bluuebunny/crossref_metadata_2025 huggingface dataset:
WILLIAM WHEWELL, THE PLURALITY OF WORLDS, AND THE MODERN SOLAR SYSTEM
Authors: Crowe Michael J.
Published: May 2016

Astronomers of the first half of the nineteenth century viewed our solar system entirely differently from the way twentieth‐century astronomers viewed it. In the earlier period the dominant image was of a set of planets and moons, both of which kinds of bodies were inhabited by intelligent beings comparable to humans. By the early twentieth century, science had driven these beings from every planet in our system except the Earth, leaving our solar system (and perhaps others) as more or less desolate regions for the most part bereft of intelligent life. This essay traces this extinction and its relation to religious thought, noting the role played in it by Sir John Herschel and especially by William Whewell. The inverse square laws for gravitation, heat radiation, and light receive special attention, as does the question of the relevance of the Christian notions of a divine incarnation and redemption.
Dynamic Models of Thermal Systems Using an Energy-Based Modeling Approach
Authors: Grossi Federica , Zanasi Roberto
Published: May 2016

The aim of this work is to give a new approach to obtain compact dynamic thermal models suitable for a variety of systems where the heat transfer can be caused by conduction, internal convection (not at the boundary), and evaporation/condensation of water. The structural properties of the proposed dynamic model are presented and discussed in this paper. These properties guarantee conservation of energy and mass within the system, thus giving a good confidence in the correctness of the model. This paper shows that the proposed model has a simple structure, can be easily implemented in simulink, and provides simulation times much shorter compared with those usually obtained using CFD programs. The proposed model proves to be suitable for real-time simulations and for control design purposes.
Resonance Oscillation of Third-Order Forced van der Pol System With Fractional-Order Derivative
Authors: Van Khang Nguyen , Thuy Bui Thi , Chien Truong Quoc
Published: May 2016

This study aims to investigate the harmonic resonance of third-order forced van der Pol oscillator with fractional-order derivative using the asymptotic method. The approximately analytical solution for the system is first determined, and the amplitude–frequency equation of the oscillator is established. The stability condition of the harmonic solution is then obtained by means of Lyapunov theory. A comparison between the traditional integer-order of forced van der Pol oscillator and the considered fractional-order one follows the numerical simulation. Finally, the numerical results are analyzed to show the influences of the parameters in the fractional-order derivative on the steady-state amplitude, the amplitude–frequency curves, and the system stability.
Method for measuring thermal accommodation coefficients of gases on thin film surfaces using a MEMS sensor structure
Authors: Grau Mario , Völklein Friedemann , Meier Andreas , Kunz Christina , Heidler Jonas , Woias Peter
Published: May 2016

A method for measuring the thermal accommodation coefficient α for surface-/gas interfaces is presented. It allows the determination of α for thin films produced by a variety of deposition technologies, such as chemical vapor deposition, physical vapor deposition, and atomic layer deposition (ALD). The setup is based on two microelectromechanical systems (MEMS) Pirani sensors facing each other in a defined positioning. Because these MEMS sensors show a very high sensitivity in their individual molecular flow regimes, it is possible to measure the accommodation coefficients of gases without the disturbing influence of the transition regime. This paper presents the analytical background and the actual measurement principle. The results for air and nitrogen molecules on sputtered Au and Pt surfaces are presented.
Simulating electrical modulation detection thresholds using a biophysical model of the auditory nerve
Authors: O'Brien Gabrielle E. , Imennov Nikita S. , Rubinstein Jay T.
Published: May 2016

Modulation detection thresholds (MDTs) assess listeners' sensitivity to changes in the temporal envelope of a signal and have been shown to strongly correlate with speech perception in cochlear implant users. MDTs are simulated with a stochastic model of a population of auditory nerve fibers that has been verified to accurately simulate a number of physiologically important temporal response properties. The procedure to estimate detection thresholds has previously been applied to stimulus discrimination tasks. The population model simulates the MDT–stimulus intensity relationship measured in cochlear implant users. The model also recreates the shape of the modulation transfer function and the relationship between MDTs and carrier rate. Discrimination based on fluctuations in synchronous firing activity predicts better performance at low carrier rates, but quantitative measures of modulation coding predict better neural representation of high carrier rate stimuli. Manipulating the number of fibers and a temporal integration parameter, the width of a sliding temporal integration window, varies properties of the MDTs, such as cutoff frequency and peak threshold. These results demonstrate the importance of using a multi-diameter fiber population in modeling the MDTs and demonstrate a wider applicability of this model to simulating behavioral performance in cochlear implant listeners.
Effect of room acoustics on speech perception by children with hearing loss
Authors: Peng Z. Ellen , Pausch Florian , Fels Janina
Published: May 2016

