bluuebunny/crossref_metadata_2025 huggingface dataset:
Sequential streaming under reverberation: Effects of the reverberant tail properties
Authors: Brandewie Eugene J. , Oxenham Andrew J.
Published: May 2015

A physical difference between two alternating stimuli can elicit perceptual segregation when the difference is sufficiently salient. One such cue involves differences in reverberation, potentially caused by differences in source distance. Here, we studied what aspects of difference in reverberation are most important in eliciting segregation using a rhythmic masking task. Two interleaved sequences of Gaussian noise bursts (target and interferer) were presented on each trial and listeners attempted to identify which of two rhythms was presented in the target sequence. The influence of the reverberation tail (or damped decay) was studied by parametrically changing its duration in the target sequence, while eliminating all binaural cues. The influence of spectral content of the tail was examined by simulating the spectral coloration produced by real rooms. Results suggest that damped tails can elicit perceptual segregation with tail durations less than 100 ms. In addition, the spectral content of the tail can further influence segregation performance. Overall, differences in reverberation can serve as a prominent cue to aid perceptual segregation, particularly if a room environment introduces differences in spectrum based on distance, which is often the case in real rooms. [Work supported by NIH grant R01DC07657.]
Streaming and sound localization with a preceding distractor
Authors: Andrejkova Gabriela , Best Virginia , Shinn-Cunningham Barbara G. , Kopčo Norbert
Published: May 2015

A previous study of sound localization with a preceding distractor showed that (1) the distractor affects response bias and response variance for distractor-target inter-stimulus-intervals of up to 400 ms, and that (2) localization responses are biased away from the distractor even on interleaved control trials in which the target is presented alone [Kopco et al., JASA 121, 420–432, 2007]. Neural mechanisms operating on time scales of milliseconds to tens of seconds are likely to cause to these effects. The current study examined how perceptual organization affects target localization performance. Sound localization was examined for 2-ms click target stimuli. On 80% of trials, the target was preceded by a distractor, designed either to be grouped with the target (distractor was an identical 2-ms click) or to be perceived in a separate stream (an isochronous train of eight clicks whose inter-click-interval was different from the distractor-target inter-stimulus-interval). As hypothesized, the single-click distractor affected target localization more than the eight-click distractor. On the other hand, the biases in the control trials were greater for the eight-click distractor. These results indicate that performance is influenced by both top-down mechanisms like streaming and bottom-up mechanisms like stimulus distribution-based adaptation. [Work supported by APVV-0452-12 and R01DC009477.]
Time dependence of the time interval in distance/ intensity threshold in human listeners
Authors: Dunai Larisa , Lengua Ismael , Iglesias Miguel
Published: May 2015

The paper presents the distance threshold for complex virtual sounds localization in dependence of the time interval. The method of measurement was a forced-choice, three-down one-up staircase. The sound is a complex record of a click sound of 47 ms at 44.1 kHz measured by using a maximum length binary sequence in an anechoic chamber. The azimuth of the experimental is 0°, at the center of the human head where the ITD = 0, varying just the intensity. The investigation reported that the lowest distance thresholds of 0.06 m occurred for high inter click intervals at 200 ms–300ms. At lower ICIs, the distance threshold increases considerably. The experimental was carried out with the objective to find the distance threshold and time interval for implementing the data for a navigation device for blind people.
Cochlear fine structure predicts behavioral decision weights in a multitone level discrimination task
Authors: Lee Jungmee M. , Long Glenis , Heo Inseok , Stoelinga Christophe , Lutfi Robert
Published: May 2015

Listeners show highly replicable, idiosyncratic patterns of decision weights across frequency affecting their performance in multi-tone level discrimination tasks. The different patterns are attributed to peculiarities in how listeners attend to sounds. However, evidence is presented in the current study that they reflect individual differences in cochlear micromechanics, which can be evaluated using otoacoustic emissions (OAEs). Spontaneous OAEs (SOAEs) and the fine structure of stimulus-frequency OAEs (SFOAEs) were measured in a group of normal-hearing listeners. The same group of listeners performed a two-tone, sample-level discrimination task wherein the frequency of one tone was selected to correspond to a SOAE and the other was selected well away from a SOAE. Tone levels were either 50 or 30 dB SPL. The relative decision weight of the two tones for each listener and condition was estimated from a standard COSS analysis of the trial-by-trial data [Berg (1989), J. Acoust. Soc. Am. 86, 1743–1746]. A strong linear relation was observed between the average relative decision weight and the average relative level of both SOAE and SFOAE.
Fast frequency selectivity measures in listeners with severe hearing loss
Authors: Hoover Eric C. , Blackburn Michael C. , Souza Pamela E.
Published: May 2015

