bluuebunny/crossref_metadata_2025 huggingface dataset:
Sparse coding for scaled bioacoustics: From Humpback whale songs evolution to forest soundscape analyses
Authors: Glotin Herve , Sueur Jérôme , Artières Thierry , Adam Olivier , Razik Joseph
Published: May 2013

The bioacoustic event indexing has to be scaled in space (oceans and large forests, multiple sensors), and in species number (thousand). We discuss why time-frequency featuring is inefficient compared to the sparse coding (SC) for soundscape analysis. SC is based on the principle that an optimal code should contain enough information to reconstruct the input near regions of high data density, and should not contain enough information to reconstruct inputs in regions of low data density. It has been shown that SC methods can be real-time. We illustrate with an application to humpack whale songs to determine stable components versus evolving ones across season and years. By sparsing at different time scale, the results show that the shortest humpack acoustic codes are the most stable (occurring with similar structure across two consecutive years). Another illustration is given on forest soundscape analysis, where we show that time-frequency atomes allow an easier analysis of forest sound organization, without initial classification of the events. These researches are developed within the interdisciplinary CNRS project “Scale Acoustic Biodiversity,” with Univ. of Toulon, Paris Natural History Museum, and Paris 6, consisting into efficient processes for conditioning and representing relevant bioacoustic. Information, with examples at sabiod.univ-tln.fr.
The effect of surfactant shedding and gas diffusion on pressure wave propagation through an ultrasound contrast agent suspension
Authors: O'Brien Jean-Pierre , Ovenden Nick , Stride Eleanor
Published: May 2013

Interest in microbubbles as agents for therapeutic and quantitative imaging applications in biomedical ultrasound has increased the need for their accurate modeling. However, effects such as gas diffusion, the properties of the shell, and changes in bubble behavior under repeated exposure to ultrasound pulses are still not well understood. A revised equation for microbubble motion is proposed that includes the effects of gas diffusion as well as a nonlinear surface tension, which depends on a non-constant surfactant surface concentration. This is incorporated into a nonlinear wave propagation model to account for these additional time-dependent effects in the response of microbubble contrast agent populations. The results from the model indicate significant changes in both bubble behavior and the propagated pulse compared with those predicted by existing propagation models; and show better agreement with experimental data. Our analysis indicates that changes in bubble dynamics are dominated both by surfactant shedding on ultrasonic timescales and gas diffusion over longer timescales between pulses. Therefore, incorporating such time-dependent phenomena in ultrasound imaging algorithms should lead to better quantitative agreement with experiments.
Rectified growth of histotripsy bubbles
Authors: Kreider Wayne , Maxwell Adam D. , Khokhlova Tatiana D. , Simon Julianna C. , Khokhlova Vera A. , Sapozhnikov Oleg A. , Bailey Michael R.
Published: May 2013

Histotripsy treatments use high-amplitude shock waves to fractionate tissue. Such treatments have been demonstrated using both cavitation bubbles excited with microsecond-long pulses and boiling bubbles excited for milliseconds. A common feature of both approaches is the need for bubble growth, where at 1 MHz cavitation bubbles reach maximum radii on the order of 100 microns and boiling bubbles grow to about 1 mm. To explore how histotripsy bubbles grow, a model of a single, spherical bubble that accounts for heat and mass transport was used to simulate the bubble dynamics. Results suggest that the asymmetry inherent in nonlinearly distorted waveforms can lead to rectified bubble growth, which is enhanced at elevated temperatures. Moreover, the rate of this growth is sensitive to the waveform shape, in particular the transition from the peak negative pressure to the shock front. Current efforts are focused on elucidating this behavior by obtaining an improved calibration of measured histotripsy waveforms with a fiber-optic hydrophone, using a nonlinear propagation model to assess the impact on the focal waveform of higher harmonics present at the source's surface, and photographically observing bubble growth rates. [Work supported by NIH EB007643 and DK43881; NSBRI through NASA NCC 9-58.]
Small directional microelectromechanical systems microphone arrays
Authors: Elko Gary W.
Published: May 2013

Directional microphone arrays that are physically small compared to the acoustic wavelength are of great interest for hand-held communication devices. Spatially directive microphones can reduce the impact of background acoustic noise without adding distortion to the signal. This talk will present some design topologies and requirements as well as a new physical design that could enable directional microphone responses while being small in size.
Consonant confusability and its relation to phonological dissimilarity
Authors: Khan Sameer ud Dowla
Published: May 2013

Consonant similarity can be measured indirectly through a language’s phoneme inventory, lexicon (e.g. cooccurrence restrictions), or phonology (e.g., processes that take similarity or dissimilarity into account). It can also be measured more directly as confusability in a perception task. Thus far, consonant similarity in Bengali has only been measured indirectly, through the inventory, lexicon, and phonology. Previous studies [Khan (2006)] claim that Bengali speakers judge the similarity of consonants in echo reduplication, where the initial consonant of the base is systematically replaced with a phonologically dissimilar consonant in the reduplicant, e.g., kashi “cough” > kashi-tashi “cough, etc.” but thonga “bag” > *thonga-tonga > thonga-fonga “bags, etc.”). This measurement of similarity assumes a set of features assigned language-specific weights; for example, [voice] is weighted more heavily that [spread glottis], to explain why speakers treat the pair [t, th] as more similar than the pair [t, d]. But does the measurement of similarity inherent in the echo reduplicative construction correspond directly to the relative perceptibility of different consonant contrasts? The current study examines data collected in a perception experiment, comparing the relative confusability of Bengali consonants produced in noise with the claims of phonological notions of similarity associated with echo reduplication.
Regional linguistic variations in Canadian French: Do they affect performance on speech perception in noise?
Authors: Lagacé Josée , Breau-Godwin Stéphanie , Giguère Christian
Published: May 2013