To study speech perception of children with hearing loss in virtual acoustic environments (VAE), a pair of research hearing aids has been previously integrated in a real-time dynamic binaural reproduction system. The auralization included simulations of room acoustics using individualized head-related and hearing aid-related transfer functions (HRTF and HARTF). In this study, a release from masking paradigm by Cameron and Dillon (2007) was adopted in German to investigate speech intelligibility by children fitted with hearing aids under realistic classroom acoustics. When immersed in VAE, each child was asked to repeat sentences spoken by a target talker always located at 0° azimuth, while two distractor talkers were continuously telling unfamiliar Grimm stories. The speech reception threshold (SRT) at 50% intelligibility was measured adaptively by changing the target talker speech level. A total of eight conditions were tested with each child by changing spatial cues (target-distractor collocated versus spatially separated at 90° horizontally), pitch cues (target-distractor sharing the same versus different voice), and room acoustics (0.4 versus 1.2 s mid-frequency averaged reverberation time). Measured SRTs from children with hearing loss will be compared with those from age-matched normal-hearing children. [Work supported by EU Seventh Framework- iCARE, Improving Children’s Auditory Rehabilitation.]
Optimizing the signal to noise ratio in classrooms using passive acoustics
Authors: D'Antonio Peter
Published: May 2016

Adults with normal hearing require a roughly 0 dB signal-to-noise ratio (SNR) for good speech intelligibility in classrooms and lecture halls. However, values in excess of 15 dB are needed to compensate for neurological immaturity, sensorineural and conductive hearing losses, language proficiency, and excessive reverberation. ANSI 12.60 addresses ways to lower the noise interference due to background levels and reverberation time. However, simply reducing the mid-high frequency reverberation time is not sufficient and may lead to masking of the high frequency consonant information, needed for speech intelligibility, with lower frequency vowel sounds. Research has shown that increasing the signal by including early reflections is also needed. This presentation will discuss a classroom design, along with several potential acoustical treatments, which aims to passively increase the SNR and improve speech intelligibility. These acoustic tools include a multi-layer, broad bandwidth absorptive panel, a hybrid diffusive/absorptive panel, an absorptive wood system simultaneously offering early reflections and mid-low frequency sound absorption and an LED lighting fixture system that simultaneously provides light and sound diffusion, as well as variable color and intensity for task lighting that has been shown to improve reading speed, increase test scores, and reduce hyperactivity.
Assessment of numerical accuracy for the direct computation of sound propagation
Authors: Sabatini Roberto , Marsden Olivier , Bailly Christophe , Bogey Christophe
Published: May 2016

The computation of long-range sound propagation, which consists in solving the full 3-D compressible Navier-Stokes equations in order to include all physical phenomena correctly, is now feasible. These progress are mostly due to the development of high fidelity algorithms in aeroacoustics over the last two decades, characterized by a high spectral accuracy and a low selective dissipation. The accuracy of these algorithms has been validated by considering various benchmark problems in computational aeroacoustics. However, assessment of the numerical accuracy remains tricky for nonlinear problems. In the context of long-range atmospheric propagation of infrasounds, nonlinear effects induced by large relative amplitudes and molecular absorption in the upper atmosphere can be significant. A direct computation of the propagation and a spectral analysis of the acoustic signals are here performed. The importance of the numerical dissipation is examined by considering the spectral transfer function of the numerical algorithm, including the shock-capturing scheme, which can be explicitly expressed for the optimized finite-difference schemes used in this work.
Eulerian methods for high frequency acoustics
Authors: Martinelli Sheri
Published: May 2016

The high frequency approximation to the wave equation yields an eikonal equation for the wave phase and a transport equation for the amplitude. Unfortunately, straightforward computation of the eikonal equation poses a challenge in that solutions may be multivalued which affects the convergence of a numerical scheme. Traditional numerical schemes invoke ray tracing methods to solve the eikonal equation in a Lagrangian frame of reference. Ray tracing performs well in many cases and handles the multivalued solution inherently. However, in long range simulations or simulations with complicated boundary conditions, ray solutions may perform poorly as rays are very sensitive to small perturbations in initial conditions. In the past few decades, several methods for solving the eikonal equation in a fixed (Eulerian) frame of reference have been devised. As with any computational method, each has its own advantages and disadvantages, but it is useful to be aware of alternatives since many problems in physical acoustics may benefit from having phase solutions computed on a fixed grid. We present a brief review of recent Eulerian approaches and discuss the merits and open issues with each. [Work supported by The Science, Mathematics, and Research for Transformation (SMART) Scholarship for Service Program.]
Inverse methods for Green’s function retrieval
Authors: Collier Sandra L. , Cain Jericho E. , Noble John M. , Ligon David A. , Alberts W. C. Kirkpatrick , Sim Leng
Published: May 2016

