bluuebunny/crossref_metadata_2025 huggingface dataset:
Estimating geoacoustic parameters from broadband data from the New Jersey shelf
Authors: Jiang Yong-min , Chapman N. Ross , Badiey Mohsen
Published: October 2013

This paper presents the geoacoustic inversion of broadband data to estimate the sea bed parameters of SWARM95 site. The data set used in this study contains air gun signals that were transmitted at the depths of 25 and 45 m, respectively, below the sea surface, while the source ship was maneuvering at small speed (towing); the signals then were collected at the ranges around 5 km by a vertical line array. A canonical geoacoustic model was generated first according to previous experimental research results, for the purpose of a parameter sensitivity study; then both hybrid optimization and Bayesian inversion techniques were applied to the real data. The data error covariance matrix is estimated to resolve the spatial correlated data error in Bayesian inversion approach. Since the ratio of range and water depth is rather large in this study, only the geoacoustic parameters at the top of the sediment are sensitive in the inversion. The optimization results show the consistency of sensitive geoacoustic/geometric parameter estimations over the ranges and different source depths. Bayesian inversion gives the estimation uncertainties of the geoacoustic properties by marginal probability distributions. [Work supported by ONR.]
Comparison of transmission loss data with predictions based on inverted seabed parameters from the Yellow Sea ’96 experiment
Authors: Wan Lin , Zhou Ji-Xun , Rogers Peter H.
Published: October 2013

Transmission loss is useful in sonar performance prediction. In this paper, the transmission loss data as a function of range, frequency, and propagation direction are obtained from the Yellow Sea ’96 experiment. The experiment was conducted along two radial directions up to 55 km and a quarter of a circle with a radius of 37 km. About 300 broadband explosive sources were detonated above and below the thermocline in the experiment. The seabed acoustic parameters are inverted from the broadband explosive source data. Based on the inverted seabed parameters, the transmission loss is predicted and compared with the experimental data. [Work supported by ONR.]
Modeling sound progation in shallow water including dynamic internal wave fields
Authors: Becker Kyle M.
Published: October 2013

Shallow-water acoustic fields are dominated by effects due to multiple interactions with the bottom. Particularly at low frequencies, the acoustic field can be modeled as a sum of normal modes that satisfy local waveguide properties. For a point source in the water column, the acoustic field is characterized in depth and range by constructive and destructive interference of the propagating modes in range. When including a range-dependent sound-speed field in the water column, such as that caused by internal waves, mode coupling occurs which can impact the observed interference pattern. In previous work, these effects have been considered by modeling acoustic propagation through a moving solibore, or by con- sidering fixed 2D realizations of the sound-speed field. In this work, acoustic fields are generated for the case of a propagating internal wave field. In this case, the modal source function and local normal modes are constantly changing. The impact of these changes on the resulting modal interference pattern is examined for frequencies between 50 and 1000 Hz. In addition, fields are compared as would be measured on fixed versus synthetic aperture horizontal arrays. [Work supported by an ONR Ocean Acoustics Entry-Level Faculty Award.]
Cavitation-enhanced ultrasound heating in vivo: Therapy protocols, mechanisms, and acoustic and MRI monitoring
Authors: Sokka S. D. , Vykhodtseva N. , Hynynen K.
Published: October 2013

Focused ultrasound is currently being developed as a noninvasive thermal ablation technique for benign and cancerous tumors in several organ systems. Although these therapies are designed to ablate tissue purely by thermal means, cavitation can occur. These bubbles can be unpredictable in their timing and location and often interfere with thermal therapies. However, gas bubbles in vivo have some potential useful features for therapy. In this research, we design and test in vivo ultrasound exposures that induce cavitation at appropriate times and take advantage of their absorption enhancing properties while yielding reliable lesion sizes and shapes. In addition, MRI and acoustic methods to monitor and potentially control cavitation induction and the associated therapy are investigated. Finally, histology of the resulting cavitation-enhanced heating lesions is performed to assess the type of tissue damage. We demonstrate that cavitation-enhanced heating can be reliable and useful with the appropriate therapy protocol. Properly timed cavitation during thermal therapy yields larger lesions at lower powers taking advantage of bubbles’ higher absorption. In addition, we show that bubble clouds that result from cavitation can block ultrasound wave propagation, providing a means to protect vital tissues during ablation of nearby pathology.
Modeling the inertial cavitation threshold for tissue-mimicking gels seeded with laser-irradiated gold nanoparticle nuclei
Authors: Wu Tianming , Roy Ronald A. , Holt R. Glynn
Published: October 2013