Communication difficulties in severe hearing loss are a combination of elevated pure-tone thresholds and suprathreshold deficits; for example, frequency selectivity, which may be substantially impaired in listeners with severe loss. The purpose of this study was to investigate frequency selectivity measures in listeners with severe loss (pure-tone average 60–90 dB HL). Two tests were used: fast psychophysical tuning curves (PTC), in which an intermittent tone was detected during a narrowband noise masker swept continuously in frequency; and spectral ripple (SR), in which the number of sinusoidal ripples in the spectrum of a broadband noise was tracked in an adaptive forced-choice procedure. Results showed that both tests have advantages when evaluating listeners with severe loss. Fast PTC provided a direct representation of tuning in a specific frequency region but was limited by the residual dynamic range of the listener. Dynamic range limitations were partially overcome by combining data from multiple trials. SR was able to be performed by listeners with minimal dynamic range but the interpretation of results was limited by potential differences in hearing across the bandwidth of the stimuli. Both fast PTC and SR revealed information about suprathreshold deficits that could help guide audiological intervention. [Work supported by NIH.]
Effects of hearing impairment on sensitivity to dynamic spectral change
Authors: Molis Michelle R. , Srinivasan Nirmal , Gallun Frederick J.
Published: May 2015

The loss of peripheral auditory sensitivity, precise temporal processing, and frequency selectivity associated with hearing loss suggests that the results obtained for pure tone glide stimuli will not necessarily correspond to results obtained with more complex dynamic stimuli for listeners with hearing impairment. Normally hearing (NH) and hearing-impaired (HI) listeners identified changes in frequency as rising or falling both for tone glides and for spectrotemporal ripples. Tones glided linearly up or down in frequency with an extent of 1, 0.66, or 0.33 octaves centered around 500 or 1500 Hz. Ripple stimuli, presented in octave bands centered around 500 or 1500 Hz or in a broadband condition extending from 20–20,000 Hz, had a spectral density of 2 cycles/octave and temporal modulation gliding up or down at rates of 1, 4, or 16 Hz. Sensitivity to dynamic changes was assessed as percent correct direction identification and bias was characterized as the ratio of correctly-identified rising versus falling glides. Substantial individual variability was observed for both measures for both the NH and HI listeners, and the ability to perform the tone glide task was not a consistent predictor of the ability to perform the spectrotemporal ripple identification. [Work supported by NIH/NIDCD.]
Automating source localization and separation using sonic detection and ranging
Authors: Lu Yazhong
Published: May 2015

The sonic detection and ranging (SODAR) technology developed previously (Wu and Zhu, JASA, 2013) is automated for performing blind sources localization and separation. In particular, source localization results are displayed in terms of space-frequency or space-time correlations. In other words, one can either view distributions of sound sources in three-dimensional (3-D) space versus any user-defined frequency bands, or distributions of sound sources in 3-D space versus time history. The sound pressure level (SPL) values associated with the identified sources are also calculated and displayed in these space-frequency and space-time correlation graphs. To acquire a better understanding of the distribution of sound sources together with their SPL values in 3-D space, a 3-D viewer using Google SketchUp software is employed to view these results in both space-frequency and space-time correlations. With this 3-D viewer, one is able to rotate and look at any source from any perspective, and zoom in and out to examine the details of relative positions of individual sources. The information on source locations together with windowing and filtering technologies enable us to separate the individual source signals. Examples of using this blind sources localization and separations in a non-ideal environment that involves random background noise and unspecified interfering signals are demonstrated.
Speech recognition and listening effort across various conditions in adults with aphasia
Authors: Richie Carolyn
Published: May 2015

Some of the common complaints among adults with aphasia are that it is difficult to understand speech in noise and that listening can be very effortful. Nonetheless, there has been limited research to date on speech recognition in noise or listening effort for adults with aphasia. Stanfield & Richie (2014) investigated speech recognition under various listening conditions for adults with non-fluent aphasia. Participants with aphasia performed better in quiet as compared to noise, as expected, and better in noise that consisted of multi-talker babble as compared to one competing talker. They also showed modest benefit from the addition of visual cues to speech over auditory-only speech recognition. However, subjective reports of listening effort were mixed and did not line up with performance on the various tests of speech recognition. In the present study, the relationship between speech recognition under various listening conditions and an objective measure of listening effort, response time, was examined. Participants’ word and sentence recognition under auditory-only and auditory-visual conditions, in quiet and in two types of noise, will be reported. The potential clinical significance to better understanding listening effort in adults with aphasia will be discussed as well.
Using dual-task paradigms to assess listening effort in children and adults
Authors: Picou Erin , Ricketts Todd A.
Published: May 2015

