phonatory process by increasing vibrational amplitudes. However, this relationship holds only when a balanced ratio between the vertical and lateral displacements is maintained. Indeed, it appears that a balanced vertical-lateral oscillation pattern may be more beneficial to phonation than strong periodicity with predominantly lateral vibrations.
Sensitivity and specificity of auditory feedback driven articulatory learning in virtual speech
Authors: Berry Jeffrey J. , Bagin Ramie , Schroeder James , Johnson Michael T.
Published: June 2017

The current work presents articulatory kinematic and acoustic data characterizing how the form of synthesized auditory feedback in virtual speech affects the sensitivity and specificity of articulatory learning. The term “virtual speech” refers to talker-manipulated synthesized speech controlled in real time using electromagnetic articulography (EMA). In the current work, 36 participants (4 with dysarthria) participated in a learning experiment requiring them to control an articulatory speech synthesizer using movements of the tongue, lips, and jaw. Participants were divided among two experimental conditions: (1) an “unmatched” condition, during which all participants received auditory feedback based on common articulatory synthesis settings (neither formant working space nor fundamental frequency were distinguishable between talkers); and (2) a “matched” condition, during which the articulatory synthesis parameters were adjusted to mimic the formant working space and average fundamental frequency of the learner. Analyses focused on the kinematic and acoustic differences in learning between the two conditions. Results suggest that the sensitivity and specificity of articulatory learning is affected by the extent to which the auditory feedback matches the learner's familiar acoustic working space. Findings have implications regarding how the acoustic characteristics of perceived speech affect sensorimotor integration and learning in typical talkers and individuals with dysarthria.
Articulatory reuse in “good-enough” speech production strategies
Authors: Faytak Matthew
Published: June 2017

Given a novel speech motor task, a speaker may optimize execution of the task for accuracy by taking feedback into account, or revert to more habitually used productions which provide a precise output not entirely optimized for accuracy. An articulatory examination comparing L1 and L2 productions was carried out in part to assess the roles of optimization and reversion to habit for individual speakers. Native speakers of American English learning French, (n = 30, 9 males) with a variety of levels of exposure to the L2, were recorded producing the monophthongs of both languages and two English approximants (/r/ and /l/) using ultrasound tongue imaging, video of lip shape, and audio. Principal component analyses run on tongue ultrasound and lip shapes of individual speakers reveal that production of L2 French essentially within L1 English articulatory habit is typical; in several cases, the approximants /r/ and /l/ are essentially reused as French vowels (e.g., /ø/ and /u/). However, a slight optimization toward native-like productions can be observed in speakers with longer exposure to the L2.
Amplitude envelope kinematics of speech: Parameter extraction and applications
Authors: He Lei , Dellwo Volker
Published: June 2017

We model the amplitude envelope of a speech signal as a kinematic system and calculate its basic parameters: displacement, velocity, and acceleration. Such system captures the smoothed amplitude fluctuation pattern over time, illustrating how energy is distributed across the signal. Although the pulmonic air pressure is the primary energy source of speech, the amplitude modulation pattern is largely determined by articulatory behaviors, especially mandible and lip movements. Therefore, there should be a correspondence between signal envelope kinematics and articulator kinematics. Previous research showed that a tremendous amount of speaker idiosyncrasies in articulation existed. Such idiosyncrasies should therefore be reflected in the envelope kinematics as well. From the signal envelope kinematics, it may be possible to infer individual articulatory behaviors. This is particularly useful for forensic phoneticians who usually have no access to articulatory data, and clinical speech pathologists who usually find it difficult to make articulatory measurements in clinical consultations.
Random effects and the evaluation of the uniform scaling hypothesis for vowels
Authors: Nearey Terrance M. , Barreda Santiago , Kiefte Michael , Assmann Peter F.
Published: June 2017