Previous experiments demonstrated the acoustic cavitation threshold in a tissue-mimicking phantom was lowered from 4.5 to 1 MPa by seeding the optically transparent phantom with light-absorptive gold nanoparticles and irradiating these absorbers with ns-pulsed laser light with radiant exposure less than 10 mJ/cm2. The absorption of laser energy deposits a remarkable amount of heat into the particle and surrounding liquid. Under appropriate acoustic conditions, explosive vaporization occurs and the resultant vapor cavity expands many times its initial size and collapses inertially. A three-stage model is developed to account for prenucleation heating, nucleation and formation of the vapor cavity, and follow-up vapor bubble dynamics. The acoustic pressure and laser-radiant exposure thresholds are determined by examining the radius-time evolution of the cavity. The acoustic threshold decreases when laser exposure increases but the efficiency of light depends on its pulse length and the particle size. A strategy for choosing optimal particles to generate inertial cavitation with acoustic and optic energy inputs is suggested. [Work supported by the Dept. of the Army (Award No. DAMD17-02-2-0014) and the Center for Subsurface Sensing and Imaging Systems (NSF ERC Award No. EEC-9986821).]
Performance of unstructured hybrid Reynolds averaged Navier-Stokes/large-eddy simulation methods in simulating shear-driven cavity flows
Authors: Slomski Joseph , Zoccola Paul , Ebert Michael , Arunajatesan Srinivasan
Published: October 2013

Two separate, unstructured hybrid RANS/LES methods are used to simulate both resonant and nonresonant shear-driven cavity flows. The first method uses an upwind-biased discretization for the inviscid flux calculations in the governing equations, along with a nonlinear k−e turbulence closure for RANS regions, and the Smagorinsky subgrid-scale closure for LES regions. The second method uses an upwind-biased discretization for the inviscid flux terms which is modified to reduce the inherently high dissipation in the associated Riemann solver when applied to cell faces not orthogonal to the flow direction. The second method uses a k−e closure for RANS regions. In LES regions, the second method solves a transport equation for subgrid turbulent kinetic energy, relating this energy to a spectrum for the energy-inertial-dissipation range, which allows calculation of a less dissipative eddy viscosity. Both methods are applied to a three-dimensional, deep cavity problem, at resonant and nonresonant flow conditions. Resulting pressure-time series are compared to experimental measurements.
Reminiscences of Brown University during the 1950s
Authors: Nyborg Wesley L.
Published: October 2013

After World War II, Brown University was known internationally as one of several major centers for fundamental research in physical acoustics, with faculty and students engaged in studying properties of sound fields in gases, liquids, and solids, under linear and nonlinear conditions. Students came to Brown to benefit from this atmosphere and went on to establish their own programs at other institutions. In the Physics Department the emphasis among areas of acoustics changed with time, following nationwide trends. For a few faculty and students, there was increased activity in areas of physical acoustics which were proving to be important to the emerging fields of biophysics and biomedical engineering. These areas included ultrasound field description, ultrasound propagation, radiation pressure, and acoustic streaming, to which major contributions had been made by R. B. Lindsay, A. O. Williams, Jr., R. T. Beyer, and P. J. Westervelt, among others. In this talk, specific examples will be discussed.
Acoustic methods used in condensed matter physics from the mid-1950s and beyond
Authors: Elbaum Charles
Published: October 2013

A brief overview will be presented of various acoustic methods that were used at Brown University to study some aspects of condensed matter physics, in the period from the mid-1950s and beyond. The main emphasis will be directed to studies of dislocation dynamics in the context of mechanical properties (elastic and plastic) of solids, to phase transitions in solids, to metallic glasses, and to properties of liquid and solid helium. document
The Metals Research Laboratory and the development of instrumentation for acoustics research at Brown University
Authors: Chick Bruce
Published: October 2013

People I worked with and instrumentation that was developed during 30+ years at the Metals Research Laboratory at Brown University, and what that original instrumentation has led to.
Professor Peter Westervelt and the parametric array
Authors: Muir Thomas G. , Korman Murray S.
Published: October 2013

Work on the interaction of large-amplitude waves in fluids, leading up to the discovery of the parametric array, is mentioned, starting with Thuras, Jenkins, and O’Neil [‘‘Extraneous frequencies generated in air carrying intense sound waves,’’ J. Acoust. Soc. Am. 6, 173–180 (1935)]. Westervelt’s theory for the scattering of sound by sound [‘‘Scattering of sound by sound,’’ J. Acoust. Soc. Am. 29, 934–935 (1957)], is reviewed. The discovery of arraylike and directivity properties of colinear waves interacting nonlinearly, as remembered by Peter Westervelt (private communication to Muir, 2005), is discussed. The formulation of the parametric array, illustrated from the original manuscript [Westervelt, P. J., ‘‘Parametric acoustic array,’’ J. Acoust. Soc. Am. 35, 535–537 (1963)], is discussed, expounding on its elegance and utility in opening new avenues for research and development, as well exploring its assumptions and limits of applicability. Subsequent benchmark experiments and theoretical mode relaxing the assumptions and extending the range of applicability, are mentioned. The First and Second International Symposia on Nonlinear Acoustics (1968 and 1969) are remembered, photographically, with mention of several discoveries made by the luminaries at that time. Numerous practical applications, developed over the past 45 years, are then briefly illustrated. Westervelt’s mentorship with graduate students at Brown is discussed. [Work in large part, supported by the U.S. Navy.]
Frequency discrimination in amplitude-modulated maskers
Authors: Buss Emily , Hall Joseph W. , Grose John H.
Published: October 2013