While evidence suggests that hearing aids can reduce listening effort, mixed results are often seen when examining the effect of specific hearing aid processing. Given methodological differences across studies, we speculated that not all paradigms are equally sensitive to changes in listening effort. Moreover, many traditional paradigms may not be appropriate for children. The purpose of this project was to evaluate dual-task paradigms for measuring listening effort in adults and school-aged children. Three tasks were developed based on existing dual-task paradigms and modified to be appropriate for children. Sixteen adults (aged 22–32) and twenty-two children (aged 9–17) with normal hearing participated. All participants were tested in quiet and in noise. For all three paradigms, the primary task was monosyllable word recognition and the secondary task was a physical response time measure. The secondary tasks varied in complexity or depth of processing. Results revealed that the paradigm requiring deeper processing was most sensitive to the effects of noise for adults, but for children, the paradigm with the highest complexity was the most sensitive. Potential explanations for these group differences and the application of the most sensitive measures for future investigations will be discussed. [This work was funded by Phonak.]
Subjective listening effort
Authors: Schulte Michael , Krüger Melanie , Meis Markus , Wagener Kirsten C.
Published: May 2015

When investigating listening effort the individual subjective feedback is important to learn about and take into account the situations in which hearing impaired suffer in terms of listening effort and the relationship of listening effort to speech intelligibility. Here, we present results from a daily life questionnaire study and results from a new tool to evaluate listening effort subjectively under controlled conditions in the lab. A research questionnaire was used to determine the individual subjective listening effort in daily life in a systematic way. The questionnaire with 29 well described situations was used in a multi-center study in Denmark, Germany, and USA with 112 subjects to identify the most relevant effortful situations (e.g., “watching news in TV” or “talking at the kitchen table”). As a lab procedure we developed an adaptive categorical listening effort scaling procedure. It is based on a non-adaptive version that turned out to be a sensitive lab method in which subjects rate listening effort at predefined signal to noise ratios. The new method automatically finds the SNRs corresponding to subjective ratings from “extreme effort” to “no effort.” First results show the relationship between listening effort and intelligibility for different SNRs and background noises.
Adaptive beamforming for acoustic echo cancellation using measured array models and subband filtering
Authors: Bai Mingsian R. , Chi Li-Wen
Published: May 2015

Acoustic echo that can substantially undermine speech quality is one of the key issues must be addressed in practical telecommunication systems. In this paper, an evolutionary exposition is given in regard to the enhancing strategies for acoustic echo cancelers (AEC). On the basis basic adaptive filtering, a highly directional microphone array is designed as a fixed beamformer (FBF) to focus on the near-end speaker and suppress the echo from the far-end as well as noise and interference from the background. Design of the beamformer is based on array models interpolated from measured frequency responses. Subband (SB) filtering with polyphase decomposition is exploited to accelerate the cancellation process with multiple choice of step size in the subbands. To further enhance the canceler, adaptive generalized sidelobe canceler (GSC) can be utilized, which is comprised of a FBF for the near-end speech and an adaptive blocking module for the echo and noise. Objective tests in terms of echo return loss enhancement (ERLE) and perceptual evaluation of speech quality (PESQ) are conducted to compare four algorithms with various combinations of enhancement. The results show that the GSC-SB-AEC approach has attained the highest ERLE and best speech quality, even in a double-talk scenario.
Tone-to-text applications for sub-Saharan African tone languages
Authors: Carter-Enyi Aaron
Published: May 2015

Based on experimental findings, this paper proposes a framework for computer recognition of speech tones in Niger-Congo languages. This language family includes over 1000 languages in Sub-Saharan Africa and approximately one billion speakers. Many of these languages use pitch contrast to differentiate words in a system of two or more pitch levels (e.g., high, mid, and low). The results of two new studies conducted in Nigeria from 2013-14 indicate speech-tone is perceived as tonemic intervals (e.g., high-high, high-mid, and high-low). An experimental study (n = 1448) identified ranges of pitch difference that form perceptual categories for these tonemic intervals. An initial application of these findings is adding tone diacritics to text by interpreting fundamental frequency. Representing tone in text has been a persistent problem for Niger-Congo languages and smartphones are ill-equipped for marking tone diacritics. A signal processing application that recognizes speech tones and marks them on computer text would be more efficient than manually highlighting and marking each syllable. Frequency distribution of tones is dynamic between speakers. Thus, there are potential benefits of using machine learning to create speaker-dependent software. Because the proposed method relies on existing algorithms for fundamental frequency, the problem of estimation errors will also be addressed.
Subarray based localization using extended coprime arrays
Authors: Pyzdek Andrew T. , Culver R. Lee , Swanson Dave
Published: May 2015

Partitioning of long linear arrays into a number of smaller subsections, termed subarrays, is a common form of processing used both to compensate for irregularity in array shape and to localize near-field sources. While this manner of processing is applicable to uniform linear arrays, as the subarrays are similarly uniform and possess the same minimum element spacing, subarray methods cannot always be applied to sparse arrays with uneven spacing along their length. One sparse array design of interest is the extended coprime array, an array composed of two uniformly spaced component arrays, each undersampled by integer factors which are selected to be coprime. By exploiting the regularity of spatial lag repetitions in extended coprime arrays, we show that appropriately selected subsections of a coprime array can be used as subarrays to determine distance to a near-field source. The performance of the coprime array processed in this manner will be compared to similar processing performed on alternative sparse array designs and the baseline performance of a uniform linear array of equal aperture.
The study of adaptive time reversal mirror as a crosstalk mechanism for underwater communication
Authors: Lin Yi-Wei
Published: May 2015