The uniform scaling hypothesis suggests that formant frequencies of vowels of one speaker can accurately predict those of any other speaker of the same dialect by applying a single multiplicative scale factor. Fant (STL QPSR, 2-3, 1-19, 1975) questioned this assumption and presented graphical evidence that scale factors vary between adult men and women for different vowels in ways that are systematically related to their position in the vowel space. However, the issue is complicated by the fact that average vowel datasets may contain multiple sources of variation, including dialect mixture. Statistical evaluation of the uniform scaling hypothesis (or any systematic deviations from it) need to account for multiple sources of variation. In preliminary random-effects analyses of formant data produced by individual speakers from three geographically distinct dialect regions of American English, we found that while relatively modest systematic trends in nonuniformity may exist, their magnitude may be smaller than suggested by earlier work and their assessment may be strongly influenced by other sources of variation within geographical dialect regions. We will present extensions of our preliminary analyses that will include speech data from Texas, Alberta, and Nova Scotia collected in our laboratories.
Speech acoustics can be modulated by cognitive interference in a vowel-modified Stroop task
Authors: Niziolek Caroline A. , Lin Kimberly R. , Beach Sara D. , Quillen Ian A. , Kiran Swathi
Published: June 2017

How are speech acoustics influenced by cognitive processes? In the current study, we used a novel variant of the Stroop test to measure whether the interference between color naming and reading could modulate vowel formant frequencies. Seventeen healthy participants named the color of words in three categories: (1) congruent words (e.g., “red” written in red), (2) color-incongruent words (e.g., “green” written in red), and (3) vowel-incongruent words with phonetic properties that partially matched their color (e.g., “rid” written in red). We hypothesized that the cognitive effort needed to inhibit reading—saying “red,” not “rid”—could affect vowel acoustics. For example, the correct spoken response (“red”) could more acoustically resemble the inhibited word “rid;” alternatively, the vowel could be influenced in the opposite direction, resembling “rad,” which would serve to accentuate the acoustic contrast between the spoken and inhibited words. As expected, participants were slower to produce words on color-incongruent trials than on congruent trials. Interestingly, vowel-incongruent trials were not significantly slower than congruent trials, but preliminary acoustic analyses of the first formant (F1) showed that some subjects systematically modulated their productions in the presence of incongruent vowels. This finding lends insight into how the brain integrates multiple pieces of information to produce speech.
Closed-syllable vowel laxing: A contrast enhancement strategy
Authors: Storme Benjamin
Published: June 2017

Closed-syllable vowel laxing describes the cross-linguistic tendency for high and mid vowels to have higher F1 values and more central F2 values in closed than in open syllables. This pattern is often analyzed as resulting from vowel shortening in closed syllables. However, vowel undershoot does not generally result in an increase of F1 for mid vowels. This paper tests an alternative hypothesis according to which laxing is a strategy to enhance coda-consonant place contrasts, with lower and more central vowels providing more informative closure transitions than higher and more peripheral vowels. Two native French speakers were recorded uttering C1VC2 nonce words with C1, C2={p,t,k} and V={i,y,u,e,ø,o,ɛ,œ,ɔ,a}. 85 English and French hearers were presented with the stimuli without word-final bursts and were asked to identify the place of the word-final consonant. In accordance with the enhancement hypothesis, lowering accompanied by centralizing was found to improve [p]-[k] and [t]-[k] contrasts for front unrounded vowels and [p]-[k], [t]-[k], and [p]-[t] contrasts for back vowels. These contrasts were not systematically more distinct after front rounded vowels than after back and unrounded front vowels with similar F1 values (e.g., [y] vs. [i]/[u]), suggesting that centralizing alone is not sufficient to enhance place contrasts.
Phoneme distribution and phonological processes of orthographic and pronounced phrasal words by syllable structure in the Seoul Corpus
Authors: Yang Byunggon
Published: June 2017

This study examined the phoneme distribution and phonological processes of orthographic and pronounced phrasal words according to syllable structure in the Seoul Corpus of spontaneous speech produced by 40 Korean speakers. To achieve the goal, the phrasal words were extracted from the transcribed label scripts of the Seoul Corpus using Praat. Then, the onsets, peaks, codas, and syllable types of the phrasal words were analyzed using an R script. Results revealed that k0 was most frequently used as an onset in both orthographic and pronounced phrasal words. Also, aa was the most favored vowel in the Korean syllable peak with fewer phonological processes in its pronounced form. For the codas, nn accounted for 34.4% of the total pronounced phrasal words and was the varied form. From syllable type classification of the Corpus, CV appeared to be the most frequent type followed by CVC, V, and VC from the orthographic forms. Overall, the onsets were prevalent in the pronunciation more than the codas. From the results, the author concludes that an analysis of phoneme distribution and phonological processes in light of syllable structure can contribute greatly to the understanding of the phonology of spoken Korean.
A new metric for calculating acoustic dispersion in stop inventories
Authors: Hauser Ivy
Published: June 2017

