bluuebunny/crossref_metadata_2025 huggingface dataset:
Effects of presentation method and duration on alarm detection threshold in the presence of loud pink noise
Authors: Karunarathne Buddhika , So Richard , Kam Anna
Published: May 2014

Detection of pure tone signals in the presence of noise has been thoroughly studied. Most of these studies have used monaural presentation of audio stimuli. Also, studies testing alarm detection in the presence of noise are limited. In 2013, Karunarathne et al., conducted a study and found out that human listeners were able to detect an alarm in negative signal-to-noise ratios (SNRs), as low as −24 dB. This study aims to investigate the effects of presentation method and duration of the alarms on detection threshold. Eight conditions varied by presentation method (monaural vs spatial) and alarm duration were tested. Sixteen human subjects with normal hearing were given the task of identifying which one of two sound intervals contained an alarm along with 80dBA pink noise. Thresholds were estimated as the 79.4% points on the psychometric functions, using adaptive 2-Interval Forced Choice (2IFC) procedure with a 3-down 1-up rule. Results indicated that detection thresholds were statistically significantly lower in spatial condition compared to monaural. The effect of alarm duration was not significant in both spatial and monaural conditions. Thresholds lower than −30 dB SNR were observed in the spatial condition, which agreed with the findings of Karunarathne et al. and further extended threshold boundaries.
Could binaural listening help segregate competing sound sources?
Authors: David Marion , Grimault Nicolas , Lavandier Mathieu
Published: May 2014

When a sound is alternatively played from two positions, the resulting sequence at one ear is an alternate of two spectrally different sounds, due to head coloration. A previous study showed that these monaural spectral differences can induce segregation. Here, binaural cues are introduced to investigate whether they could strengthen segregation. A rhythmic discrimination task evaluated obligatory streaming with speech-shaped noises. In a first experiment, head-related transfer-functions were modified to introduce independently the interaural time and level differences (ITD and ILD). The results suggested that both ITD and ILD favored segregation. Moreover, the perceptive organization was rather based on the monaural variations in spectrum and intensity at one ear rather than on the spectral interaural differences. Since the binaural cues allow the auditory system to lateralize sounds, a second experiment was intended to determine to which extent the influence of ITD was due to the interaural difference and/or to the resulting associated perceived position. Temporal delays were introduced to simulate different ITDs. The perceived position was modified by manipulating these ITDs independently across frequency. The results, combined with a subjective test of lateralization, showed that both ITD and perceived position influence stream segregation.
Feedback loops in engineering models of binaural listening
Authors: Blauert Jens P. , Kolossa Dorothea , Danés Patrick
Published: May 2014

Hearing models for tasks like auditory scene analysis or sound-quality judgments can run into severe problems when acting in a purely bottom-up, that is, signal driven manner, as they may have to follow up on all possible output options until a final decision has been taken. This may lead to a combinatorial explosion. A way out is the inclusion of top-down, that is, hypothesis-driven processes. In top-down processing, the number of states to be evaluated can be reduced substantially, when the system knows what to look for and thereby focuses attention on states which make sense in a given specific situation. To implement adequate top-down processes, various feedback loops will be included in our models, some more specific, others more general. The general ones originate from the concept that the listener model (“artificial listener”) actively explores acoustic scenes and thereby develops its aural world in an autonomous way. Following this notion, it is attempted to model listeners according to the autonomous-agents paradigm, where agents actively learn and listen. [Work performed in the context of the EU project TWO!EARS,.]
Quantifying the better ear advantage in the presence of interfering speech
Authors: Schoenmaker Esther , van de Par Steven
Published: May 2014

In cocktail party listening with spatially separated speech sources, better ear listening is known to make a major contribution to speech intelligibility. The better ear is generally defined as the ear that receives the highest signal-to-noise ratio (SNR). Usually, this SNR is calculated based on the total length of the signal. However, this seems inappropriate when speech signals are involved since these are highly modulated both in the time and frequency domain. On a perceptual level, modulated maskers give rise to a higher target speech intelligibility than their unmodulated counterparts through the presence of glimpses. A simple measure to quantify the better ear advantage while taking these spectrotemporal fluctuations into account is introduced. In a headphone experiment, three simultaneous sequences of vowel-consonant-vowel utterances were presented at a fixed target-to-masker ratio. The stimuli were rendered with head-related transfer functions and contrasted against stimuli that did not contain any interaural level differences (ILDs) and, as a consequence, allowed no better ear listening. Using the proposed metric, we are able to explain differences in intelligibility for these speech-in-speech mixtures that would remain unexplained by the conventional SNR both for stimuli with and without ILDs.
Concurrent control and plant identification to support overall noise reduction of autonomous underwater vehicles
Authors: Holmes Jason D. , Laferriere Alison B. , Park Christopher G.
Published: May 2014