Under some conditions, the detection threshold for a pure tone signal masked by a bandpass Gaussian noise can be reduced by the introduction of amplitude modulation (AM) to the masker. Previous data suggest that this advantage may be reduced for discrimination tasks, such as frequency discrimination of a masked tone [Hall et al., J. Acoust. Soc. Am. 102, 1746–1754]. In the present study, frequency discrimination thresholds were estimated for a 500-Hz pure tone presented at a range of levels (3, 6, and 12 dB re: masked detection threshold). The masker was either a Gaussian noise or an AM noise (bandpass filtered 50–4000 Hz), with a range of square wave AM rates (10–40 Hz). Masking release was greater for 10-Hz AM than 40-Hz AM, but frequency discrimination was better for the 10 than the 40-Hz rate when measured at equal sensation levels. This result suggests that temporal resolution and integration may have important contributions to frequency discrimination measured for tones presented in modulated maskers. [Work supported by NIH NIDCD Grant R01 DC000418.]
Responses of cochlear nucleus units recorded from chinchillas trained to discriminate iterated rippled noise from flat-spectrum noise
Authors: Shofner William P.
Published: October 2013

Behavioral responses obtained from chinchillas using a stimulus generalization paradigm show that the perception of pitch strength is dependent on the stimuli used for training. When animals are trained to discriminate a cosine-phase harmonic tone complex (COS) from a flat-spectrum, wideband noise (WBN), generalization gradients indicate that animals rely primarily on temporal information in the envelope. Gradients from animals retrained to discriminate infinitely iterated rippled noise (IIRN) from WBN indicate animals rely more on information in the fine structure. Thus, chinchillas learn to use the information in the fine structure when trained with an appropriate stimulus. In order to begin to examine whether there exists a neural correlate of this learning at the level of the cochlear nucleus, single-unit responses to IIRNs and WBN were recorded from three anesthetized chinchillas that had received considerable behavioral training in discriminating IIRNs from WBN. Preliminary data were obtained for 19 single units consisting of primarylike, chopper, and onset response types. All-order interspike-interval histograms obtained for IIRNs and WBN were qualitatively similar to those obtained from naive, untrained animals for each unit type. Quantitative comparisons among units recorded from trained and untrained animals will be presented. [Work supported by NIDCD R01 DC005596.]
Effects of aging and masker-level variability on backward-masked frequency discrimination
Authors: Espinoza-Varas Blas , Jang Hyun-sook , Jajoria Praveen
Published: October 2013

Backward-masked frequency discrimination thresholds (FDTs) were measured to probe the ability to ignore irrelevant auditory information in young and older adults. For 1500-Hz, 40- or 60-ms, 75-dB-SPL target tones, FDTs were measured unmasked and in three backward-masked conditions in which a 1500-Hz, 80-ms, 75-dB-SPL masking tone followed the target with 20–240-ms ISI. In two masked conditions, there was within-trial irrelevant variability in masker level, with negative or positive polarity (i.e., 75±5 or 10 dB SPL) but not in the third. The no-variability condition yielded more backward masking in older than in young adults. Irrespective of polarity, masker-level variability elevated the FDTs of both age groups, but relative to those of the young, the elders’ elevations were larger and did not decrease with training; elders had difficulty learning to ignore irrelevant variability. In young adults, the decrease in FDTs with training tended to be steeper with negative than with positive level variability. Memory capacity and long-term memory constraints seem to limit the elders’ ability to ignore irrelevant information.
Detection thresholds for amplitude modulation of noise with simultaneous modulating tone
Authors: Pincas Jonathan , Jackson Philip J. B.
Published: October 2013

In speech, the combination of amplitude-modulated (AM) noise and a low-frequency periodic source characterizes voiced fricative sounds. To investigate the modulation’s contribution to fricative auditory quality, AM white noise, with simultaneous sinusoidal component at the modulating frequency, provided stimuli for perceptual tests. Two AM detection-threshold experiments were conducted to establish the effect of varying the relative amplitude and phase of the tone. In the first experiment, tone and noise stimuli were separated within each trial by short pauses; in the second, the tone played continuously throughout the trial. AM detection thresholds when the tone was absent were similar to previously published thresholds for AM noise with no tone (averaging approximately −22 dB). The effect of the quieter tones (10–30 dB tone-to-noise ratio, TNR) was dependent on test condition. In the broken-tone experiment, a slight increment in threshold occurred for all relative phases. The constant-tone experiment revealed impaired detection when tone and noise modulation were in phase, and the converse when in antiphase (5-dB difference in threshold at 30-dB TNR). As TNR increased further (40–50-dB TNR), thresholds rose in all cases. Future work is needed to evaluate how this detectable AM percept influences quality and categorization of fricatives.
Psychophysical and physiological assessment of the representation of high-frequency spectral notches in the auditory nerve
Authors: Alves-Pinto Ana , Palmer Alan R. , Lopez-Poveda Enrique A.
Published: October 2013