Dispersion Theory (DT; Liljencrants and Lindblom, 1972) claims that acoustically dispersed vowel inventories should be typologically common. Literature on DT has focused on vowels, where the predictions are robust, and less work has been done on consonants. This paper uses vocal tract model data of stops (as in Schwartz et al. 2012) to extend the predictions of DT to consonants, revealing problems with the conventional method of calculating dispersion. Dispersion is often quantified using triangle area between three category means as points in acoustic space. This approach ignores distributions and reduces entire acoustic categories (which have large variances and different distribution shapes) to single points. Within-category variance is a factor in DT (Lindblom, 1986) and experimental data shows that it affects perception (Clayards 2008), yet conventional dispersion metrics do not take it into account. Here, a new metric based on the Jeffries-Matusita distance is proposed and compared with the more conventional mean to mean distance approach. The incorporation of covariance better reflects human perception, which has implications for considering dispersion in any acoustic space. Nevertheless, this does not recover the predictions of DT, suggesting DT does not apply to consonants and vowels in the same way.
Anatomically oriented Principal Components Analysis of three-dimensional tongue surfaces
Authors: Lulich Steven M. , Nelson Max , de Jong Kenneth , Berkson Kelly
Published: June 2017

A procedure for carrying out Principal Components Analyses of three-dimensional tongue surfaces segmented from 3D/4D volumetric ultrasound images is presented. The segmented surface is transformed to a spherical coordinate system with the origin defined at the anterior visible extreme of the tendon of the genioglossus (near the mandibular symphysis). Principal Components Analyses of tongue surface shapes for monosyllabic real words carried out in this spherical coordinate system are robust to variations in the location of the origin, and show similarities across speakers, based on a corpus of 10 speakers.
Source localization with three-dimensional sound intensity probe with high precision
Authors: Ih Jeong-Guon , Jung In-Jee , Woo Jung-Han
Published: June 2017

When an array module measuring three-dimensional sound intensity is employed for detecting the sound source, a compact space usage and small number of sensors are advantageous than the other source localization methods. However, because of severe bias errors, it has not been popular. We analyze the major sources of bias estimation error and seek for the compensation method. Spectral bias error is due to the reflected signal from the environment, which is proportional to the difference of distance between direct and reflective paths. Spatial bias error is due to the inhomogeneous directivity of the intensity module stemming from discrete arrangement of sensors on the hypothetical sphere surrounding sensors. Simulation with changing the source direction by 1 deg. in spherical angle can generate an error map for all incidence angles. A measurement is conducted using a tetrahedral intensity module with 30 mm spacing for the compensation of errors. Low pass filtering of the cross spectral density function is used for the spectral bias error, and spherical error map is used for the directional bias error. By compensating such bias errors, it is shown that the localization errors of all bearing angles are less than 1° in an anechoic chamber when kd<1.1.
3-D ocean acoustics with normal modes
Authors: Knobles David P.
Published: June 2017

A new derivation is presented for coupled mode equations applicable to 3-D ocean environments possessing strong horizontal variability. The 3-D acoustic field is represented by a bi-orthonormal expansion with both the depth- and azimuthal-dependent eigenfunctions. Two classes of coupling integrals emerge. One class is associated with azimuthal modes and their azimuthal derivatives and the other class is associated with both the depth-dependent and the azimuth-dependent modes and their range and azimuth derivatives. The coupled integral equations for the scattering amplitudes are solved using a method previously developed for a 2-D integral equation coupled mode approach. The method is applied to several 3-D environments including single and multiple seamounts. The effect of neglecting various types of coupling integrals is examined. [Work supported by ONR Code 322 OA.]
Three-dimensional modeling in global acoustic propagation
Authors: Heaney Kevin D.
Published: June 2017