Commercial, off-the-shelf (COTS) autonomous underwater vehicles (AUVs) are not usually designed to be low-noise in the 100 Hz–10 kHz band. Options to mitigate the noise include mechanical re-design, adaptive filtering of the noise on any acoustic sensor using the vehicle as a platform, or active noise control (ANC). The first of these options eliminates the benefits of using a COTS platform. The second option can be attractive but is most effective for propulsor noise (and not other sources like the depth sounder) and does not mitigate the issue if multiple vehicles are used in coordination. An on-board projector can enable ANC of the vehicle noise. For small, nearly acoustically compact vehicles, the directionality of propulsor noise is not very complex, suggesting global control using a limited set of the acoustic sensors as a control metric. This paper explores methods for performing concurrent control of narrow-band vehicle noise while performing the low-noise, broad-band plant identification necessary to support the control. In addition to the reductions in self noise (on a single vehicle and multiple vehicles), the ability to use the information in the plant estimate to perform “quiet” depth sounding is explored.
Effectiveness of energy finite element analysis applied to submerged undersea vehicle noise prediction
Authors: Jandron Michael A. , Koch Robert M. , Bower Allan F. , Vlahopoulos Nickolas
Published: May 2014

Traditionally, solving structural-acoustics problems has posed significant computational challenges at very high wavenumbers because of the mesh refinement required. As such, instead of resolving each wavelength, the fundamental goal in Energy Finite Element Analysis (EFEA) is to average the energy over many wavelengths. The envelope of average energy behaves as a slowly varying exponential, which is much easier to solve numerically. EFEA thus computes the average energy over a region which can be used to determine undersea vehicle self- and radiated noise in much the same way as conventional FEA modeling but the size of the mesh does not suffer the same restrictions. In this talk, self-noise model predictions for a representative undersea vehicle are discussed to demonstrate the effectiveness of EFEA. This model includes an explicitly modeled acoustic domain to allow radiated energy to propagate in the acoustic medium and possibly reenter the vehicle structure. Results are validated with dense FEA models and the post-processing algorithms used for this purpose are discussed.
Validation of relative fundamental frequency using an aerodynamic estimate of vocal effort
Authors: Lien Yu-An S. , Michener Carolyn M. , Stepp Cara E.
Published: May 2014

Clinical assessment of the voice disorder vocal hyperfunction currently relies on subjective interpretations of clinicians, which can be unreliable, necessitating the development of objective measures. One relatively established objective measure of vocal hyperfunction is the ratio of sound pressure level to subglottal pressure (dB SPL/cm H2O). However, this measure provides unreliable results in individuals with vocal fold lesions, which often accompany vocal hyperfunction. A new acoustic measure, relative fundamental frequency (RFF), defined as the normalization of voiced fundamental frequencies surrounding a voiceless consonant, has also been developed for vocal hyperfunction assessment. Here, RFF was validated against dB SPL/cm H2O in a cohort of healthy speakers simulating various degrees of vocal effort. Twelve healthy participants (M = 22 years; five males) produced stimuli at five levels of effort (ranging from relaxed to maximally strained). The relationship between dB SPL/cm H2O and RFF was determined. RFF parameters were able to explain 37–47% (R2) of the variance seen in the aerodynamic measures when averaged across speakers. For six of the speakers, RFF parameters explained 43–97% of the variance, suggesting that the relationship between RFF and dB SPL/cm H2O may depend on the strategy for modulating vocal effort employed by speakers.
Independent and interacting effects of sentential context and phonological neighborhood structure in spoken word production
Authors: Fox Neal P. , Reilly Megan , Blumstein Sheila E.
Published: May 2014

Models of speech production typically invoke at least two cognitive systems: a structurally static lexicon and a dynamic, compositional system. A speaker must have access to a network of lexical representations that is vested with certain intrinsically associated information (e.g., words' phonological structure), but typical speech involves sequential, online production of words to construct meaningful sentences. Thus, a complete model of speech communication must overlay the lexicon's static architecture with a dynamic, context-driven system. The present study examines effects of these two systems on spoken words' phonetic realizations. Participants produced monosyllabic words beginning with voiceless stop consonants (e.g., coat) that varied in their phonological neighborhood density. Targets were produced aloud after subjects silently read sentence contexts that were either highly predictive (e.g., She sported her stylish new fur...) or relatively neutral (e.g., The artist had trouble drawing the...). Results show that subjects hyperarticulate (i.e., produce with longer voice-onset times) initial stop consonants of words with denser neighborhoods and words that appear in less predictable contexts. Interestingly, the effect of a word's context on its phonetic realization only arises for words in sparse neighborhoods, supporting a model in which the lexicon's static architecture constrains the influence of dynamic processing during production.
Using criterio voice familiarity to augment the accuracy of speaker identification in voice lineups
Authors: Plante-Hébert Julien , Boucher Victor J.
Published: May 2014