High-frequency spectral notches such as those generated by the pinna are useful for sound localization. Because their center frequency (6–8 kHz) is beyond the cutoff of phase locking, the common view is that the internal representation of such notches does not involve the timing of action potentials and that, instead, they must be exclusively represented in terms of auditory nerve rate profiles. Recently, we [A. Alves-Pinto and E. A. Lopez-Poveda, J. Acoust. Soc. Am. 118, 2548 (2005)] have shown that the threshold notch depth for discriminating between a flat-spectrum noise and a noise with a square spectral notch centered at 8 kHz varies nonmonotonically with increasing sound level, and that discrimination is worst for levels around 80 dB SPL. Here, we show that the information in the auditory nerve rate profiles is insufficient to account for this psychophysical result, and that temporal information is almost certainly used for discriminating between noise bursts with and without high-frequency spectral notches.
Individual differences in dynamic binaural-masking-level differences at 500 Hz in listeners older than 50 years
Authors: Akeroyd Michael A.
Published: October 2013

One demonstration of binaural sluggishness is that the binaural masking level difference (BMLD) for a tone burst is strongly dependent upon the rate of modulation of the interaural correlation of a masking noise [D. W. Grantham and F. L. Wightman, ‘‘Detectibility of a pulsed tone in the presence of a masker with time-varying interaural correlation;’’ J. Acoust. Soc. Am. /bf 65, 1509–1517 (1979)]. This paradigm was used to measure such dynamic BMLDs for a group of 18 listeners, aged 50–78 and with low-frequency hearing losses of 7–43 dB. The signal was a 20-ms, 500-Hz tone placed in the temporal center of a 500-ms, 100–2000-Hz, 63-dB overall-level noise. The interaural correlation of the noise was either fixed at 1.0 (static conditions) or 100% modulated (dynamic conditions). As expected, the dynamic BMLD reduced as the modulation rate increased; the across-listener means were 11, 8, 6, 2, 0, and 0 dB at rates of 0.5, 1, 2, 4, 8, and 16 Hz. But the individual differences were substantial: e.g., for the 11 listeners who showed a static BMLD of 15 dB or more, their 2-Hz dynamic BMLDs varied between 2 and 10 dB. Those listeners with poor dynamic thresholds (and so more sluggish than the others) may have increased difficulties in detecting sounds in changing spatial environments.
The role of monaural spectral cues in modeling the precedence effect
Authors: Zahorik Pavel
Published: October 2013

Various evidence suggests that the precedence effect may generalize to spatial locations of sound source and echo that minimize binaural information available to the listener. In these situations, it is likely that monaural spectral cues to the direction of the source play an important role in determining the apparent direction of the source+echo composite. Here a model of directional sound localization that incorporates both binaural temporal cues and monaural spectral cues is described and evaluated. The binaural portion of the model relies on interaural cross-correlation to extract binaural cues to lateral direction. The monaural spectral portion uses a spectral correlation metric to evaluate the similarity of the incoming sound spectrum with a catalog of spectral shapes resulting from a listener’s own head-related transfer functions (HRTFs) to known source directions. The model was used to predict sound localization performance of seven human listeners in 16 spatial configurations of source and echo (presented using virtual auditory space techniques with individualized HRTFs), including six configurations where binaural information was minimized (four on the median plane). In general, precedence effects that relied primarily on spectral cue information were considerably less predictable than those that relied on binaural information. [Work supported by NIDCD.]
Interaural correlation and loudness
Authors: Edmonds Barrie A. , Culling John F.
Published: October 2013

When the same sound is presented to both ears rather than to just one, a phenomenon called binaural summation occurs, in which the overall loudness is greater than when either ear is individually stimulated. The present investigation focused on the effect of interaural correlation on this phenomenon. An adaptive matching paradigm was used to measure the difference in intensity between monotic, diotic, uncorrelated, and anticorrelated bands of noise needed to equate them in loudness. The results for 460–540-Hz bands showed (1) consistent with binaural summation, diotic noise was matched to monotic noise that was 4.6 dB more intense and (2) uncorrelated noise was matched to diotic noise that was 2.0 dB more intense. Anticorrelated and diotic noises did not differ significantly in intensity when matched in loudness. Similar results were observed for 100–900-Hz bands: (1) was extended to 6.5 dB and (2) remained at 2.0 dB, but anticorrelated noise was now matched to diotic noise that was 1.0 dB more intense. For 100–5000-Hz bands results showed (1) was extended to 5.5 dB, (2) was reduced to 1.0 dB, and anticorrelated noise was now matched to diotic noise that was 0.5 dB more intense.
The inclusion of time-varyng jitter in the equalization-cancellation model of binaural processing
Authors: Wan Rui , Durlach N. I. , Colburn H. Steven
Published: October 2013