The time reversal mirror technique has been widely applied to mitigate the inter-symbol interference in underwater channels. Meanwhile, an adaptive time reversal mirror is introduced to improve the crosstalk quality between receivers in underwater communication. To explore the effectiveness of this method, this study extended the analysis of adaptive time reversal mirror as a crosstalk mechanism and explored the mechanism in an experiment using a towing tank as a testing platform. The advantage of this process is its simplicity in examining the effects of the array configuration of this crosstalk mechanism. Results of parametric experiments are discussed i.e. the effects of number of receivers, spacing between sources and noise energy threshold. Experimental data at 10 and 16 kHz with a 5-kHz bandwidth demonstrate as much as an 8-dB signal to noise ratio improvement for four receivers dual sources over a 30 m communication range in a 3.3 m depth testing platform. The results indicate array configuration affects this mechanism drastically.
Array shape calibration using long-term ambient noise records
Authors: Nichols Stephen , Bradley David L.
Published: May 2015

When using an acoustic array to determine the bearing of a source, errors in the sensor positions severely reduce the accuracy of any bearing measurements. This issue is particularly problematic for three sensor arrays, designed to work in two dimensions, because of the lack of redundancy built into the array. This paper presents a method for correcting errors in sensor positions. It is mathematically straightforward to determine the bearing of a far-field impulsive signal by determining the time difference of arrival between sensors by cross-correlation. These time delays can also be used to estimate the sound speed in the vicinity of the array. In an isotropic acoustic medium, the local sound speed is expected to be independent of the source bearing. If the sensor positions used to determine the sound speed and bearing are incorrect, the resulting sound speed measurements will be bearing-dependent. Using an analytically derived function, the correct array shape (with only translational and rotational ambiguity) can be backed out from the directional sound speed distribution. This method will be demonstrated using long-term ambient noise records from the Comprehensive Nuclear-Test Ban Treaty Organization’s (CTBTO) hydrophone arrays.
Maximum entropy inference of seabed attenuation parameters using ship radiatedbroadband noise
Authors: Knobles David P.
Published: May 2015

The use of ambient noise as a means of inferring physical properties of the ocean is of significant interest in ocean acoustics. This study employs the received acoustic field generated by the passage of the R/V Knorr on the New Jersey continental shelf to test a new approach that estimates the probability distributions of both the aspect dependent source levels and the parameters that represent the geoacoustic structure of the seabed. Since the source levels and the environmental parameters have an intrinsic ambiguity, both classes of parameters must be included in the model hypothesis space as random variables. The statistics of the error function, needed to uniquely specify the likelihood function, are estimated with a maximum entropy approach by creating a data ensemble that includes samples from time periods where the ship-receiver geometry is dominated by stern, bow, and port aspect. This method has its origins from the observation that Bayes’ rule is symmetric with the interchange of the data space and the hypothesis space. [Research supported by ONR 32 OA.]
Passive acoustic source localization using sources of opportunity
Authors: Verlinden Christopher , Sarkar Jit , Hodgkiss William , Sabra Karim , Kuperman William
Published: May 2015

Rather than use replica fields for matched field processing (MFP) derived from acoustic models requiring detailed environmental input, we demonstrate that data derived replicas from ships of opportunity can be used for assembling a library of replicas for MFP. The Automatic Identification System (AIS) is used to provide the library coordinates for the replica library and a correlation based processing procedure is used to overcome the impediment that the replica library is constructed from sources with different spectra and will be used to locate another source with it own unique spectral structure. The method is illustrated with simulation and then verified using acoustic data from a 2009 experiment for which AIS information was retrieved from the Unites States Coast Guard Navigation Center (USCG NAVCEN) Nationwide AIS (NAIS) database.
Eigenvector-based signal subspace estimation
Authors: Quijano Jorge E. , Zurk Lisa M.
Published: May 2015

In this work, we explore the performance of a new algorithm for the estimation of signal and noise subspaces from limited data collected by a large-aperture sonar array. Based on statistical properties of scalar products between deterministic and complex random vectors, the proposed algorithm defines a statistically justified threshold to identify target-related features (i.e., wavefronts) embedded in the sample eigenvectors. This leads to an improved estimator for the signal-bearing eigenspace that can be applied to known eigenspace beamforming processors. It is shown that data projection into the improved subspace allows better detection of closely spaced targets compared to current subspace beamformers, which utilize a subset of the unaltered sample eigenvectors for subspace estimation. In addition, the proposed threshold gives the user control over the maximum number of false detections by the beamformer. Simulated data are used to quantify the performance of the signal subspace estimator according to a normalized metric that compares estimated and true signal subspaces. Improvement on beamforming resolution using the proposed method is illustrated with simulated data corresponding to a horizontal line array, as well as experimental data from the Shallow Water Array Performance experiment.
Visualization of multiple array acoustic source localization
Authors: Scanlon Michael V.
Published: May 2015