The Green’s function, or medium impulse response, provides information on the propagation channel. Here, we develop different inverse methods, based on time-reversal and flow-reversal methods, to retrieve the Green’s function using audible acoustic sources. These methods are then applied to experimental data collected using four 12-element, tri-axis, microphone arrays. Initial findings indicate that Green’s function retrieval is feasible.
Investigation of the relationship between cochlear gain reduction and speech-in-noise performance at positive and negative signal-to-noise ratios
Authors: Milvae Kristina D. , Alexander Joshua M. , Strickland Elizabeth A.
Published: May 2016

The ability to understand speech in noisy environments is vital to effective communication, but the auditory mechanisms driving this ability are not well understood. The auditory system may decrease the response to stimulated frequency regions so that changes from the acoustic environment are emphasized and information transfer is maximized. A known physiological mechanism that may contribute to this ability is the medial olivocochlear reflex (MOCR). The MOCR reduces cochlear gain via a neural feedback pathway between the brainstem and cochlea. A forward masking paradigm was used to examine cochlear gain reduction of a 2-kHz tone, an important frequency for speech perception. Speech-in-noise performance was also measured for the same participants at positive and negative signal-to-noise ratios (SNRs). Correlational analysis was used to examine the relationship between cochlear gain reduction and speech-in-noise performance across SNR. It was hypothesized that the correlation would depend on SNR, because turning down the gain should only improve performance when the overall response to the noise is decreased more than the response to the speech. Results and implications will be discussed. [Research supported by NIH(NIDCD) T32 DC000030.]
Evaluation of a two-interval, observer-based behavioral procedure for assessing detection performance in 2- to 4-year-old children
Authors: Leibold Lori , Bonino Angela Y. , Corbin Nicole , Buss Emily
Published: May 2016

Collecting reliable behavioral data from toddlers and preschoolers is challenging, limiting our understanding of human auditory development for these age groups. This study evaluated an observer-based procedure for measuring detection thresholds using a two-interval, two-alternative forced-choice paradigm. Listeners were normal-hearing adults and 2- to 4-year-olds. Detection performance was assessed in one of three conditions: (1) 1000-Hz warble tone in quiet; (2) disyllabic word in speech-shaped noise; or (3) disyllabic word in two-talker speech. Listeners were trained to perform a play-based, motor response whenever they heard a signal (e.g., putting a block in a bucket). An experimenter observed the child’s behavior and judged whether the signal was presented during the first or second observation interval; the experimenter was blinded to the true signal interval, so this judgment was based solely on the child’s behavior. Following training, signal level was adjusted to estimate 71% correct detection. The masker, when present, was played continuously throughout testing at a level of 55 dB SPL. The yield rate was 100% for adults and >80% for children. Good reliability was seen within a track and across testing sessions for individuals, indicating the two-interval procedure is feasible and reliable for use with toddlers and preschoolers.
Effect of asymmetrical organ of Corti mechanics on cochlear fluid pressure patterns
Authors: Zhou Wenxiao , Nam Jong-Hoon
Published: May 2016

In the cochlea, the sensory epithelium called the organ of Corti (OC) vibrates due to hydrodynamic pressures. The cochlear fluid pressure can be divided into differential and average components. The differential fluid pressures across the elastic OC complex (the OC, the tectorial membrane, and the basilar membrane) generate slow traveling waves toward the apex of the cochlea. Theoretical studies have assumed that the differential pressure component is relevant to cochlear mechano-transduction because the average component cannot vibrate the OC complex. This assumption has an implication that the entire section of the OC complex vibrates in synchrony. That is, the basilar membrane can represent the sole fluid-interacting surface of the OC complex. However, several recent studies considered that the active motility of the outer hair cell can break the synchrony between the top and the bottom fluid-interacting surfaces of the OC complex. Using a computational cochlear model with fully deformable OC complex, we present how different vibration modes of the OC complex can affect the cochlear fluid pressure patterns. The intra-cochlear fluid pressure patterns of our model are compared with recent measurement results. A condition under which the average pressure component can affect cochlear mechano-transduction is discussed.
A discrete time parametric model of auditory filters at peak sensitivity
Authors: Sullivan Christopher L. , Allen Jont
Published: May 2016

The first stage of human auditory processing is the filtration performed by the ear. The middle ear, basilar membrane, and tectorial membrane contribute to a filter bank specialized for amplitude and frequency selectivity. Accurately modeling this system is of critical importance, because the properties of the cochlear filter bank determine what audio information is available to humans. Because the cochlea is such an intricate and non-linear system, many cochlear models are computationally taxing and unusable for real time applications. By making a judicious linear approximation of the system, digital filters modeled after the cochlea can be implemented efficiently in real time. However, the digital filter models presently in use (often eighth order gammatone or elliptic filters) do a poor job of capturing the most perceptually critical properties of the cochlear filter bank. In this report, a set of parametric digital filters are proposed, which accurately model cochlear filters at peak sensitivity from 80 Hz to 19.5 kHz best frequency. The filters were designed using a gradient of steepest descent method to fit target frequency responses generated by a physical model of the cochlea. The final result is a set of polynomial equations which describe the locations of the poles and zeros of the digital cochlear filter approximations as a function of normalized cochlear best place.
Binaural pitch averaging and dominance trends in cochlear implant users
Authors: Oh Yonghee , Reiss Lina A.
Published: May 2016