The ocean is nearly transparent for acoustic propagation at low frequencies (<100 Hz), leading to the detection of signals (seismic events, volcanoes, and man-made signals) at distances as large as the ocean basin. Historically, basin acoustic modeling has neglected out-of-plane effects and has been performed with the model computed in the range/depth plane for multiple radials following geodesics (Nx2D). Both oceanographic and bathymetric features can lead to out-of-plane effects. In this paper, a summary of computational approaches to this problem will be presented, including vertical-mode, horizontal ray hybrid approaches, and full-3D Parabolic Equation modeling. Out-of-plane effects include refraction and diffraction—which have different effects as well as different approaches to modeling. Experiments where 3D propagation effects were significant will be presented within this context, including Perth-Bermuda (1960), the Heard Island Feasibility Test (1993), and a recent seismic tomography test off the coast of Japan (2015). Three physics mechanisms will be addressed: horizontal deflection due to mesoscale eddies and fronts, reflection from islands (refraction), and diffraction behind bathymetric edges.
Broadband acoustic wave propagation in three-dimensional shallow waveguide with variable sound speed profile and boundary roughness
Authors: Badiey Mohsen
Published: June 2017

Propagation of broadband acoustic signals in shallow water environment is a complex four-dimensional problem that needs to be addressed with input from the spatial and temporal physical parameters of waveguides with rough boundaries. To construct a numerical model of this four-dimensional problem, methods such as the Parabolic Equation (PE), Horizontal Rays and Vertical Modes, 3D Ray Method, and Nx2D PE have been utilized in recent years. However, data/model comparison still remains a challenge and accurate comparison between measured and modeled acoustic fields in the waveguide is badly needed. Lack of environmental input to use for modeling is one reason, but with proper sampling of environment it may be overcome by novel experimental design based on the knowledge of waveguide physics. Acoustic frequency can also be utilized as one of the key parameters to simplify the problem and adopt strategies in conducting calibrated experiments. In this paper, we provide a broad view of recent advancements in three-dimensional acoustic wave propagation in shallow water waveguides in the presence of variable volumetric and boundary conditions. The effect of broadband acoustic wave center frequency and bandwidth with respect to the scale of environmental variability is also discussed. [Work supported by ONR 322 OA.]
Computation of sound field, reflected from ideal non-flat boundary, and reflected and refracted from non-flat boundary of two different media in parabolic approximation
Authors: Maltsev Nikolai
Published: June 2017

Four Euler equations for harmonic sound field ρc2(▽,v) = iωp,▽p = iωpv can be reduced to two equations for sound pressure p and radial velocity vr in cylindrical coordinate system (r,φ,z): ∂p/∂r = iωpvr ∂vr/∂r = i/(ωp)(k2+(1/r2)∂2/∂φ2+∂2/∂z2)p-1/rvr Solutions for this system of equations in a set of 3-D local normal modes is constructed. A stable numerical method was created, for computation of sound field, reflected from ideal non-flat boundary, and reflected and refracted from non-flat boundary of two different medias, in parabolic approximation. Solutions of different model problems are presented.
Three dimensional acoustic parabolic equation based on Chisholm approximation with the splitting denominator
Authors: Lee Keunhwa , Seong Woojae
Published: June 2017

We propose the generalized form of three dimensional acoustic parabolic equation (3DPE) based on Chisholm approximation [Chisholm, Math. Comp. 27, 841-848 (1973)] of a rational approximant for two variables, and the splitting denominator assumption. The proposed form has wider angle accuracy to the inclination angle of ±620 from the range axis of 3DPE at the bearing angle of 450. Moreover, the splitting denominator makes the split-step algorithm with finite-difference depth solver more efficient in that the 3DPE can be easily transformed into the tridiagonal matrices system. One drawback of this method is the increase of the phase error in the evanescent region, but should be practically remedied by several skills. In this study, the comparative study of other PE approximations will be conducted based on the phase error analysis. Also, numerical examples with three dimensional problems will be given for the performance and benchmark test.
Dynamic imaging of a gravity wave caused by laser-induced breakdown in a fluid waveguide using acoustic waves
Authors: van Baarsel Tobias , Roux Philippe , Nicolas Barbara , Mars Jérôme , Bonnel Julien , Arrigoni Michel
Published: June 2017