Voice familiarity is a principal factor underlying the apparent superiority of human-based vs machine-based speaker identification. Our study evaluates the effects of voice familiarity on speaker identification in voice lineups by using a Familiarity Index that considers (1) recency (the time of last spoken contact), (2) duration of spoken contact, and (3) frequency of spoken contact. Three separate voice-lineups were designed each containing ten male voices with one target voice that was more or less familiar to individual listeners (13 per lineup, n = 39 listeners in all). The stimuli consisted in several verbal expressions varying in length, all of which reflected a similar dialect and the voices presented a similar speaking fundamental frequency to within one semitone. The main results showed high rates of correct target voice identification across lineups (>99%) when listeners were presented with voices that were highly familiar in terms of all three indices of recency, duration of contact, and frequency of contact. Secondary results showed that the length of the verbal stimuli had little impact on identification rates beyond a four-syllable string.
Within-session stability of acoustic features of conversational and clear speech
Authors: Ferguson Sarah H. , Morgan Shae D. , Rogers Lydia R. , Hunter Eric J.
Published: May 2014

In several planned studies, talkers will perform a set of speech tasks several times in each recording session. For example, one study will have talkers perform a set of tasks four times: in quiet, in two levels of noise, and in reverberation. It is unknown, however, whether or how much the acoustic details of speech are affected by simple repetition of the speech task. Over the course of a recording session, talkers' speech might become less careful due to fatigue, boredom, familiarity with the speech materials, or some combination of these factors. Such repetition effects could offset the effects of speaking style instructions given at the beginning of a session. The present study assessed speech acoustic changes over four repetitions of a speech production task set performed under conversational or clear speech instructions. Sixteen talkers performed three speech production tasks (a passage, a list of sentences, and a picture description) four times in each condition. The two speaking styles were recorded in separate test sessions. Several acoustic features relevant to clear speech (e.g., vowel space and speaking rate) will be compared between the first and fourth repetition in each speaking style as well as between the styles.
The scope of boundary lengthening as a function of lexical stress and pitch accent
Authors: Katsika Argyro , Krivokapić Jelena , Mooshammer Christine , Tiede Mark , Goldstein Louis
Published: May 2014

Although the phenomenon of boundary lengthening is well established, the scope of the effect and its interaction with prominence is not well understood. It is known that phrase-final prominence is a determining factor. However, it is unclear whether it is lexical stress or pitch accent that drives the effect, and whether the affected domain is continuous or discontinuous. An electromagnetic articulometer (EMA) study of five speakers of Greek was conducted to examine the effect of (1) boundary (word and IP), (2) stress (ultima, penult, or antepenult), and (3) prominence (accented and de-accented) on the duration of phrase-final word articulatory events. In both accented and de-accented conditions, lengthening affected events that immediately preceded the boundary in stress-final words, but was initiated earlier in words with non-final stress. The affected domain was continuous. The stress effect could also be observed in pausing behavior: pauses following phrase-final words were realized with specific vocal tract configurations, and the articulatory movements forming them were longer when stress was final than when non-final. Based on these results, a theoretical account of boundaries is proposed within the Articulatory Phonology framework, with implications for a cross-linguistic model of prosody. [Work supported by NIH.]
Principal component analysis of formant transitions
Authors: Wennblom-Byrne Jamie , Watson Peter J. , Huang He , Montgomery Erwin
Published: May 2014

Formant transitions represent dynamic articulation processes and are useful to describe differences of articulation (e.g., normal and disordered speakers). Our method, considers the formant frequency change over time as a vector. We first reduce this vector to a single value which is plotted multi-dimensional space. Each dimension represents a data point in time and the position of the point of each dimension is the value of the formant frequency. Once the vector has been converted to a single point in the multi-dimensional space, principal component analysis is applied to reduce the dimensionality. These dimensions can then be applied to multi-variant statistical analysis. Additionally, the eigenvectors in each principal component can be used to plot the differences between sets of formant transitions through time. The eigenvectors are first weighted by calculating the Pythagorean distance in the multidimensional space of the first set of principle components and then each eigenvector is multiplied by the percentage of the variance explained by that principle component. Then, the eigenvectors are plotted and superimposed over the formant transitions. The change of height along the length of the curve represents areas along the transition of maximum and minimum differences between sets of transitions.
Phonetics exercises using the Alvin experiment-control software
Authors: Hillenbrand James
Published: May 2014

A collection of computer exercises was developed for use in teaching phonetic transcription to students taking introductory coursework in phonetics. The exercises were developed using Alvin, a software package for the design and online control of behavioral experiments [Hillenbrand and Gayvert, J. Speech, Lang., Hearing Res. 48, 45–60 (2005)]. The main goal was to allow students to work independently on the routine drill that is needed to learn sound-symbol associations. For example, for a vowel transcription exercise, students hear naturally spoken /hVd/ syllables and are asked to choose among twelve buttons labeled with phonetic symbols. Feedback is provided on each trial, and students have the option of repeating any trials with incorrect responses. Also included are word/phrase transcription exercises in which students hear an utterance and are asked to provide a phonetic transcription. Correct transcriptions are provided following each trial, and a summary of the student's performance is displayed at the end of the exercise. Reverse transcription exercises are also included in which a phonetic transcription is displayed and the student's job is to enter the word or phrase in ordinary orthography.
Towards the computation of a variability index in orofacial muscle activity patterns during speech production using performance verification studies
Authors: Bernadin Shonda , Macpherson Megan , Agramonte Itiel , Udhan Tejal
Published: May 2014