Both cochlear implant (CI) and hearing aid (HA) users exhibit broad binaural pitch fusion, the fusion of stimuli differing in pitch across ears by as much as 3 octaves (Reiss et al., 2014, ARO 2015). Typically, two distinct trends are observed for the fused binaural pitch: dominance by the pitch perceived in one ear, or averaging of the pitches perceived in the two ears. In this study, detailed fusion pitch weighting trends as a function of pitch difference were investigated in both bimodal and bilateral CI users. Weights depended on the pitch differences, with averaging occurring for tones or electrodes closer in pitch and dominance occurring for tones or electrodes farther apart in pitch. The averaging region was typically 0.3–0.8 octaves for bimodal CI users and 1–3 electrodes for bilateral CI users around the reference, with weighting bias, in some cases, toward the ear with greater stimulus variability. Overall results show that the fusion pitch trends of the CI users were similar to those observed previously in HA users (Oh and Reiss, submitted); however, CI users showed greater inter-subject variability in both pitch weighting ranges and weighting biases. These findings suggest that binaural pitch averaging could be a potential underlying mechanism of binaural interference in hearing-impaired listeners. [Research supported by NIH-NIDCD grant R01 DC01337.]
Temporal structure processing reflected by mismatch negativity
Authors: Zhao T. C. , Kuhl Patricia K.
Published: May 2016

Temporal information comprises significant aspects of complex sounds and is hierarchically organized. Using music, we examined higher-level temporal structure processing. While the note-to-note intervals are variable, the underlying isochronous beats can be realized reliably. Temporal structure processing (i.e., meter) involves further grouping of the beats. Yet, the effects of tempo (beats/min) and meter type have not been examined systematically. In this experiment, electrophysiological methods were adopted to examine how duple and triple meters are processed in different tempos pre-attentively. Using an oddball paradigm, temporal structure (Duple versus Triple), established through alternating strong and weak tones, was violated occasionally (16%). The tempo varied across conditions (Easy versus Hard). The neural sensitivity to this violation is quantified by the magnitude of mismatch negativity (MMN), using EEG. Repeated ANOVA 2 (tempo: Easy versus Hard) x 2 (structure: Duple versus Triple) examined the effects on data from 17 adults (7 males, mean age = 27.07) and results showed that the MMN was significantly larger in Duple than Triple in the Easy condition, while equal between meters in the Hard condition. The results suggest that in an easier tempo, duple structure is better tracked than triple. However, this duple-meter advantage is overridden in difficult tempos.
Extent of lateralization of click trains elicited by interaural time differences—A comparison of measurement paradigms
Authors: Baumgaertel Regina M. , Dietz Mathias
Published: May 2016

Extent of lateralization measurements have previously been performed for various stimulus types, mainly using two measurement paradigms: (1) an acoustic pointer paradigm where the interaural level difference of a pointer stimulus is adjusted to match the intracranial position of a target stimulus and (2) a visual pointer paradigm where the intracranial position of a target stimulus is indicated on a linear scale. Both measurement paradigms are widely used, but rarely within the same study. In this study, the extent of lateralization elicited by unfiltered as well as 4 kHz bandpass-filtered click trains carrying interaural time differences (ITDs) up to 3 ms was measured using an acoustic as well as a visual pointer paradigm. Data from six young normal-hearing listeners were compared. Using the visual pointer paradigm, the extent of lateralization behavior as a function of stimulus ITD for unfiltered as well as bandpass filtered click trains coincided. The extent of lateralization as determined by the acoustic pointer paradigm, however, showed systematic differences at ITDs between 0.2 ms and 1 ms: filtered click trains were matched to smaller pointer ILDs than unfiltered click trains. Caveats and usability of both measurement paradigms will be discussed.
Spatial release from masking in listeners with asymmetric hearing thresholds
Authors: Freyman Richard L. , Boothroyd Derina S. , DeMaio Decia A.
Published: May 2016

Spatial release from masking was evaluated as a function of target-masker spatial separation in listeners with normal hearing and with chronic asymmetric hearing loss. The target was one of 20 vocoded words spoken by a female talker. The masker was a stream of vocoded nonsense sentences spoken by two other female talkers. Subjects listened in an anechoic chamber with loudspeakers at 1.9 m distance. The target was always presented from directly in front while the masker was presented from the front or from left or right at varying angular separations. In some conditions a copy of the masker with a delay of 4 ms was introduced at the front loudspeaker, simulating a reflection. Subjects detected the target in an adaptive 4AFC task. Results showed strong indications of informational masking in the co-located condition and spatial release that increased with masker angular separation, reaching almost 27 dB at wide separations in normal hearing listeners, and 22 dB even with the reflection. Listeners with asymmetric thresholds showed high across-listener variability in the spatial conditions but often smaller and strongly asymmetric release, presumably due to head shadow. The reflection reduced this head shadow advantage, and in one subject almost completely obliterated spatial release.
Method and application of auditory-visual attention training
Authors: Czyzewski Andrzej , Kostek Bozena , Gorski Adam
Published: May 2016