The dynamic imaging of a gravity wave propagating at the air-water interface is a complex task that requires the sampling of every point at this interface. Using two source-receiver vertical arrays facing each other in a shallow water environment, we manage to isolate and identify each multi-reverberated eigenray that interacts with the air-water interface. The travel-time and amplitude variations of each eigenray are then measured during the crossing of the gravity wave. In this work, we present an ultrasonic experiment in a 1 m-long, 5 cm-deep waveguide. At frequencies in the MHz range, the waveguide transfer matrix is recorded 100 times per second between two source-receiver arrays while a low-amplitude gravity wave is generated by a laser-induced breakdown above the water surface. The breakdown causes a blast wave that interacts with the air-water interface and penetrates into the water, creating ripples at the surface. This event is easily controlled and therefore repeatable. The inversion performed from a few thousand eigenrays leads to accurate imaging of the dynamic of the air-water interface, using either the travel-time or the amplitude variation.
Statistical inference for source localization using multi-frequency machine learning
Authors: Niu Haiqiang , Gerstoft Peter , Reeves Emma
Published: June 2017

As a classification problem in machine learning, source localization is solved by training a feed-forward neural network (FNN) on ocean acoustic data. The FNN is fed with normalized sample covariance matrices (SCMs). The output of network, which represents the probability for range, is used to determine the source ranges. Since it is a data-driven method, no acoustic propagation models are needed. As shipping noise has a broad frequency band, an approach of statistical inference for source localization is presented by taking advantage of multi-frequency information. It is demonstrated by the vertical array data from Noise09 experiment.
Blocked pressure based Transfer Path Analysis (TPA) method to diagnose airborne sound transfer through building partitions
Authors: Patil Nikhilesh , Moorhouse Andy , Elliott Andy S.
Published: June 2017

Airborne sound transmission through building elements or the sound insulation of the building element is usually rated by its Sound Reduction Index (SRI) or the Sound Transmission Class (STC). SRI/STC quantifies the overall sound transfer but gives no information about how the transfer takes place and what are the contributions of different sound transfer paths involved. Such problems are fairly common to the vehicle acoustics industry and are generally tackled by TPA techniques. The paper formulates an in-situ Airborne TPA technique to quantify the contributions of different sound transfer paths to the transmitted pressure. The airborne source is characterized by its blocked pressure and its direct measurement is discussed. Results are presented for dual leaf partitions excited by an airborne source. The method has shown to be significantly faster than the Blocked force based TPA method which relies on inverse measurement methods. The accuracy of the method is closely related to the wavelength of incident airborne waves.
The acoustics of rooms for music rehearsal and performance—The Norwegian approach
Authors: Olsen Jon G. , Rindel Jens Holger
Published: June 2017

Each week, local music groups in Norway use more than 10,000 rooms for rehearsal and concert; many of the rooms are in schools. The size of the rooms vary from under 100 m3 to over 10,000 m3. The users cover a broad variety of music ensembles, mostly wind bands, choirs, and other amateur ensembles. Since 2009, the Norwegian Council for Music Organizations («Norsk musikkråd») has completed more than 500 room acoustical measurement reports on rooms used for rehearsal and concert. The measurements include reverberation time, the strength parameter G, and background noise. All the reports are made available online in a Google Map. The analysis shows that 85% of the rooms do not comply with the Norwegian Standard NS 8178:2014 and are evaluated more or less unsuitable for the purpose for acoustical reasons. The important criteria are volume, room dimensions, reverberation, acoustic treatment of surfaces, and background noise. In particular, the importance of volume is clearly documented. Analysis of room strength indicates that this also is an essential factor for this type of rooms. The systematic collection of acoustic reports gives important background for recommendations on how to build or refurbish rooms for music in schools and cultural buildings.
Sound masking in office environments—Trade-off between masking effect and user acceptance
Authors: Martin Noemi D. , Liebl Andreas
Published: June 2017