Transient acoustic events such as gunfire, rocket launches, and explosions can be located using triangulation with multiple lines-of-bearing (LOBs) generated by distributed acoustic microphone arrays. The Army has fielded diverse sound localization and ranging systems that work well in open field environments, but suffer reduced performance in urban environments where echoes, multipath, diffraction, and blockages modify both the signatures and the propagation path. This demonstration will use light projected through a water basin to a screen to show traveling waves (ripple diffraction patterns) from transient events and how those wavefronts pass differently across the distributed array locations. The observers will gain an understanding of how multiple microphone elements in a small cluster of microphones (an array) can derive time-difference-of-arrival (TDOA) estimates to create array LOBs, as well as an introduction to triangulation through multiple distributed array LOBs. Objects simulating walls and buildings will be placed in the basin to demonstrate how complex waveforms and propagation paths can decrease the accuracy of acoustic localization systems.
Man-made noise and aquatic Life: data, data gaps, and speculation
Authors: Popper Arthur N.
Published: May 2015

This talk will consider how man-made sounds may impact aquatic life—with a focus on fishes (with possible digressions to invertebrates, marine mammals, and turtles). The talk will start with a discussion of a Jacques Cousteau movie and then ask why animals (and humans) hear. After considering how fishes (and invertebrates) hear, the focus of the talk will turn to potential impacts of sound on these animals and what is currently actually known (based on data and not speculation) about effects on physiology and behavior. The talk will conclude with a consideration of recent guidelines on effects of sounds and then provide an overview of data gaps and the most important areas for future research.
Prescribing healthy hospital soundscapes
Authors: Ryherd Erica E. , Persson Waye Kerstin
Published: May 2015

In The Effects of Noise on Man, Karl Kryter emphasized that human reaction to sound is quantitatively related to the physical nature of sounds. He specified five unwanted characteristics of sound: “(a) the masking of unwanted sounds, particularly speech, (b) auditory fatigue and damage to hearing, (c) excessive loudness, (d) some general quality of bothersomeness or noisiness, and (e) startle.” Hospital soundscapes have been shown to demonstrate all five characteristics in one fashion or another. Some unwanted effects of sound have also been shown, such as fragmented sleep and recuperation, cardiovascular response, pain and intensive care delirium—however, few studies have been able to causally link sounds to patient outcomes and few examine detailed characteristics other than loudness. This paper will summarize what we know about noise in hospitals and the health effects on occupants, including highlights from the Healthcare Acoustics Research Team (HART) body of research. Results will be used to identify important areas for future research.
Age-related hearing loss, noise-induced hearing loss, and speech-understanding performance
Authors: Humes Larry E.
Published: May 2015

Karl Kryter was keenly interested in the effects of noise on hearing, the interaction of age-related hearing loss (ARHL) with noise-induced hearing loss (NIHL), and, ultimately, the consequences of both on an individual’s ability to understand speech in quiet and in noise. This review will touch on each of these topics and the interrelationships among them. The focus of this presentation will be on discussion of factors impacting the speech-understanding performance of older adults, with parallels drawn for NIHL where possible. Factors considered will include those represented by a variety of peripheral-auditory, central-auditory, and cognitive measures. Interactions among these factors will also be reviewed.
Audio-vocal feedback in bats and new roles for echolocation calls in social communication
Authors: Bohn Kirsten M. , Smotherman Michael
Published: May 2015

An important aspect of auditory scene analysis is the specialized neurocircuitry required for vocal production. in a dynamic acoustic environment. Although often taken for granted, enhanced audio-vocal feedback is relatively uncommon in animals and yet an important precursor to vocal learning. We argue that vocal complexity in bats is an exaptation of the highly specialized audio-vocal feedback system that has evolved for echolocation. First, we explore how audio-vocal feedback enhances echolocation. Second, we review how echolocation pulses serve social functions by providing information to receivers (like gender, identity, or food availability). Third, using our research on molossid bats (family Molossidae), we explore whether vocal plasticity in sonar has contributed to an expanded role for sonar pulses in social communication. In at least three molossids, roosting bats rapidly sing in response to echolocation pulses of flying conspecifics. However, more importantly, we show that in multiple species, echolocation is produced in purely social contexts. Roosting bats embed pulses and feeding buzzes into their courtship songs that are not acoustically distinct than when foraging. Finally, some molossids not only sing in roosts, but also in flight—that is, they echolocate and sing simultaneously. These findings indicate that echolocation plays even more of a role in social communication than commonly believed and that the production of echolocation and social communication is tightly coupled and coordinated at a high level.
Nocturnal peace at a Conservation Center for Species Survival?
Authors: Wiseman Susan M. , Wilson Preston S.
Published: May 2015