The main idea underlying the proposed attention training is to perform stimulation of the hearing and sight senses employing digital signal processing algorithms controlled by electroencephalography signals. The auditory and visual stimuli are designated to force the perception through hearing and sight senses by the appropriate hemisphere. The applied speech modification uses a non-uniform real-time speech stretching algorithm. The video content retrieval showing the speaker's face is slowed down, accordingly. Research experiments employed subjects with central auditory and visual processing disorders revealing severe communication difficulties. The effectiveness of the proposed method has been shown using formal attention focus tests. It was demonstrated that the proposed method of attention training helps improve speech understanding and reading skills in examined subjects. [Research sponsored by the Polish National Science Centre, Dec. No. DEC-2014/15/B/ST7/04724.]
The trade-off relationship between spatial tuning and multipulse integration in human subjects with cochlear implants
Authors: Zhou Ning
Published: May 2016

Guinea pigs studies have shown a correlation between the slope of the function relating psychophysical detection thresholds to pulse rate (known as multipulse integration) and the spiral ganglion cell density in electrical stimulation. If this relationship holds true in human subjects with cochlear implants, we hypothesize that multipulse integration should reduce (i.e., become shallow sloped) at stimulation sites with sharp tuning or when electrode configuration changes from broad (monopolar) to focused stimulation (bipolar). Multipulse integration was measured at all stimulation sites using two pulse rates with phase duration of 75 us in a monopolar stimulation mode (MP1 + 2) and bipolar stimulation mode (BP + 0). At selected stimulation sites, a psychophysical spatial tuning curve was also measured using a forward-masking paradigm. The multipulse integration slopes were steeper with MP electrode configuration than with BP. In MP mode, the multipulse integration slopes were steeper at stimulation sites with broader tuning. These results are consistent with the idea that multipulse integration is dependent on the number of available neurons. The degree to which the slopes were affected by focused stimulation, which measured local neural survival near the stimulation sites, was highly correlated with the subjects’ speech recognition performance.
Identification and localization of concurrent speech signals in an auditory dual task
Authors: Iyer Nandini , Thompson Eric R. , Simpson Brian D.
Published: May 2016

In Air Force operations, there is often a competing demand for auditory attention: listeners must identify and repeat a critical message, while also detecting and localizing the presence of a threat. Previous research (Iyer et al., 2014) has shown significant dual task interference, so that the ability of listeners to maintain a high level of performance on a primary task (keyword identification) was offset by decreased performance in a secondary task (critical call sign detection), relative to the respective single-task performances. The current experiment was designed to examine dual-task performance when the secondary task was changed from detection to localization. Listeners identified color-number keywords originating at 0o azimuth (primary task), while they localized the presence of a critical call sign originating from one of thirty-two locations surrounding them in 360° on the horizontal plane. The difficulty of the primary task was varied by employing an N-back task (0, 1, or 2) to increase memory load. Results indicated that the cost of dividing attention between the two tasks increased as the memory load in the primary task increased from 1-back to 2-back. More interestingly, while localization accuracy was unaffected, listeners’ ability to recognize the critical call sign degraded significantly as the difficulty of the primary task increased. The results have important implications for spatial auditory displays in multitalker environments.
Using sociolinguistic phonetic perception to fine tune cochlear implant simulations
Authors: Winn Matthew
Published: May 2016

Simulations of the sound of cochlear implants (CIs) are used for people with acoustic hearing for counseling, pedagogy, and to rigorously control stimulus parameters in predictions of CI performance. The most commonly used CI simulation is the eight-channel noise vocoder, which approximates intelligibility scores in better-performing CI users. However, this method underestimates the ability of CI users to demonstrate subtle effects in phonetic perception, including the adjustment of phonetic perceptions based on talker gender (phonetic “accommodation”), which CI users can do, despite no such effect found in normal-hearing listeners when using the eight-channel noise vocoder. Toward the goal of simulating CI performance more accurately, gender-driven phonetic accommodation was measured using three styles of vocoder, including an all-channel (conventional style) noise vocoder, and a peak-picking noise vocoder that delivers the top eight channels out of 22 per time bin. Carriers were variable-bandwidth noise or a pulsatile harmonic complexes, each delivered with multiple levels of spectral resolution. CI-like phonetic accommodation effects emerged in varying degree for all vocoders except the conventional eight-channel noise vocoder, suggesting that, despite good match of overall intelligibility, it is limited in its ability to predict more subtle aspects of speech perception.
Evaluation of frequency-dependent hearing loss by words identification
Authors: Chen Jing , Yang Hongying , Wu Xihong
Published: May 2016