Solving cognitive tasks is adversely affected by speech sound (Irrelevant Speech Effect). In particular, the interfering potential of speech sound is high in open-plan office environments. Sound masking is one of the most effective methods to reduce the disturbing speech sound by covering certain fractions of it with a masking signal. Since the most effective masking signals are subjectively perceived as annoying themselves, the goal of this work was to develop a masking signal which fulfills both, a good masking effect and a high user acceptance. To achieve this goal, qualitatively different masking signals with varying structural properties were developed. The efficacy of these signals was tested in a laboratory experiment using a cognitive task (serial recall). In addition, a subjective evaluation of the signals (loudness, annoyance, etc.) was carried out. In the second part of this work, a questionnaire was developed, which can be used to evaluate the acceptance and the subjectively assessed effectiveness of newly developed masking signals by their potential users. Using this questionnaire, the newly developed masking signals were evaluated by persons working in open-plan offices. Findings and implications for practical usage are presented.
Cumulative sound exposure levels—Insights from seismic survey measurements
Authors: Martin Bruce , McDonnell Jeff , broker koen
Published: June 2017

The weighted cumulative sound exposure level from man-made noise sources is often recommended as a method of measuring possible injury to marine life hearing. However, the behavior of this metric over large areas and its evolution in time is poorly documented in the scientific literature. Similarly, the differences in sound exposure levels as a result of changing the frequency weighting functions are only vaguely discussed. In this presentation, we provide insights into the behavior of the SEL metric based on measurements of several seismic surveys and simulations of seismic vessels passing a recording location. Based on the real-world measurements, we show how the range at which the surveys exceed the regulatory thresholds is highly dependent on the weighting functions and threshold values.
Taking into account uncertainties in environmental impact assessment of underwater anthropogenic noise
Authors: Le Courtois Florent , Kinda G. Bazile , Stéphan Yann
Published: June 2017

The management of underwater anthropogenic noise is becoming a significant component of marine policies. Rules to limit noise exposure in a way that marine animals are not adversely affected are expected to be set up in the future. However, modeling or measuring underwater noise levels (orother suitable metrics) can still be subject to a lot of uncertainties due to difficulty in estimatation or measuring key parameters as source pressure levels and waveguide features. On the other hand, the audition threshold values inferred from bioacoustics studies may also lack of statistical robustness and are difficult to generalize. The combination of these two major sources of uncertainties may lead to misestimate the noise exposure risk and may hinder marine space planning efficiency. This work aims at developing a framework for impact studies while considering uncertainties on the acoustic metrics and confidence in the threshold values. It relies on probabilistic description of the acoustics pressure, as proposed in several recent studies. The formulation provides the interpretation of the results in terms ofimpact risk considering exposure time and source distance. Taking intoaccount of the uncertainties becomes then a strong tool for decision support.
Advanced methods for passive acoustic detection, classification, and localization of marine mammals
Authors: Mellinger David K. , Lu Yang , Lending Curtis , Klay Jonathan , Moretti David , Jarvis Susan M. , Baggenstoss Paul M. , Martin Stephen W. , Roch Marie A. , Marsh Christopher A. , Frasier Kaitlin E.
Published: June 2017

For effective long-term passive acoustic monitoring of large data sets, automated algorithms provide the ability to detect, classify, and locate (DCL) marine mammal vocalizations. Several DCL algorithms were developed and enhanced, with emphasis on methods robust to non-Gaussian and non-stationary noise sources. (1) A subspace model was developed to separate odontocete clicks from noise sounds. (2) A multi-class support vector machine was improved by resolving confusion among species’ overlapping-frequency clicks, and a beaked whale buzz class developed. (3) For dolphin whistles, shape-related features, extractable automatically, were shown to carry species-specific information. (4) Equipment and site differences were discovered to affect Gaussian mixture model classifiers, and methods were developed to mitigate these differences. (5) A nearest-neighbor approach to detection association and 3D localization across multiple phones with multiple arrivals was developed (and applied to beaked whales) using time-difference-of-arrival (TDOA) hyperbolic methods, retaining TDOAs with fewer than the usual three detections and using associations between a given phone’s detections with nearest neighbors. (6) Minke whale “boing” frequency estimates were improved to differentiate individuals, and a kinematic tracking algorithm was developed. (7) A generalized-power-law detector for humpback whales was improved. (8) A software interface for detection was developed, then tested by sending data from Ishmael to a detection process in MATLAB.
Organizing metadata from passive acoustic localizations of marine animals
Authors: Roch Marie A. , Miller Philip , Helble Tyler A. , Baumann-Pickering Simone , Širović Ana
Published: June 2017