E.O. Wilson suggested that the natural world is the most information-rich acoustic environment, especially for animals and indigenous peoples, and that is directly and indirectly essential to their survival (Wilson, 1984, Biophilia, Harvard University Press). Agriculturalists still value and constantly monitor their soundscape since it provides invaluable cues that may lead to success or failure. Krause reports that healthy natural soundscapes comprise a myriad of biophony, and indeed the ecological health of a region can be measured by its diverse voices (Krause, 1987, “Bio-acoustics: Habitat ambience & ecological balance,” Whole Earth Rev.). But do such soundscapes fall silent for much of the night? And are there extensive periods of silence at a highly successful Conservation Center for Species Survival? This study analyzes the soundscape continuously recorded at Fossil Rim Wildlife Center in Texas for a week during Fall 2013, to determine the prevalence of quiet periods and the acoustic environment in which such periods appeared to occur.
Underwater auditory scenes near coral reefs
Authors: Staaterman Erica , Paris Claire
Published: May 2015

Recent studies suggest that reef soundscapes may serve as long-distance navigational cues for settlement-stage larval fishes. To understand the role of acoustic signals during larval settlement, we investigated temporal and spatial patterns in coral reef soundscapes in the Florida Keys, USA. We used 14-month simultaneous acoustic recordings from two nearby reefs, coupled with environmental data, to describe temporal variability in the soundscape on scales of hours to months, and to understand abiotic and biological components. We also recorded acoustic pressure and particle acceleration (which fish larvae likely perceive) with increasing distance from the reef. High acoustic frequencies typically varied on daily cycles, while low frequencies were primarily driven by lunar cycles. Some of these patterns were explained by environmental conditions, while others were attributed to biological sounds. At both reefs, the highest sound levels (~130 dB re:1μPa) occurred during new moons of the wet season, when many larval organisms settle on the reefs. Acoustic particle acceleration did not propagate as far as predicted from the plane-wave equation. The patterns uncovered here provide valuable insights into underwater acoustic scenes, and this study represents an example of novel acoustic instruments and analysis techniques that can be applied to any ecosystem.
Thermoacoustic approach for the cloaking of a vibrating surface
Authors: Manela Avshalom , Pogorelyuk Leonid
Published: May 2015

A vibrating surface in a quiescent fluid transmits pressure fluctuations, which propagate into far-field sound. The vibroacoustic mechanism involved, coupling vibration and sound, is common in a large number of applications. Yet, there is a growing interest in developing means for achieving “acoustic cloaking” of an animated boundary. We suggest the heating of a surface, generating thermoacoustic perturbations, as a mechanism for monitoring vibroacoustic sound. Considering a setup of an infinite planar wall interacting with a semi-infinite expanse of an ideal gas, we investigate the system response to arbitrary (small-amplitude) vibro-thermal excitation of the confining wall. Analysis is based on continuum Navier-Stokes-Fourier and kinetic Boltzmann equations, and supported by stochastic direct simulation Monte Carlo calculations. Starting with a case of a sinusoidally excited boundary, a closed-form solution is derived in both continuum and collision-free limits. The results, found valid at a wide range of frequencies, indicate that effective cloaking of the boundary may be achieved through optimal choice of boundary-generated heat flux. The results are rationalized by considering the quasistatic limit of low-frequency excitation. The analysis is then extended to consider the system response to impulse (delta-function) actuation, where the independence of cloaking conditions on perturbation frequency is demonstrated.
Monitoring floor impact sounds using detection algorithms in multi-story residential buildings
Authors: Jeon Jin Yong , Hong Joo Young , Lim Hansol , Imran Muhammad
Published: May 2015

A monitoring and alarming system was designed to detect household noise occurrence in the multi-story residential buildings. This system is composed of a microphone combined with a home smart pad installed on a wall and a main server that can store data to discriminate household noise occurrence. Sound levels such as LAeq and LAmax were calculated in real time using developed digital signal processor (DSP) of the acoustical parameter analyzer in the home smart pad. A correction equation was applied based on the standard measurement method to provide accurate measurements. After measurement, the results were saved in the main server using Ethernet and inter-compared with the data of each household to discriminate the household noise occurrence. Finally, the alarm was raised depending on the decision criteria based on both the regulations of Korean government and subjective evaluation results from the previous study.
Experimental quantification of microphone windscreens and noise cones in a low speed wind tunnel
Authors: Barnard Andrew R.
Published: May 2015

Wind noise is a common challenge in all types of atmospheric acoustics as well as in wind tunnel applications in the aerospace and automotive industry. The dynamic pressure field measured by the microphone includes both the acoustic pressure and the pressure induced by the turbulent air flow over the diaphragm. There are three types of microphone accessories that are commonly used to perform wind noise isolation: protective grid-caps (no isolation), windscreens, and nose-cones. In this study, each of these microphone accessories is tested in a wind tunnel using ¼” and ½” microphones in turbulent and laminar flows at flow speeds up to 55 mph and in both the head-on and parallel diaphragm orientations. Two types of flush mount microphones, a surface mount and a side-vented pressure microphone, are also evaluated. The effects of background flow noise on the microphone measurement is shown for all conditions and a case study is presented with an arbitrary acoustic source. Finally, recommendations are presented for which accessories are best used in different measurement situations.
Pipe systems for infrasound wind-noise reduction: Issues and approaches
Authors: Gabrielson Thomas B.
Published: May 2015