Pure-tone audiometric test can reflect decrease of auditory sensibility at certain frequencies for the hearing loss; however, it requires critical environment and not suitable for the self-test in general environment, e.g., for some related applications on smart equipments. This study was aimed to evaluate the frequency-dependent hearing loss by words identification tests. Thirty-nine word-pairs in Mandarin Chinese were selected according the vowels/consonants confusion matrix. Every word is a monosyllable with CV (consonant-vowel) structure, and in each pair, the two words are easily confused by their formants (the same consonant conjunct with two different vowels) or the formant transitions (two different consonants conjunct with the same vowel). Hearing loss was simulated by band-stop filtering with different center frequencies. Eight filtering conditions, corresponding to eight center frequencies of the band-stop filter, were tested for nine normal hearing young students. The results show the word identification was significantly decreased for one certain filtering condition (worst-performance frequency, WPF) for 16 word-pairs, indicating these word-pairs could be effective to evaluate the frequency range of hearing loss. The relation of the WPF and the formant/formant transition frequencies were analyzed and discussed.
Spectral-normalization filter for subjective analysis of the aging voice
Authors: Berardi Mark L. , Hunter Eric J. , Gee Kent L.
Published: May 2016

Voice quality changes with age. One way of identifying voice quality changes and the possible physiological degeneration related to vocal function is through perceptually estimating talker age and then correlating estimated talker age with acoustical analysis. The most common perceptual studies investigating estimated talker age are cross-sectional. While these studies are useful, longitudinal studies provide additional details in the progressive degeneration of a single speaker’s (or group of speakers’) voice quality. Nevertheless, one limitation of longitudinal studies is that perceptual ratings of voice quality or talker age could be biased by recording quality (e.g., the spectral qualities of recordings from earlier decades are limited by the technology used). In this paper, a spectral-normalization filter was developed and applied to a corpus of recordings from an individual spanning about 50 years (1959–2007). The filter was shown to be effective in normalizing the autospectra of the recordings. The fundamental frequency was unaffected by the filter, suggesting that the core characteristics of the voice were unchanged. Preliminary subjective analysis suggests that, in spite of significant differences in the original recording equipment, the recording quality of all the files was perceptually similar.
Individual differences in self-adjustment of hearing aid amplification
Authors: Perry Trevor T. , Nelson Peggy , VanTasell Dianne , Gregan Melanie
Published: May 2016

Listening in noisy environments remains a common complaint among hearing aid users. Self-adjustment of amplification parameters could improve listener satisfaction in noise. Previous data (Nelson et al., AAS 2015) indicated considerable variability among hearing-impaired listeners when self-adjusting amplification in noisy environments, but the sources of variability are not understood. Data will be presented from 30 adult listeners with mild to moderate hearing loss who self-adjusted amplification parameters in laboratory-simulated restaurant environments. Participants listened using a real-time simulation of a multichannel compression hearing aid and adjusted gain/compression parameters via a simple user interface (EarMachine). On average, listeners selected less gain relative to NAL-NL2 targets, and they reduced gain as noise levels increased. Hearing loss did not predict variation in self-selected amplification. Small but significant gender differences were observed. Female subjects selected more high-frequency gain (2000–8000 Hz) but less low-frequency gain than male subjects. Listeners with prior hearing-aid experience selected more high-frequency gain than listeners with no hearing-aid experience. Younger age was associated with selecting more low-frequency gain. Within-group variability was large compared to mean differences between groups and only a small proportion of variance was explained. These data suggest that self-adjustment of amplification was not easily predicted by listener-specific factors.
Sound system Déjà Vu—A chance to revisit the design approach in an acoustically challenging multi-purpose space
Authors: Britton Deb
Published: May 2016

More than a decade ago, we were tasked with designing a sound reinforcement system that would provide great speech intelligibility, in a highly reverberant space, without modifying any of the architectural finishes. The sound system had to support different types of events, held in different locations within the space, and with varying audience sizes. In 2015, we were asked to revisit that original loudspeaker design, and upgrade the sound system in that same space. This paper presents a case study of both upgrades, and describes the challenges, the lessons learned, and the changes in approach from the original upgrade to the most recent sound system design.
A presentation on the Infocomm spectral balance standard
Authors: Murray John A.
Published: May 2016