The Tethys system is a set of schemata designed to organize spatiotemporal data from detection, classification, and localization (DCL) tasks targeting sound producing marine animals. These metadata are based on the analysis of recordings collected through passive acoustic monitoring. Tethys is accompanied by a scientific workbench implementation of the schemata rules. The system is designed to promote the retention and organization of acoustic metadata in a manner that allows long-term retention of detailed DCL outputs. We present recent work on the localization schemata showing the ability to organize localizations from a large naval hydrophone range. The Tethys implementation permits these data to be analyzed in the context of Internet available data products containing oceanographic, atmospheric, and ephemeris data. Moving beyond simple detections, this addition of localizations to Tethys will allow even more powerful interpretation of the spatiotemporal occurrence of marine animals in the context of their ecology.
Analysis of marine mammal bearing tracks from two-hydrophone recordings made with a glider
Authors: Küsel Elizabeth T. , Siderius Martin
Published: June 2017

An underwater glider fitted with two hydrophones recorded approximately 19 hours of data during an opportunistic sea experiment in the summer of 2014. The acoustic data were collected with a sampling frequency of 96 kHz and 16-bit resolution in deep waters off the western coast of the island of Sardinia, Mediterranean Sea. Detection and classification of sounds by a trained human analyst indicated the presence of sperm whale (Physeter macrocephalus) regular clicks as well as dolphin clicks and whistles. A period of 90 min during which the glider did not surface, and which contained extensive sperm whale clicking activity was chosen for analysis. Cross-correlation of the data from both channels allowed the estimation of the direction (bearing) of clicks, and realization of animal tracks. Several bearing tracks were observed through this analysis, closely following the oscillatory pattern of the glider’s heading, suggesting that such information has the potential to break the left-right ambiguity of the bearing estimates. Results from the bearing tracking analysis, including accuracy and performance, will be shown followed by a discussion on how they can aid in population density estimation studies.
A comparison of whale tracking algorithms in the Indian Ocean using the Comprehensive Test Ban Treaty Hydrophone Data
Authors: Lechner David A. , Guan Shane , Vignola Joseph F.
Published: June 2017

This paper will present results comparing several geo-location and tracking algorithms applied to whale signals as monitored by the Comprehensive Test Ban Treaty (CTBT) sensor network. The performance of three tracking algorithms are compared using several hours of acoustic data collected off Cape Leeuwin, Australia, containing the apparent broadcast call of a blue whale (Balaenoptera musculus). The first approach used cross-correlation in time on the entire signal, while the second used a smaller time sample that covers the leading or trailing edge of the signal and creates a motion vector from the pair. The third approach used the Cross-Ambiguity-Function Mapping technique to generate 2-D energy maps on a geographic plot. Accuracy and energy detection algorithms based on blind-processing of the data are compared.
Bat population censusing with passive acoustics
Authors: Kloepper Laura , Fu Yanqing , Ralston Joel
Published: June 2017

Passive acoustic monitoring is a widely used method to identify bat species and determine spatial and temporal activity patterns. One area where acoustic methods have not yet been successfully applied, however, is in determining population counts, especially from roosts. Typically, most roost counts are obtained with thermal imagery that may be prohibitively expensive for many natural resource managers or require complex computer programming. Here, we demonstrate a new acoustic technique to estimate population size of Brazilian free-tailed bats (Tadarida brasiliensis) emerging from large cave colonies. Data were acquired across multiple nights and at 9 cave locations with different roost structures and flight behavior profiles. We used a single microphone to monitor echolocation activity and simultaneously recorded the emerging bats with thermal video. Bat abundance counts were determined from a single video frame analysis (every 10 s) and were compared to different acoustic energy measures of a 1-s long acoustic sequence recorded at the time of the analyzed video frame. For most cave locations, linear regression models successfully predicted bat emergence count based on acoustic intensity of the emerging stream. Here, we describe our method and report on its application for counting bats from different roost locations.
Subtleties in the acoustics of marine sediments
Authors: Buckingham Michael J.
Published: June 2017