In previous work (Wohlert and Smith, 2002), it was determined that variability in children's speech production is reflected in upper lip muscle activity using electromyographic (EMG) data across repetitions of a phrase. Later studies (MacPherson and Smith, 2013) showed that a lip aperture variability index, which represents the difference in upper lip displacement and lower lip displacement, can also be used as a reliable variability measure from EMG data to determine the effects of multiple repetitions of the same utterance on speech motor production. This study is an extension of previous work and examines the orofacial muscle activity patterns (i.e., EMG data) during speech production in efforts to quantify EMG variability. This information can yield significant insights into how well the speech motor system is functioning in different groups of speakers (e.g., healthy young adults vs. healthy older adults). Initially for this study, kinematic measures of articulatory variability in healthy young adults and healthy older adults were verified using previously developed customized, interactive Matlab programming. The analysis provided the platform for studying the quantification of variability in EMG data. The results of these performance verification analyses are described in this paper.
Tracking four dimensional muscle mechanics from high-resolution and tagged magnetic resonance imaging
Authors: Xing Fangxu , Woo Jonghye , Ziemba Joseph K. , Stone Maureen , Prince Jerry L.
Published: May 2014

Assessment of tongue muscle mechanics during speech helps interpret clinical observations and provides data that can predict optimal surgical outcomes. Magnetic resonance imaging (MRI) is a non-invasive method for imaging the tongue that provides information about anatomy and motion. In this work, we aim to develop a pipeline to track 4D (3D space with time) muscle mechanics in order to measure motion similarities and differences in normal and glossectomy speakers. The pipeline comprises of several modules including super-resolution volume reconstruction of high-resolution MRI (hMRI) and cine-MRI, deformable registration of hMRI with cine-MRI to establish muscle correspondences, tracking tissue points using incompressible motion estimation algorithm (IDEA) from tagged MRI, followed by calculation of muscle mechanics including displacement, rotation, elongation, etc. IDEA is the 3D motion estimated from harmonic phase (HARP) motion that is obtained from tagged MRI. The proposed pipeline was evaluated on five subjects including both normal and glossectomy speakers, yielding accurate tracking results as visually assessed. In addition, we were able to differentiate normal and abnormal muscle mechanics, potentially providing invaluable information for interpreting clinical observations and predicting surgical outcomes.
The path from trilling to frication: Synthesizing Puerto Rican Spanish velar fricative realizations of apical trills
Authors: Llorens Mairym
Published: May 2014

Puerto Rican Spanish (PRS) speakers variably produce velar fricative realizations of etymological apical trills. One way to understand this alternation is by attributing it to the two types of trill realization having different gestural goals in PRS. Alternatively, both types of realization may be the result of invariant gestural goals that give rise to distinct attractor states. This study tests whether invariant articulator tasks specified for PRS apical trills can result in successfully synthesized trill and fricative realizations. Comparative measures present in the acoustics of tokens of both types of realization as recorded by native PRS speakers serve as landmarks to determine the success of the articulator model used to produce the synthesized versions. Manipulation of tongue tip gestural stiffness on the one hand and constriction degree on the other provide a way to test which parameter values best capture a bi-goal articulator task. While these findings do not argue against the possibility that PRS velar fricative realizations arise from gestural goals that differ from their apical trill counterparts, they show that this possibility is not the only candidate account for the alternation exhibited in PRS.
Development of a parametric basis for vocal tract area function representation from a large speech production database
Authors: Lammert Adam , Narayanan Shrikanth
Published: May 2014

Parametric representations of the vocal tract area function have long been of interest in speech production research because of their close relationship with speech acoustics. Many representations are possible depending on which qualities are considered desirable. For instance, several representations have been developed which trade articulatory interpretability for low-dimensionality and other useful mathematical properties, such as orthogonality. It has not been well-established whether these abstract representations are related to inherent constraints on vocal tract deformation (e.g., articulatory subspaces), or whether these representations can achieve high accuracy with a small number of parameters when applied to area functions from large amounts of continuous speech. The present study approaches these issues by extracting a data-driven basis representation for vocal tract area functions, and subsequently demonstrating the relationship between that basis and previously proposed spatial Fourier bases. Analysis was performed on more than a quarter-million individual area functions extracted from the USC-TIMIT database using region-of-interest analysis. Results suggest that a spatial Fourier basis with only four harmonics provides over 90% accuracy, and that one additional basis function corresponding to labial aperture increases representation accuracy to near 100%.
Misparsing coarticulation
Authors: Kingston John , Jesse Alexandra , Rysling Amanda , Moura Robert
Published: May 2014