InfoComm is finalizing a Spectral Balance Standard soon to be released for public review. The standard is not a “how to” for system equalization; it only verifies properly equalized systems. It defines a validation process that ensures sound systems reproduce a relatively uniform spectral balance for (3) classes of systems: full-bandwidth, limited bandwidth, and voice-communications. As conformance to the Standard confirms optimum spectral performance, this makes it an integral part of a complete optimization suite. And, as a part of system optimization, this makes the Spectral Balance Standard relevant to the Sound System Design, Optimization, and Intelligibility ASA Special Session. The measurement portion divides the audio spectrum into two sections: impulse-response measurements above the modal range of each room and steady-state spatially averaged measurements in and possibly below the modal range. The two measurement types are conjoined for each measurement position and a range of variation limits and fractional-octave smoothing criteria, specific to each class of sound system, are applied. Sound system responses that fit within the limits conform to the standard. This presentation will discuss the overall philosophical approach towards a spectrally balanced sound system that drives the specifics of the measurements, limits, and criteria chosen for the Standard.
A review of sound reinforcement system design and applications as continually changing over the years through the introduction of new design techniques, equipment, and commissioning
Authors: Coffeen Robert C.
Published: May 2016

Since the author's first commercial sound system design in 1960 and the first use of this system in 1962 system design techniques, audio equipment, and methods used for system test, adjustment, and equalization have been continually changing. Most of these changes have been good and have resulted in better performing sound reinforcement systems. However, a few of the changes have not contributed to improved system performance. This presentation will review the following changes: in design methods such as techniques for determining loudspeaker sound distribution in a particular venue, in audio equipment including the use of digital signal processing and audio signal transmission, in loudspeaker directivity both better and worse, in the introduction of new microphone types and the disappearance of several useful microphone types, and in system commissioning including equalization from the days of passive filters to digitally produced filters. The presentation will include several microphone and equalization demonstrations.
Dick Fay, man of many tastes
Authors: Yost William
Published: May 2016

Hearing in Vertebrates: A Psychophysics Databook and Hot Dog Chicago seem like fitting bookends for the amazing career of Richard (Dick) R. Fay. Both books were researched and written in the 1980s, a fertile time in Dick’s career. The Databook was written by Dick and Hot Dog Chicago with his friend and colleague at Loyola University Chicago, Rich Bowen (Rich is a Professor of Psychology studying visual psychophysics). The Databook is unique, representing a scholarly tour-de-force which has become the ‘”bible” for those interested in vertebrate behavior and hearing; and Hotdog Chicago is an informative Chicago fast-food restaurant guide written with a whimsical sense of humor and read by hundreds, if not thousands, interested in an iconic Chicago culture. In the years-long preparation and production of the Databook, Dick arrived at a new view of the advantages of the comparative study of auditory processing. This talk will describe Dick’s efforts in writing these two books and how those efforts helped shape him as a scholar and friend of so many.
Psychological and physiological acoustics: The encoding and interpreting of sound energy by biological systems
Authors: Gallun Frederick J.
Published: May 2016

Hearing science, or the study of how the auditory system functions and responds to sound in humans and other animals, is a topic that has been studied using modern scientific methods for nearly 200 years. The current approaches to hearing science are highly multidisciplinary, integrating acoustics with many fields, such as psychology, biology, engineering, computer science, statistics, and audiology. The study of the encoding and perception of simple and complex sounds means that there is considerable overlap with other technical areas of the Acoustical Society of America, including architectural, speech, music, and animal bioacoustics. Hearing scientists are also concerned with hearing disorders, vibrotactile and vestibular sensation, the interaction of hearing with other sensory modalities, development, and aging, as well as learning and plasticity effects in auditory function. This talk will highlight (1) the various brain areas involved in auditory processing and (2) the technological innovations in hearing aids and cochlear implants that have made sound processing available to many people who, for most of human history, would have been either completely deaf or would have had only minimal awareness of sound.
Dynamic road traffic density estimation employing noise mapping with the use of grid supercomputing
Authors: Czyzewski Andrzej , Marciniuk Karolina , Kostek Bozena
Published: May 2016

A noise prediction model of a large city agglomeration was elaborated in order to allow for a dynamic road traffic density estimation in vehicular networks. The implemented application adopts the model fed with traffic noise data based on frequently refreshed LDEN levels. Calculations were made with the use of the numerical model developed for his purpose and then implemented on the PL-Grid supercomputing infrastructure. Data obtained through supercomputing and through the use of a standard noise map computing software were collated with measured levels acquired from the acoustic city monitoring system and then analyzed. The comparison performed afterwards shows a relatively good accuracy of the developed model. The numerical model of traffic noise and its main sources are briefly characterized. A full day dynamic noise map can be browsed as a set of 24 noise maps, one for each hour of the day which in turn allows for vehicular traffic density estimation based exclusively on acoustical data. [Research was subsidized by the Polish National Centre for Research and Development within the Grant No. OT4-4B/AGH-PG-WSTKT.]
Probability of receiving a blast noise complaint: Spatial analysis of complaint behavior around military installations
Authors: Morrison Dawn A. , Gonzalez Jose , Nykaza Edward T.
Published: May 2016