Wind-noise-reduction for infrasound monitoring below 10 Hz often takes the form of multiple-inlet pipe networks that average the field over distances small with respect to the acoustic wavelength but large with respect to the turbulence scale. There is a long tradition of pipe systems; however, traditional approaches can result in mediocre performance. Three aspects of the traditional design—inlets arranged in a regular geometric pattern, resistive resonance suppression, and equal-length inlet-to-sensor paths—can all be questioned. This paper reviews pipe-system performance with respect to design, modeling, and data quality. Waveguide models that account for the acoustic adiabatic-to-isothermal transition inside the pipes and summing junctions have been validated for prediction of complex frequency response and horizontal and vertical directionality. Techniques have also been developed for in-situ measurement of system frequency response and wind-noise-reduction effectiveness. The importance of on-site measurement is underscored by the number of performance problems uncovered at operational infrasound monitoring sites by in-situ techniques. These issues will be illustrated with measurements at operational infrasound-monitoring stations and at test-bed systems.
Long duration wind noise abatement measurements at the University of Mississippi Field Station
Authors: Talmadge Carrick L.
Published: May 2015

A series of wind noise abatement tests were performed at the University of Mississippi Field Station from January through June of 2014. These tests compared bare sensors to sensors treated with porous hoses, and domes of different dimensions (40, 48, 60, and 96 in.ch) that were covered either with foam or with metal meshing (30% and 45% opening). Because these measurements were obtained continuously, measurements were obtained over a wide range of weather conditions. As expected, the frequency of maximum wind noise attenuation decreased with increasing diameter of the dome, and the maximum attenuation obtained increased commensurately with diameter. Tests were also performed with nested domes (96 and 40 in.ch), which provided greater attenuation than just the larder of the two domes. The 30% opening mesh proved superior to the 45% opening mesh. Generally, the porous hoses provided superior wind noise protection below 1-Hz, but did not provide as much wind noise attenuation as was provided by the 96 and 40 in.ch nested wind dome configuration. We found that a number of issues (e.g., less robust, more affected by wet conditions) make this a less idea solution than the metal mesh covered nested domes, for permanent installations.
Inserting the speech signal above the cochlea: How can this work?
Authors: McKay Colette M.
Published: May 2015

Although cochlear implants (CIs) have successfully restoring hearing to many deaf people, there remains a subgroup who cannot benefit from them due to malformation of the cochlea or damage to the cochlear nerve. Auditory brainstem implants (ABIs) and auditory midbrain implants (AMIs) aim to restore hearing to this group. ABIs stimulate the surface of the cochlear nucleus (CN). Despite some ABI users gaining benefit similar to a CI, the majority achieve poor speech understanding with the device alone. The reasons for poor outcomes are not fully understood but include tumor damage to particular cells in the cochlear nucleus, surgical approach, and placement of the electrodes. One factor in poor outcome is the difficulty of stimulating specific tonotopic planes within the CN. This factor and tumor damage to the CN were reasons for the development of the AMI, which directly stimulates the tonotopic layers of the inferior colliculus (IC) using a shank electrode. Psychophysics and physiological studies have thrown light on the way that new electrode designs and processing strategies may provide patients with AMIs and potentially ABIs with hearing commensurate with CIs. [The Bionics Institute acknowledges the support it receives from the Victorian Government through its Operational Infrastructure Support Program.]
Sufficiency of a four-parameter spectral model of the voice source
Authors: Kreiman Jody , Gerratt Bruce R. , Signorello Rosario , Rastifar Shaghayegh
Published: May 2015

Our proposed perceptually motivated spectral-domain model of the voice source comprises spectral slopes in four ranges (H1-H2, H2-H4, H4-the harmonic nearest 2 kHz, and that harmonic to the harmonic nearest 5 kHz). Previous studies established the necessity of these parameters by demonstrating that listeners are sensitive to all of these parameters, and that all are needed to adequately model voice quality. The present study examines the sufficiency of the model to quantify voice quality by applying it to a very large set of voices. To determine the range of phenomena for which the source model is valid, 200 natural voices (half male; 50 normal/mildly pathologic, 30 moderately pathologic, 20 severely pathologic) were copy-synthesized using the model, and adequacy of synthesis was assessed in a listening test by comparing the synthetic tokens to the natural targets. Validity of the model as a function of severity of pathology and the particular manner in which the stimuli deviate from normal will be discussed. Sources of mismatches between synthetic and natural stimuli will be identified, and their status with respect to definitions of voice quality will be addressed. [Work supported by NIH.]
Detecting palatalization in spontaneous spoken English
Authors: Renwick Margaret E. , Cassidy Caitlin N.
Published: May 2015