Considerable evidence shows that listeners often successfully compensate for coarticulation, and parse the speech signal's acoustic properties into their articulatory sources. Our experiments show pervasive misparsing of the acoustic effects of anticipatory coarticulation. Listeners categorized more of a front-back vowel continuum as back before [p] than [t]. A following [p] causes F2 and F3 to fall at the end of preceding vowels, and these listeners treated these falls as information that the vowel was back because F2 and F3 are also lower in back vowels. In experiment 1, misparsing was more extensive in a lax front-back continuum [ɛ-ْ] than in a tense one [e-o], but the lax vowels were also shorter. In experiment 2, the durations of lax and tense vowels were equalized, and the steady-state:transition ratios were varied from the naturally occurring 70:30 to 50:50 and 30:70. Listeners still misparsed consonantal formant transitions more for lax than tense vowels, likely because the acoustic differences between [ɛ] and [ْ] are smaller than those between [e] and [o]. They also misparsed more as transitions lengthened relative to the steady-state, which suggests that they did not adjust for differences in speaking rate that would affect the steady-state:transition ratios.
Nonspeech-nonspeech auditory contrast
Authors: Krug Tracy , Kingston John
Published: May 2014

Many perceptual adjustments for a target speech sound's context have two explanations: listeners could compensate for coarticulation or perceive the target as contrasting auditorily with its context. Only auditory contrast can explain listeners' similar adjustments for non-speech contexts that acoustically resemble the original speech contexts. This experiment tests auditory contrast further by replacing the target sound with non-speech, too. Stimuli consisted of a sequence of two equal ERB fraction spaced tone complexes separated by a short gap. Either the same higher or lower 1 ERB-wide band of tones was amplified in both complexes, creating HH and LL stimuli, or a band 1 ERB higher or lower, creating LH and HL stimuli. Tones were amplified to create HL, LH, HH, and LL stimuli near lower, intermediate, and higher frequencies (1800, 2300, and 2800 Hz). If frequency change within a stimulus evokes auditory contrast, then HL-LH stimulus pairs should be more discriminable than HH-LL pairs, where frequency only changes between stimuli. Listeners discriminated HL-LH stimulus pairs better than HH-LL pairs in all three frequency ranges, as predicted if the complexes contrast within but not between stimuli. They also discriminated both kinds of pairs worse at higher than lower frequencies.
Using lateral sensors in flesh-point tracking of /I/ and /ɹ/ in American English
Authors: Katz William F. , Wang Jun , Mehta Sonya
Published: May 2014

Whereas electromagnetic articulography (EMA) studies commonly use a midsagittal sensor array to record articulatory patterns, higher spatial imaging data (e.g., MRI, ultrasound) suggest that some sounds, such as /ɹ/ and /I/, involve tongue concave/convex shape differences that are more effectively measured along a coronal axis. We therefore explored the use of a lateral sensor in the EMA measurement of liquid consonants. Ten adult subjects produced /I/ and /ɹ/ in /ɑCɑ/, /iCi/, and /uCu/ syllables while seated in a 3D electromagnetic articulography system (AG501, Carstens Medezinelektronic, GmbH). Speech movement was tracked for tongue sensors (tongue tip, TT, tongue lateral, TL, and tongue body, TB) and lips (upper lip, UL, and lower lip, LL). Preliminary results suggest that the TL sensor, taken together with TT and TB, provide an improved characterization of American English liquid consonants. Further results will be discussed in the context of developing methods to optimize real-time speech training and speech rehabilitation systems.
Phonetic realization of Japanese vowel devoicing
Authors: Nielsen Kuniko
Published: May 2014

Despite the widely used term “vowel devoicing”, previous studies have reported that the phonetic realization of Japanese devoicing ranges from devoiced vowel to complete vowel deletion, and is often described as deleted or reduced as opposed to “devoiced” (Beckman, 1982; Keating and Huffman, 1984; Tsuchida, 1997; Kondo, 1997; Maekawa and Kikuchi, 2005, Ogasawara and Warner, 2009). The current study presents an acoustic investigation of phonetic implementation of Japanese vowel devoicing to examine the relative likelihood of vowel devoicing, reduction, and deletion. Twenty-four native speakers of Tokyo Japanese produced 80 words containing high vowels in single devoicing environments. The data revealed that the majority of vowel devoicing tokens (>95%) were phonetically realized as vowel deletions, with no trace of devoiced vowels (= energy excited at frequencies of vowel formants with an aspiration source). Given this result, vowel “deletion” may better characterize the phonetic implementation of Japanese devoicing. Theoretical implications of this finding for Japanese phonological structure will be discussed, including an account that considers mora as the fundamental phonological unit at which devoicing takes place and subsequently views devoicing as a loss of mora sonority.
Review of the role of uncertainties in room acoustics
Authors: Muehleisen Ralph T.
Published: May 2014