A more realistic version of the analytically defined equalization-cancellation(EC) model of binaural processing can be simulated in the time domain by introducing time-varying jitters, both in time and amplitude. Although the classic EC model can predict a wide range of binaural-masking-level-difference (BMLD) data in binaural detection, it is more reasonable physiologically to assume that internal noise is time varying, as opposed to the original assumption that the internal time and amplitude errors are different from waveform to waveform but stay constant within a waveform. In the current study, the BMLD predictions for both the analytical model and the simulation are compared for the representative conditions(NoSπ, NπSo, NoSm). It is also proved theoretically that the simulation results converge to the analytical prediction in the NτnSτs case. Differences between the EC model and the simulation are discussed. Applying this simulation to speech perception based on the speech intelligibility index(SII) in anechoic environments predicts binaural advantages consistent with results described in the literature (Zurek,1993). The simulation is also being applied to the speech-transmission-index (STI) prediction in complex environments. [Work supported by NIDCD R01 DC00100.]
Improving listener decision strategy in the perception of complex auditory events
Authors: Flint Jesse , Pastore Richard , Gaston Jeremy
Published: October 2013

Natural auditory events are complex. In perceiving these events, listeners tend to use various, and not necessarily optimal, samplings of information. Prior research examined static and dynamic source and sound characteristics for a posture contrast (upright versus stooped) in human gait, identifying several acoustic cues that sometimes differentiate the posture contrast (e.g., pace, spectral slope and amplitude), but with no one acoustic cue always doing so. Successful listener decision strategy thus needs to be a complex process of attending to and evaluating several cues in a flexible manner. With a long-term goal of determining how to improve listening skills, the current research examines ways to enhance attention to and decisions about specific acoustic cues to the posture distinction. For the stimulus property of pace, both information (or knowledge) of cue relevance and experience making relative decisions about the cue, as opposed to experience with sample stimuli, significantly improve listener decision strategy in making the posture judgment. The current study provides a similar evaluation for the acoustic cue of spectral slope. [Research supported by grant from NSF.]
Effect of decision weights and internal noise on the growth of d′ with N.
Authors: Lenhart Erica , Lutfi Robert A.
Published: October 2013

A general finding of psychoacoustic studies is that detectability d′ of a noisy signal grows less than optimally with the number N of independent observations of the signal. Traditional accounts attribute the nonoptimal growth to a form of central internal noise that accumulates with each observation. A more recent interpretation, however, implicates an upper limit on the number of observations that listeners can process at any one time in a given stimulus. The latter account was clearly favored in the present study by the results of a COSS analysis [Berg ‘‘Observer efficiency and weights in a multiple observation task,’’ J. Acoust. Soc. Am. 88, 149–158 (1990)] applied to the detectability of level increments added to simultaneous and sequential N-tone patterns. Estimates of decision weights showed a predominant reliance of listeners on no more than 2–3 tones in each pattern independent of N. Measures of weighting efficiency systematically decreased with N while noise efficiency remained constant across N. The results are discussed in terms of their implications for the interpretation of early experiments on the effects of the bandwidth-duration product on signal detection. [Research supported by NIDCD Grant 5R01DC1262-14.]
Individual differences in source identification for synthesized impact sounds
Authors: Lutfi Robert A. , Arora Raman
Published: October 2013

Impact sounds were synthesized according to standard textbook equations given for the motion of freely vibrating bars, membranes, and plates. In the two-interval, forced-choice procedure highly practiced listeners identified from these sounds various physical attributes of these sources and the manner in which they were driven to vibrate. A COSS analysis of the trial-by-trial data [Berg, ‘‘Observer efficiency and weights in a multiple observation task,’’ J. Acoust. Soc. Am. 88, 149–158 (1990)] revealed large differences across listeners in the acoustic cues used within each identification task, but similarity in the cues used within listeners across tasks. Only in conditions in which the acoustic information for identification was highly constrained did listeners use identical cues. Despite the large individual differences in listening strategy, identification accuracy was similar across listeners. Where there were differences in identification accuracy the differences appeared large, related to differences in internal noise and not listening strategy. [Research supported by NIDCD Grant 5R01DC006875-01.]
Using multidimensional scaling to determine distances in the perceptual space of rolling sounds
Authors: Stoelinga Christophe , Hermes Dik , Kohlrausch Armin
Published: October 2013

We recorded sounds generated by a ball rolling over a wooden plate. Ball diameter, rolling speed, and plate thickness were varied. In a difference-scaling experiment, the important dimensions of the space spanned by these sounds were determined. All sounds were combined in pairs and subjects were asked to quantify the difference they heard in each pair. In this way all possible perceptual differences could play a role in the difference scalings without a need to interpret these differences in terms of underlying object properties such as size or speed. Using multidimensional scaling, a good fit for the scaled differences was obtained with a minimum of three dimensions. None of the dimensions corresponded directly with one of the three mechanical parameters varied in the rolling sounds. Differences in speed lead to a smaller distance in the thus-obtained perceptual space than differences in size, for the ranges we used. This corresponds to results from experiments where the subjects were directly asked to judge size and speed differences using the same sounds. Remarkably, the plate thickness resulted in large distances in perceptual space, while this mechanical property is poorly recognized when participants are directly asked for it.
A miniature, low-cost, user-friendly personal sound-level dosimeter
Authors: Cheyne Harold A. , von Wiegand Thomas E.
Published: October 2013