Based on the pioneering work of David Farmer, it is now is well recognized that the layer immediately beneath the sea surface is a highly dynamic, two-phase region, where bubbles created by wave breaking form an upward refracting sound speed profile that acts as an acoustic waveguide. The bottom boundary, although less dynamic than the sea surface, possesses its own unique complexities that are no less challenging to understand than those of the near-surface bubble layer. For instance, a growing body of experimental evidence indicates that the acoustic attenuation in a marine sediment obeys a frequency power law, extending over a wide bandwidth, in which the exponent takes a value close to unity. A long-standing problem has been to identify the frequency dispersion in the sound speed associated with such a power-law attenuation. Several solutions to this problem have been proposed over recent decades but are in fact unphysical in that they fail to obey the Kramers-Kronig dispersion relations. An alternative solution that has recently been developed, which overcomes the previous difficulties, contains a number of subtleties that, it is hoped, will appeal to David Farmer’s keen sense of scientific curiosity. [Research supported by ONR.]
Efficacy of up and down-chirps for two-pulse sonar techniques
Authors: Mistry Nikhil , White Paul , Leighton Tim
Published: June 2017

Twin-Inverted Pulse Sonar (TWIPS) and Biased Pulse Summation Sonar (BiaPSS) were inspired by a video of dolphins using bubble nets to hunt. There are a number of animals that have been observed to use the chirp (a sweep in sine waves of shifting frequency) in echolocation. This talk will present the difference in efficacy of the sonar techniques, mentioned above, when using upwards and downwards sweeping chirps as sonar signals. These techniques have been simulated and experimentally tested with a number of bubble size distributions (BSD). This includes simulations using the results of BSD measurements by Farmer and Vagle (1989). Further, another two-pulse technique will be outlined, proposed as a means to overcome the limitation on detection range, owing to the inter-pulse delay in TWIPS and BiaPSS.
Beamforming methods for large aperture imaging
Authors: Trahey Gregg , Bottenus Nick , Pinton Gianmarco
Published: June 2017

Maintaining image quality at large tissue depths remains a clinically significant and unmet challenge for ultrasonic scanners. For tissue structures beyond 10 cm, commonly encountered in obstetric and abdominal scans, diffraction and propagation through tissue can limit azimuthal resolution to be larger than 5 mm and elevation resolution can exceed a few centimeters, making the evaluation of sub-centimeter fetal anatomical features or renal or hepatic lesions very difficult. We describe simulation and ex vivo human tissue studies which evaluate the image quality achievable with large aperture arrays and with associated beamforming methods. We imaged through human abdominal tissue layers and synthetically formed very large coherent apertures (2 cm X 10 cm) apertures. We also performed matched simulations using Visible Human Project-derived tissue models and full-wave simulation code. Using both datasets, we assessed : 1) the image quality improvements attainable with large arrays, 2) the factors degrading the image quality of deep-lying tissues, and 3) the beamforming methods best suited for large array imaging of deep tissues. Our results indicate that large improvements in resolution are obtainable for deep-lying tissues when imaging with large arrays. The major source of tissue-induced image degradation was observed to be clutter due to reverberation and beamforming limitations, rather than phase errors. Coherence-based beamforming methods were seen to be especially applicable in large array imaging.
A Fresnel-inspired approach for steering and focusing of pulsed transmit beams by matrix array transducers
Authors: Verweij Martin D. , Pertijs Michiel A. , de Wit Jos , Fool Fabian , Vos Hendrik J. , de Jong Nico
Published: June 2017

Matrix ultrasound transducers for medical diagnostic purposes are commercially available for a decade. A typical matrix transducer contains 1000 + elements, with a trend towards more and smaller elements. This number renders direct connection of each individual element to an ultrasound machine impractical. Consequently, it is cumbersome to employ traditional focusing and beamforming approaches that are based on transmit and receive signals having an individual time delay for each element. To reduce cable count during receive, one approach is to apply sub-arrays that locally combine the element signals using programmable delay-and-sum hardware, resulting in reduction by a factor 10. In transmit, achieving cable count reduction while keeping focusing and steering capabilities turns problematic once it becomes impossible to locally equip each element with its own high voltage pulser. To overcome this bottleneck for decreasing element size, here we present a Fresnel-inspired hardware and beam forming approach that is based on transmit pulses consisting of several periods of an oscillating waveform. These will be derived from one oscillating high voltage signal by using local switching and timing hardware. To demonstrate the feasibilities of our approach, we will show beam profiles and images for a miniature matrix transducer that we are currently developing.