While many aspects of room acoustics such as material characterization, acoustic propagation and room interaction, and perception have long been, and continue to be, active areas of room acoustics research, the study of uncertainty in room acoustics has been very limited. Uncertainty pervades the room acoustic problem: there is uncertainty in measurement and characterization of materials, uncertainty in the models used for propagation and room interaction, uncertainty in the measurement of sound within rooms, and uncertainty in perception. Only recently are the standard methods of uncertainty assessment being systematically employed within room acoustics. This paper explains the need for systematic study of uncertainty in room acoustic predictions and review some of the most recent research related to characterizing uncertainty in room acoustics.
Uncertainty and stochastic computations in outdoor sound propagation
Authors: Wilson D. Keith , Pettit Chirs L.
Published: May 2014

Outdoor sound propagation provides an interesting and informative example of uncertainty in acoustics. Predictions are strongly impacted by imperfect knowledge of the atmospheric and ground properties, as well as by random turbulence and unresolved elements of the landscape. This presentation describes the impact of such uncertainties and how they can be quantified with stochastic sampling techniques that are applicable to a wide variety of acoustical problems. Efficient and accurate computational approaches result from simultaneously sampling over frequency, uncertain environmental properties, and random processes. Among the techniques considered are ordinary Monte Carlo and Latin hypercube sampling, importance sampling based on relatively simpler propagation models, and adaptive importance sampling. When uncertainties in the atmospheric and ground properties dominate, importance sampling is found to converge to an accurate estimate with the lowest calculation time. When random turbulent scattering dominates, the sampling method has little impact.
Investigations into the sound absorbing properties of gypsum wall board
Authors: Healey Robert , Butler Kevin , Patrick Ian , Quigley Sean
Published: May 2014

Gypsum wall board is one of the most common materials encountered in buildings and thus encountered in architectural acoustics. Recent research into gypsum wall board sound absorption has indicated that the material may have significantly less absorption than is usually assumed when employed in certain construction assemblies. This paper examines both traditional mountings and a new method for determining sound absorption recently introduced involving mounting a gypsum board assembly in an opening between two reverberant rooms commonly used for transmission loss testing. Sound absorbing data were obtained with the two-room method along with the traditional measurement method, for several gypsum wall board wall assemblies, and with an attempt to compare the measured results to real world experience.
The effects of uncertain scattering coefficients on the reverberation time
Authors: Stephenson Uwe M. , Pohl Alexander
Published: May 2014

Uncertain scattering coefficients are still a weak point in room acoustical computer simulations and predictions. The definition of the scattering coefficient s, their values as well as the simulation model are uncertain. Not only the roughness of infinite walls but also the edge effect are included in a combined diffusivity coefficient ray tracing models utilize. Most interesting in practical room acoustics is the influence on the uncertainty of the reverberation time (RT), which depends very sensitively on s. However, there is no analytical formula for the RT for only partially diffusely reflecting surfaces. In non-diffuse sound fields, the RT depend on the room shape, the distribution of absorption and especially on the scattering coefficients, too. A crucial example is a long rectangular room with totally absorbing side walls and partially scattering front walls. For this case a semi-analytical approach had been found. These and other typical cases in 2D and 3D have now been investigated numerically by the Sound Particle Simulation Method (SPSM) and the Anisotropic Reverberation Model (ARM). The aim is to find semi-analytically formulae or rules to estimate the uncertainty in the prediction of the RT by the Sabine formula. Is an “equivalent scattering area” a useful concept?
Characterization of the uncertainty and error propagation in sound field diffusion measurements
Authors: Jeon Jin Yong , Imran Muhammad , Jang Hyung Suk
Published: May 2014

ISO 3382-1 describes instructions for conducting acoustical measurements in auditoria. This standard defines how to measure and derive general properties of the acoustic conditions. By applying the general instructions in ISO, a modified Np was investigated by counting the number of peaks from room impulse responses in an auditorium to characterize the sound field diffuseness. The accuracy of the parametric value in a space depends on the uncertainties involved in all stages of the measurements and analyzing processes. Uncertainties related to source/receiver characteristics, directional aspects, and computational techniques are considered in computing the modified Np. Therefore, we discuss the extent to which we are uncertain about the modified Np and the factors affecting its accuracy in all processes involved in measurements and computations.
Characteristics of echolocation system in Japanese house bat, Pipistrellus abramus
Authors: Riquimaroux Hiroshi
Published: May 2014