We present an analysis of palatalization from /s/ to [ʃ] at word boundaries in UK English. Previous work has considered the effect of lexical frequency (LF) on this phenomenon, but without combining acoustics and spontaneous speech in one study, which we undertake using data gathered from the Audio BNC (http://www.phon.ox.ac.uk/AudioBNC). We analyze 5,259 word pairs in five phonological contexts, comparing the acoustics of test tokens subject to palatalization (e.g., /s/ in miss you), to control tokens containing non-alternating [s] (miss it) or [ʃ] (mission, wish it, wish you). Word and segment boundaries were obtained via forced alignment, but hand-checked. We measured the spectral moments and duration of each fricative; following vowel duration and formant values were also extracted. LF was calculated using the Audio BNC. We find that the spectral center of gravity (CoG) of [s] before [j] (miss you) is intermediate (p<0.01) between those of [s] and [ʃ]. Furthermore, CoG of test tokens correlates negatively with LF, indicating increased palatalization in high-frequency contexts; no control pair exhibits such correlations. LF predicts CoG even considering speaker gender and other phonetic co-predictors. This supports the view of palatalization as gestural overlap, which increases with LF in casual or fast speech.
Individual differences in phonetic drift by English-speaking learners of Spanish
Authors: Huffman Marie K. , Schuhmann Katharina
Published: May 2015

Second language learning affects L1 pronunciation even in early stages of adult L2 acquisition (e.g., Guion 2003, Chang 2012), an effect sometimes called “phonetic drift.” English voiceless stops assimilate to Korean stops in novice L1 English-L2 Korean learners in Korea, arguably due to English voiceless stops and Korean aspirated stops being linked to the same phonological category (Chang 2012), and the “novelty” of the Korean language input with its longer VOTs (Chang 2014). We tested whether novice L1 English-L2 Spanish learners in the United States also show assimilatory L2-to-L1 effects. Stronger typological and orthographic similarities might lead speakers to equate English and Spanish stops, but phonetic differences—short-lag VOT on voiceless stops and pre-voicing of voiced stops in Spanish—might impede L2-to-L1 assimilation. Monolingual English college-students were recorded every other week for the first 12 weeks of an introductory Spanish course. VOT data show considerable individual differences—some subjects show assimilation, some dissimilation, between English and Spanish stops. This suggests that while orthography and typological similarity may play a role, how a learner’s first and second language interact in novice adult L2 acquisition is also affected by individual factors such as perceptual sensitivity to phonetic differences, sociolinguistics considerations, and possibly speaker strategies for maintaining contrast (e.g., Nielsen 2011).
Developmental time course of auditory perception of modulation speech cues
Authors: Lorenzi Christian
Published: May 2015

Speech contains strong amplitude modulation (AM) and frequency modulation (FM) cues, which are commonly assumed to play an important role in speech identification for adults. We will review recent studies aiming to characterize the development of auditory perception of AM and FM speech cues for 6 and/or 10-month-old infants learning French or Mandarin [e.g., Cabrera, L., Tsao, F.-M., Gnansia, D., Bertoncini, J., & Lorenzi C. (2014), J. Acoust. Soc. Am. 136, 877–882; Cabrera, L., Bertoncini, J., & Lorenzi, C. (2013), J. Speech, Lang., Hearing Res. 56, 1733–1744]. These studies were based on vocoders, which are analysis and synthesis systems designed to manipulate the modulation components of speech sounds in a given number of frequency bands. Overall, the results suggest that: (i) the auditory processing of AM and FM speech cues is “functional” by 6 months, (ii) the auditory processing of the AM and FM cues is fine-tuned by language exposure between 6 and 10 months. These findings may help improving current models of modulation processing that do not take into account the plasticity of the auditory and speech-processing system.
Using automatic speech recognition to identify pediatric speech errors
Authors: Sadeghian Roozbeh , Ratnagiri Madhavi , Zahorian Stephen A. , Bunnell H. Timothy
Published: May 2015

Speech delay is a childhood language problem that might resolve without intervention, but might alternatively presage continued speech and language deficits. Thus, early detection through screening might help to identify children for whom intervention is warranted. The goal of this work is to develop Automatic Speech Recognition (ASR) methods to partially automate screening for speech delay in young children. Speech data were recorded from typically developing and speech delayed children (N = 63) aged 6 to 9 years old during administration of the Goldman Fristoe Test of Articulation (GFTA). Monophone Hidden Markov Model (HMM) acoustic models were trained on speech data obtained from 207 typically developing children in the same age range. These training data consisted of a total of about 18,000 single-word utterances. The HMMs were then used to develop an utterance verification system to distinguish correct versus error productions. Several variations of the recognition strategy, feature extraction, and scoring methods were investigated. The best overall ASR result for distinguishing normal versus abnormal speech is approximately 86%. It is hypothesized that the ASR methods could approach the level of accuracy of speech therapists for this task (agreement among multiple therapists is over 95%), but a much larger database may be needed.