The objective of this research is to develop a miniature, low-cost, user-friendly personal sound-level dosimeter that can meet the ANSI S1.25-1991 standard. This dosimeter is intended for people whose work-related or recreation-related activities expose them to noises that cause hearing loss. In contrast to typical dosimeters that are controlled by a manager or clinician, the wearer of the miniature dosimeter will also control it and receive its data output to better appreciate his or her sound exposure dose. To significantly miniaturize the dosimeter and lower its cost as compared to typical dosimeters, it will have a reduced feature set, simplified signal processing, an inexpensive microphone capsule, and a microcontroller rather than a digital signal processor. Preliminary tests, using a Texas Instruments embedded controller and an LCD display, suggest that approximations to the standard dosimeter function blocks can be made while maintaining the ANSI tolerances. Tests with inexpensive microphone capsules show acceptable drift due to temperature and humidity for a dosimeter. Following further testing for compliance with the ANSI standard, a limited set of field tests is planned, with wearers using both the miniature dosimeter and a commercial dosimeter, to compare their dosimetry performance.
Intimacy in performance venues: Towards a socioecological methodology
Authors: Gibbs Bobby , Braasch Jonas , Krueger T.
Published: October 2013

Intimacy can be subjectively defined as the feeling of closeness to performance. Intimacy has been widely correlated with the initial time delay gap, the delay between the direct sound and early reflections in a venue [Beranek, Concert and Opera Halls: How They Sound (Acoustical Society of America, New York, 1996)]. Contemporary research suggests that visual stimuli have considerable bearing upon intimacy perception [e.g., Cabrera and Nguyen, J. Acoust. Soc. Am. 116, 2475 (2004)]. Building upon paradigms of ecological psychology, the author hypothesizes that social affordances, perceived offerings of social propinquity in a venue, will significantly enhance the auditory impression of intimacy. A socioecological approach to weigh the hypothesis is proposed wherein auditory perception is evaluated in the context of both the physical and the social environment. The methodology will employ interviews, ethnographic analyses, field studies, and laboratory tests. The research results will likely have immediate ramifications for the tools of visualization and auralization, its acoustical analogue. The study has broader ramifications for the design process, engendering the possibility of more humane design of artistic listening spaces through understanding reciprocal relationships between physical, acoustical, and social domains. [Work supported by ASA Minority Fellowship and RPI Humanities, Arts, Architecture, and Social Sciences Fellowship.]
How do task instructions influence speech-on-speech masking?
Authors: Helfer Karen S. , Freyman Richard L.
Published: October 2013

This study examined how the nature of instructions to the listener affects masking in speech perception. A new set of stimuli (the TVM sentences) was recorded from three male talkers. These sentences, which are similar to the widely used CRM sentences but are open-set and therefore more difficult, have the format NAME discussed the *** and the *** today, where NAME is Theo, Victor, or Michael and *** is a one- or two-syllable noun used for scoring (e.g., Theo discussed the airplane and the textbook today). On a given trial, three sentences were presented simultaneously (one beginning with each of the three names), each spoken by a different talker. Subjects participated in 6 Name conditions (where they were told to repeat the sentence beginning with a certain name) and in 12 voice conditions (where they were told to listen for a certain voice with which they had been familiarized). Stimuli were presented in both spatially coincident and spatially separated conditions at 3 S/N ratios. Analyses of differences in percent-correct performance among the listening conditions and of error patterns (e.g., errors related to confusion between target voice and masking voices versus errors related to energetic masking) will be shown. [Work supported by NIH DC01625.]
Segregation of vowel sequences having spectral cues reduced using a noise-band vocoder
Authors: Gaudrain Etienne , Grimault Nicolas , Healy Eric W. , Bera Jean-Christophe
Published: October 2013

Previous studies have shown that a reduction in spectral cues can be associated with a deficit in pitch-based sequential segregation. The difficulties experienced by hearing-impaired listeners in cocktail party situations may then be explained in part by a segregation deficit due to the broadening of auditory filters. The reduced speech-recognition abilities of cochlear implant users in noisy environments could also potentially be related to a loss of spectral cues leading to a segregation deficit. The current study was designed to test the ability of listener’s hearing cochlear implant simulations to segregate sequences of vowels having alternating F0 values simulating concurrent voices. The vowels were either unprocessed or processed using noise vocoders having limited numbers of channels. Performance on the segregation task was evaluated using a paradigm in which listeners judged the temporal order of items within the sequence. By relating systematic reductions in the number of channels to performance on the segregations task, the current results have the potential to clarify potential limitations placed upon auditory scene analysis by cochlear implant processing.
A theoretical model of cochlear processing improves spectrally degraded speech perception
Authors: Smith Evan C. , Holt Lori L.
Published: October 2013