Japanese house bats, Pipistrellus abramus, emit harmonically structured downward FM sweeps for echolocation where the fundamental frequency changes from 80 to 40 kHz. Previous studies with an on-board telemetry microphone revealed that they conduct echo-amplitude compensation to stabilize amplitude of returning echoes. When they hunt preys in the field, emitted pulses are prolonged, and the fundamental frequency little change where the terminal fundamental frequency is fixed at about 40 kHz. About 40% of neurons in the inferior colliculus are tuned best to a frequencies range between 35 and 45 kHz, where higher frequency resolution than other frequency ranges is implied and suited for detecting wing beats of insects. However, their audiogram has not yet been known. The present study constructed their audiograms between 4 and 80 kHz by using the auditory brainstem responses evoked by tone pips. Results show that threshold around 40 kHz is lower than other frequencies except for frequency regions around 20 kHz. Findings suggest that Japanese house bats appear to have high sensitivity around 40 kHz, the terminal frequency, and around 20 kHz, corresponding to their communication frequencies. Characteristics of their cochlear microphonics will be discussed. [Work supported by ONR.]
Origin of the off-axis double-pulse in an echolocating bottlenose dolphin
Authors: Houser Dorian S. , Branstetter Brian , Mulsow Jason , Moore Patrick , Finneran James J.
Published: May 2014

It has been proposed that two pairs of phonic lips might be utilized by a dolphin to adaptively manipulate the frequency content and directionality of the echolocation beam in response to acquired target information. The presence of two pulses appearing off-axis of the echolocation beam and separated in time has been proposed as evidence supporting this hypothesis. An array containing 35 hydrophones was used to measure the beam pattern of a bottlenose dolphin performing a phantom echo-change detection task. Simulated target ranges varied from 2.5 to 80 m and clicks were measured at 5–10° resolution from + /150°. At recording angles beyond + /30°, the click appeared as two distinct pulses that declined in amplitude and distorted as the off-axis angle increased. A simple model utilizing the time difference of arrival for the two pulses was used to compare the direct source-receiver path to one of two source-reflector-receiver paths. Assuming a range of constant sound speeds, distances traveled were compared to a CT scan of the same animal to predict anatomical regions potentially contributing to the second pulse. Results suggest that the second pulse is due to reflections from internal structures of the dolphin head and not a second sound source.
Can you hear me now? Sensitive hearing and limited variation in wild beluga whales (Delphinapterus leucas)
Authors: Mooney T. Aran , Castellote Manuel , Quakenbush Lori , Hobbs Roderick , Goertz Caroline , Gaglione Eric
Published: May 2014

Odontocetes use sound for vital biological functions such as foraging, navigation, and communication, and hearing is considered their primary sensory modality. Yet, hearing abilities within wild species or populations are essentially unknown. Hearing sensitivities and variations can greatly influence biosonar signals and performance. Here we present the hearing abilities of seven wild beluga whales. Data were collected in a capture-release event in Bristol Bay, AK. The goal was to establish the mean audiogram and variation within a subset of animals, defining what wild belugas hear and examining how sound-sensitivities may differ between individuals. Hearing was measured using auditory evoked potentials, from 2 to 150 kHz. All belugas tested could hear up to at least 128 kHz, and two heard up to 150 kHz, exceptionally high for belugas. Regions of “best” hearing (<60 dB) were found between 22 and 100 kHz, showing generally broad sensitivity. Greatest variation (>40 dB) was at regions of best sensitivity and the highest frequencies. This substantial variation was less than that of bottlenose dolphins suggesting differences between populations and the need to collect comparative data on new species. While generally sensitive, the hearing variability suggests that multiple analyses better describe the maximum sensitivity and population variance for odontocetes.
Eigenbeam analysis of the diversity in bat biosonar beampatterns
Authors: Caspers Philip , Leonessa Alexander , Mueller Rolf
Published: May 2014

A quantitative analysis of the interspecific variability in the biosonar beampatterns of bats has been performed on a data set that consisted of 267 emission and reception beampatterns from 98 different species. The beampatterns were aligned using a pairwise optimization framework defined by a rotation for which a cost function is minimized. The cost function was defined by a p-norm computed over all direction and summed across a discrete set of evenly sampled frequencies. For a representative subset of beampatterns, it was found that all pairwise alignments between beampatterns result in a global minimum that fell near the plane bisecting the mean direction of each beampattern and containing the origin. Following alignment, the average beampattern was found to consist of a single lobe that narrowed with increasing frequency. Variability around the average beampattern was analyzed using principle component analysis (PCA) that resulted in “eigenbeams”: The first three “eigenbeams” were found to control the beamwidth of the beampattern across frequency while higher rank eigenbeams accounted for symmetry breaks and changes in lobe direction. Reception and emission beampattern could be differentiated based on their PCA scores using only a small number of eigenbeams.
Temporal dynamics of echolocation during clutter navigation in Eptesicus fuscus
Authors: Wheeler Alyssa , Kloepper Laura , Matsuo Ikuo , Simmons James A.
Published: May 2014