This study explores the probability of receiving a blast noise complaint in response to military training noise. Community noise complaints are an ongoing impediment to military training and often result in training curfews and restrictions. As such, we want to understand where, by whom, and under what conditions complaints are used as a coping strategy in response to annoyance caused by military training noise. We build upon previous research conducted (Nykaza et al., 2012), incorporating physical and human geography into the analysis to identify the situational context of the population subset that complains about military noise. Using both spatial and statistical analysis, we look at land use (LULC), census and socio-economic housing data paired with complaint and noise monitor data collected around one US Military Installation, to examine who (i.e., what subset of the population) is more likely to use complaint behavior as a coping strategy. Our results support the development of a noise complaint-forecasting model that military range managers can use to improve the coordination, communication, and relationships with surrounding communities.
Estimates of residential floor vibration induced by sonic booms
Authors: Klos Jacob
Published: May 2016

Future civil supersonic aircraft are expected to produce sonic boom noise within overflown communities that is substantially quieter than previous supersonic aircraft, which may allow regulators to replace the current prohibition on civil supersonic flight with a noise-based certification standard. In support of standards development, NASA has used an indoor simulation laboratory to investigate people’s reaction to combinations of transmitted sonic boom signatures and related floor vibration levels. Given the limited extent of experimental data, numerical models were used to estimate the expected range in floor vibration level that people may experience within a large assortment of houses for various low-amplitude sonic booms from different aircraft. This presentation will discuss the predictive models, compare predicted floor vibration to measurements from field experiments in similar houses, and document the range of predicted floor vibration that informed a recent psycho-acoustic laboratory study. The analyses of subjective annoyance from this laboratory study are discussed in a companion presentation (Rathsam and Klos).
Vibration penalty estimates for indoor annoyance caused by sonic boom
Authors: Rathsam Jonathan , Klos Jacob
Published: May 2016

Commercial supersonic flight is currently forbidden over land because sonic booms have historically caused unacceptable annoyance levels in overflown communities. NASA is providing data and expertise to noise regulators as they consider relaxing the ban for future quiet supersonic aircraft. One key objective is a predictive model for indoor annoyance based on factors such as noise and indoor vibration levels. The current study quantified the increment in indoor sonic boom annoyance when sonic booms can be felt directly through structural vibrations in addition to being heard. A shaker mounted below each chair in the sonic boom simulator emulated vibrations transmitting through the structure to that chair. The vibration amplitudes were determined from numeric models of a large range of residential structures excited by the same sonic boom waveforms used in the experiment. The analysis yielded vibration penalties, which are the increments in sound level needed to increase annoyance as much as the vibration does. For sonic booms at acoustic levels from 75 to 84 dB perceived level, vibration signals with lower amplitudes ( + 1 sigma) yielded penalties from 0 to 5 dB, and vibration signals with higher amplitudes ( + 3 sigma) yielded penalties from 6 to 10 dB.
Two weeks in the woods: Noise from a natural-gas compressor station
Authors: Gabrielson Thomas B.
Published: May 2016

Natural gas development is permitted on Pennsylvania State Forest land; however, the Pennsylvania Bureau of Forestry is tasked not only with stewardship of natural resources but also with maintaining a natural environment for wildlife and recreational users of the land. As a precursor to human-impact studies of gas-compressor noise, a two-week long series of measurements was made near a natural-gas compressor station in the Tiadaghton State Forest in north-central Pennsylvania. Although several standard metrics were determined, the intent was to collect and analyze full time-series data at a number of different distances and directions from the station. Recordings were made in two bands: 5 to 10 000 Hz with 5 min of recording every hour for two weeks and 0.1 to 250 Hz recorded continuously for one week. Analyses of the narrowband spectral densities and of the spectrograms of the noise reveal specific source mechanisms and source variability. The high-fidelity recordings will be used subsequently both for listening tests with human subjects and exposure tests for wildlife. [Work supported by the Pennsylvania Department of Conservation and Natural Resources, Bureau of Forestry.]
Soundlapse: audio strategies for documenting soundscapes with timelapse photography
Authors: Carter J. Parkman
Published: May 2016

Timelapse photography—the procedure of taking photographs at regular intervals and then sequencing them together to produce a compressed portrayal of time passing—is widely popular across diverse areas such as scientific research, weather tracking, filmmaking, and even reality TV shows, which commonly use timelapse footage to segue between scenes and reflect the passing of time. The compelling visual effect and broad legibility of timelapse photography makes it a powerful tool for revealing flows, shifts, and patterns in the environment which would otherwise be too subtle to perceive in real time. One of the most important challenges in documenting the salient features of a soundscape is revealing similar shifts and patterns across time in the acoustic environment. However, temporal logic would suggest it is simply not possible to sequence such brief samples of the audio environment with any meaningful result. This project explores different audio capture and editing techniques to produce viable strategies for documenting the changes in a soundscape which accompany timelapse photography of the environment under study. Examples will be demonstrated, emphasizing the significant variables for adequately representing the elusive temporal characteristics of a soundscape.