Smith and Lewicki, Neural Comp. 17, 19–45, 2005a; Adv. Neural. Inf. Process. Syst. 17, 1289–1296, 2005b; Nature, 439, 7079, 2006, demonstrated that mammalian hearing follows an efficient coding principle (Barlow, Sensory Communications, 217–234, 1961; Atick, Network, 3(2), 213–251, 1992; Simoncelli and Olshausen, Ann. Rev. Neurosci., 24, 1193–1216, 2001; Laughlin and Sejnowski, ‘‘Communications in Neuronal Networks,’’ Science, 301, 1870–1874 2003). Auditory neurons efficiently code for natural sounds in the environment, maximizing information rate while minimizing coding cost (Shannon, Science, 270, 303–304, 1948). Applying the same analysis to speech coding suggests that speech acoustics are optimally adapted to the mammalian auditory code (Smith and Lewicki, Neural Comp. 17, 19–45, 2005a; Adv. Neural. Inf. Process. Syst. 17, 1289–1296, 2005b; Nature, 439, 7079, 2006). The present work applies this efficient coding theory to the problem of speech perception in individuals using cochlear implants (CI), for which there exist vast individual differences in speech perception and spectral resolution (Zeng et al., Auditory Prostheses and Electric Hearing, 20, 1–14, 2004). A machine-learning method for CI filterbank design based on the efficient-coding hypothesis is presented. Further, a pair of experiments to evaluate this approach using noise-excited vocoder speech (Shannon et al., Bell Systems Technical Journal 27, 379–423, 623–656, 1995) is described. Participants’ recognition of continuous speech and isolated syllables is significantly more accurate for speech filtered through the theoretically-motivated efficient-coding filterbank relative to the standard cochleotopic filterbank, particularly for speech transients. These findings offer insight in CI design and provide behavioral evidence for efficient coding in human perception.
Cognitive effects of noise reduction strategies
Authors: Sarampalis Anastasios , Kalluri Sridhar , Edwards Brent W. , Hafter Ervin R.
Published: October 2013

Noise reduction algorithms that attempt to improve speech intelligibility in noise have been used in hearing aids for decades. Yet, the surprising result has been that they are not effective in improving the speech reception threshold (SRT). One explanation is that the artificial processing is redundant, providing the same information as that extracted by the listener’s own auditory system. Our working hypothesis is that natural processing is effective only if there are sufficient cognitive resources to support it and that a reduction in those resources may increase the functional value of the computational algorithm. In the present experiments, subjects performed in a dual-task paradigm in which they listened to and repeated sentences presented in a noisy background while doing a competing visual task, such as a simple driving game. Using driving performance as an indicator of mental effort, we evaluate the effects of different noise reduction algorithms on speech reception. The logic is that, while there will be a loss in performance due to doing two difficult tasks at once, this loss will be smaller when auditory sentences are processed by noise reduction.
Strategies for phonological learning
Authors: Jantzen McNeel G. , Tuller Betty
Published: October 2013

An adult’s ability to perceive speech sounds that do not occur in their native language is influenced by the relationship between the new and native language phonology and by the individual’s capacity to perceive acoustic differences. The study investigated how learning a non-native speech sound occurs over time. First, a perceptual mapping procedure was performed using a synthetic continuum that ranged from the Malayalam voiced, unaspirated, dental stop consonant to the American English alveolar [d]. Subjects identified the stimuli (2AFC) and judged the goodness of stimuli as exemplars of each of the two categories. Next, subjects received 15 sessions (one/day) of 2AFC training with feedback, using natural stimuli produced by native speakers of Malayalam and performed the difference ratings on selected pairs of synthetic stimuli. The perceptual mapping procedure was repeated immediately following training and at two weeks post. Results are consistent with previous work [Case, et al., ‘‘The dynamics of learning to hear new speech sounds,’’ Speech Pathol., Nov. 17 (2003)]. Moreover, comparison of results from the different tasks suggests that some subjects learn the acoustic differences among stimuli, while others treat the stimuli as belonging to two phonological categories. [Work supported by NSF and ONR.]
Automatic perceptual tuning of hearing aids for multidimensional perceptual preferences
Authors: Lin Shih-Hao , Li Pei-Chun , Young Shuenn-Tsong
Published: October 2013

The numerous parameters of modern digital hearing aids and the multi-dimensional perceptual preferences of a subject present the audiologist with a difficult fitting task. A system was designed for integrating various real-world stimuli selected from the questionnaire of the most important hearing aid benefits for people in Taiwan and common fitting guidelines, and an automatic modified simplex procedure to perform efficient search for the optimal preference structure in a subject’s perceptual space. Three experiments were conducted using six normal and six hearing-impaired subjects. Four parameters, the low-frequency gain, high-frequency gain, compression threshold, and compression ratio, were varied to find the desired tonality in the first experiment, the desired Mandarin speech intelligibility in the second, and the overall satisfaction in the third. The results showed that the modified simplex procedure worked efficiently for all cases, however, the test-retest reliabilities of the last two were lower than that of the first experiment. Besides, the preferred parameters were different by comparing the intersubject judgments for the same objective and the within-subject judgments for different objectives. The results indicated that there are multiple peaks in one’s response surface and further studies are required to improve reliability and satisfaction when fitting more complicated parameter sets.
Perception of pitch and prominence in a speech context
Authors: Renato Alejandro C.
Published: October 2013