The echolocating big brown bat, Eptesicus fuscus, must both avoid background objects that it may collide with such as vegetation or clutter, and identify insect prey targets. When bats fly through clutter, they emit groups of echolocation sounds in rapid succession called strobe groups. Here, we investigate the limitations of strobe grouping for flight guidance during clutter navigation. We hypothesized that as clutter conditions become extreme, bats will be no longer able to use strobe groups to navigate. We exposed bats to our laboratory flight room cluttered with plastic chains—obstacles that are acoustically similar to vegetation. Bats flew down a corridor 1.0 m wide, 0.7 m wide, or 0.4 m wide, and echolocation sounds were recorded with a 22-microphone array. We used a means clustering analysis to quantify inter-pulse intervals (IPIs) as belonging within a strobe group or between strobe groups. We found that bats made significantly more sounds per flight as the corridor width decreased. Strobe groups in the wide condition contained longer within-strobe IPI, while strobe groups in the narrowest condition had the shortest. We also investigated how many sounds per strobe group were used. These results show a relationship between clutter density and the temporal structure of echolocation calls.
Photoacoustic imaging of brain cortex in rhesus macaques
Authors: Yang Xinmai
Published: May 2014

Functional detection in the brain by photoacoustic (PA) imaging has been an area of interest because of its potential to overcome the limitation of the current available techniques. Both small and large animal brains have been studied by PA imaging. Functional detection in primate brains is particularly of interest because of the similarity between non-human primate brain and human brain and the potential for relevance to a wide range of conditions such as stroke and Parkinson's disease. In this presentation, we show the application of PA imaging technique in detecting functional changes in primary motor cortex of awake rhesus monkeys. Strong increases in PA signal amplitude during forelimb movement indicate an increase in activity in primary motor cortex. The results demonstrate that PA imaging can reliably detect primary motor cortex activation associated with forelimb movement in rhesus macaques. The potentials of PA imaging on clinical human brain imaging will also be discussed.
Transcranial ultrasound hemorrhage detector
Authors: Meral Faik C. , Bennoui Amber B. , Ferro Joleigh V. , Maneval Charles D. , Jafferji Mufaddal A. , Giordano Nicholas J. , Clement Greg T. , White Phillip J.
Published: May 2014

The present state-of-the-art technology in clinical neurosonography requires an acoustic window to achieve a sufficient signal-to-noise ratio (SNR) for detection of intracranial hemorrhage (ICrH). Conventional clinical practices use ultrasound in the range of 2–10 MHz to image and diagnose patients, a frequency range that is too high to produce high-quality transcranial images. This study examined the use of ultrasound frequencies on the order of 0.5 MHz, applied through the skull bone, to detect ICrH. We demonstrated with exvivo human skull specimens, ex vivo animal blood, and soft tissue-mimicking phantoms that a single element ultrasonic transducer can likely be used to detect the presence of ICrH immediately adjacent to the skull bone. The lower frequency design of the transducer together with optimized aperture geometry demonstrated efficient signal transmission through the skull bone with a focal depth optimized for intracranial anatomy. The identification of hemorrhage-brain interfaces was demonstrated with bench-top experiments that mimicked the clinical conditions of acute epi- and subdural hemorrhage. Finally, the device limitations were demonstrated through positive predictive value analysis.
On the use of fast marching methods for transcranial beam focusing
Authors: Wang Tianren , Jing Yun
Published: May 2014

In this talk, we will present our recent studies on the use of fast marching methods for transcranial beam focusing. Three topics will be included: beam focusing for transcranial B-mode imaging, beam focusing for transcranial photoacoustic tomography, and beam focusing of a spherical array for therapy. To correct for the phase aberration from the skull, two critical steps are needed prior to brain imaging or treatment. In the first step, the skull shape and speed of sound are acquired by either CT scans or ultrasound scans. In the second step, fast marching methods are used to compute the phase delay based on the known skull shape and sound speed from the first step, and the computation can be completed in seconds even for 3D problems. The computed phase delays are then used in combination with the conventional delay-and-sum algorithm for generating images. They can also be readily used for transcranial beam focusing for therapeutic purposes. Numerical simulation results will be presented to show the robustness of fast marching methods.
A novel device for guiding ventriculostomy with transcranial ultrasound
Authors: Meral Faik C. , Persaud Michael A. , Silva Aaron E. , Mundra Abhishek , Clement Greg T. , Vosburgh Kirby G. , White Phillip J.
Published: May 2014

A ventriculostomy is often performed to relieve symptoms of emergent hydrocephalus. This involves the placement of an external ventricular drain (EVD) into the cerebral ventricles to remove excess cerebrospinal fluid. Free-handed EVD cannulation results in high rates of misplacement (~50%), leading to an increased risk of iatrogenic complications. Extant technical approaches to improve ventriculostomy guidance are either too complex or inaccurate. We have investigated the possibility of a novel device to guide EVD placement using transcranial ultrasound. The device uses three specifically aligned transducers delivering pulse-echo 0.5-MHz ultrasound through the skull bone to detect and localize the targeted ventricle. It also incorporates a cannula guide that is registered with the ultrasound FOV to integrate guidance with surgery. Results from the design, fabrication, and testing of the prototype device with ex vivo human skulls and brain phantoms will be presented.