Different researchers have proposed a dominance region that refers to the band of harmonics whose frequencies determine the perceived pitch. This work examined the different cues appearing in the dominance region in order to discriminate the pitch and the prominence in a speech context. It has been done as a reiterant speech task. Native Spanish speakers were instructed to imitate the intonation and prominence perceived in the stimuli presented. The stimuli were composed by trisyllabic words of the form CV1CV2CV2, where C was the voiced lateral /l/, V1 = /a/, and V2 the five vowels of Spanish /a/, /e/, /i/, /o/, and /u/. The degree of prominence was varied from the first to the last syllable, and the final tone. The stimuli were passband filtered between 250 and 900 Hz, their harmonics extracted and synthesized in three ways: (a) The unresolved harmonic more close to 600 Hz; (b) The three resolved harmonics around 600 Hz; and (c) The two resolved harmonics with larger amplitud by phonetic segment, next to the vowel formant. Results indicate that the stimuli (c) described above presents a better relationship between pitch and prominence discrimination.
Parsing time and rate normalization versus durational contrast
Authors: Brenner-Alsop Eve
Published: October 2013

An acoustic interval is perceived as shorter next to a long neighboring interval than next to a short one. This finding has been attributed to rate normalization (Miller and Liberman, Percept. Psychophys. 25, 457–465, 1979) and to durational contrast (Diehl and Walsh, J. Acous. Soc. Am. 85, 2154–2164, 1989). In the first experiment, we present listeners with a vowel varying in duration from short to long in a context where it could be interpreted as one vowel or two in sentences manipulated to sound as though they were spoken at fast, moderate, and slow rates. We predict that the vowel is more often perceived as two vowels in the fast than the slow sentence. The second experiment tests the effect of parsing the vowel into one versus two vowels on judgments of a [b-w] transition duration continuum and a [b-p] voice onset time continuum. If the vowel is parsed into two vowels, a given transition duration or VOT will sound longer and thus more [w] or [p]-like than if it is parsed into just one. Such a result would reflect durational contrast between the transition duration or VOT and the vowel’s perceived duration. It would not reflect rate normalization directly but only indirectly via its effect on parsing. [Work supported by NIH.]
Sequential contrast or compensation for coarticulation?
Authors: Chambless Della , Kingston John
Published: October 2013

English listeners identify a stop ambiguous between /t/ and /k/ more often as ‘‘k’’ after [s] than [sh] (Mann and Repp, J. Acous. Soc. Am. 69, 548–558, 1981). Judgments shift similarly after a fricative ambiguous between [s] and [sh] when its identity is predictable from a transitional probability bias but perhaps not from a lexical bias (Pitt and McQueen, J. Mem. Lang. 39, 347–370, 1998, cf. Samuel and Pitt, J. Mem. Lang. 48, 416–434, 2003). In replicating these experiments, we add a discrimination task to distinguish between the predictions of competing explanations for these findings: listeners respond ‘‘k’’ more often after [s] because they compensate for the fronting of the stop expected from coarticulation with [s] or because a stop with an F3 onset frequency midway between [t]’s high value and [k]’s low value sounds lower after the [s]’s high-frequency energy concentration. The second explanation predicts listeners will discriminate [s-k] and [sh-t] sequences better than [s-t] and [sh-k] sequences because sequential contrast exaggerates the spectral differences between [s-k]’s high-low intervals and [sh-t]’s low-high intervals and distinguishes them more perceptually than [s-t]’s high-high intervals and [sh-k]’s low-low intervals. Compensation for coarticulation predicts no difference in discriminability between the two pairs because it does not exaggerate differences between the two intervals. [Work supported by NIH.]
Sequential contrast and assimilation in the perception of neighboring vowels and consonants
Authors: Kingston John , Chambless Della , Mash Daniel , Katz Jonah
Published: October 2013

Sequential contrast exaggerates perceptual differences between neighboring segments. Our experiments measure this effect in consonant-vowel-stop and stop-vowel-consonant strings. Listeners identify and discriminate vowels from a front-to-back continuum in next-to-coronal versus labial stops, or stops from a coronal-to-labial continuum in next-to-front versus back vowels. If these intervals contrast sequentially, a vowel midway between front and back should sound backer next to a coronal than a labial stop because a coronal concentrates energy high in the spectrum. Similarly, a stop midway between coronal and labial should sound more labial next to a front vowel, because it too concentrates energy high in the spectrum. Exaggeration of spectral differences would also make a front vowel + labial stop versus a back vowel + coronal stop more discriminable than a front vowel + coronal stop versus a back vowel + labial stop. The strings’ lexical status is manipulated orthogonally via the other consonant to test whether sequential contrast effects are independent of lexical biases. Both identification and discrimination results show independent sequential contrast effects in stop judgments after front versus back vowels, but assimilation in vowel judgments before labial versus coronal stops. The stop-vowel-consonant strings test whether this difference reflects segment order or a consonants-vowel difference. [Work supported by NIH.]