Many audiologists working with a Canadian French population use word recognition tests to measure speech perception abilities in noise. The TMB test (“Test de Mots dans le Bruit”) includes four lists of 35 words presented in babble noise. The test is intended to measure the pre-cognitive perceptual stage of auditory processing and does not require understanding of the phonetic differences between speech sounds at a cognitive level. Previous studies examining performance on auditory tests similar to the TMB showed differences between populations speaking the same language but with different accentuations, such as the English spoken in the United States versus the United Kingdom. Variations in performance were attributed to accentuation differences between the speaker and the listener. To the authors’ knowledge, no study appears to have investigated the effect of regional linguistic variations of Canadian French on word recognition in noise. Normative data for the TMB are being collected in three regions of Canada: Moncton, Montréal, and Ottawa. Participants are all native speakers of Canadian French, but there are important linguistic variations across the three regions. Knowledge of the effect of regional linguistic variations on the TMB performance will help refine interpretation of test results in the audiology clinics.
An adaptive noise power spectral density estimation of noisy speech using generalized gamma probability density function
Authors: Dang Xin , Nakai Takayoshi
Published: May 2013

An estimation of the power spectral density (PSD) of noise is a crucial part to retrieve speech in a noisy environment. A novel estimation method for non-white noise of noisy speech on the basis of a generalized Gamma distribution is proposed. Because of highly non-stationary nature of speech, its probability density function (PDF) is difficult to derive using any modeling technique, while a segmental noise is more stationary and can be fitted more accurately by a generalized Gamma PDF, which is a natural extension of the Gaussian modeling of a non-white components distribution. In the experiment, different types of non-white noises are added to the clean speech signal at different SNRs to study the estimation of noise using different types of PDF. It is found that non-white noise spectrums fit more accurately on the generalized Gamma PDF with adaptive parameters instead of a Gaussian distribution function. The reported generalized Gamma PDF model shows the best performance to estimate the noise spectral amplitudes as compared with Minimum Statistics (MS), Speech absence Probability (SAP), and MMSE based PSD estimation methods. The performance of the proposed noise estimation is good when it is integrated with the speech enhancement technique as demonstrated by both the subjective and objective measures.
Markov random field in speech enhancement: Application for tonal languages
Authors: Saimai Tanawan , Tantibundhit Charturong , Onsuwan Chutamanee , Wutiwiwatchai Chai
Published: May 2013

This paper proposed speech enhancement algorithm based on Markov random field (MRF) model for Thai, a tonal language. Firstly, a noisy speech signal is transformed using the short time Fourier transform (STFT). In so doing, noise is removed and speech is preserved, especially harmonics information as f0 patterns are relevant perceptual cues for lexical tones. The voice activity detector is used to classify each STFT time frame into voiced and unvoiced. Harmonics information is retrieved from each voiced time frame, where four neighborhoods of the analyzed STFT coefficients include its adjacent time frames (left, right) and nearest harmonics (top, bottom). For the unvoiced, four adjacent coefficients (left, right, top, and bottom) are used. A two-state MRF model is used to classify STFT coefficients into speech and noise. Those with speech state are retained, while the rest is set to zero. The enhanced speech is estimated by the inverse STFT. Results from quality evaluation test on four sets of Thai rhyming words corrupted by white noise at SNR levels of 0, 5, and 10 dB showed that the proposed algorithm significantly improved SNR of noisy speeches compared with spectral subtraction (1.3 dB on average) and Wiener filtering (1.9 dB on average).
Wind-dependence of low-frequency ambient noise in the deep-ocean sound channel
Authors: Nichols Stephen , Bradley David L.
Published: May 2013

In the low-frequency range (1–125 Hz), the deep-ocean ambient noise field is produced by seismic, marine life, ship traffic, and wind-dependent hydrodynamic noise mechanisms. This study focuses on the contribution of wind-related source mechanisms to the overall ambient noise field, as well as previous attempts to understand the physics of these mechanisms. The Comprehensive Nuclear-Test Ban Treaty Organization (CTBTO) hydroacoustic monitoring system has produced nearly continuous recordings of the low-frequency deep-ocean ambient noise field at sites in the Pacific, Atlantic, and Indian Oceans, each spanning several years in length. Additionally, wind speed data have been recorded at the host island of each station by the National Oceanic and Atmospheric Administration (NOAA). Correlation techniques are used with these two datasets to determine the relationship between wind speed and the sound level in different frequency bands, and to determine the prominence of wind-related noise in the combined ambient noise spectrum. Results from the three sites are compared to each other to assess the uniformity of wind-generated noise over the world’s ocean basins.
Towards the state of the art in virtual acoustics technology
Authors: Woszczyk Wieslaw , Ko Doyuen , Hong Jonathan
Published: May 2013

Active acoustics enhancement systems have been applied in room acoustics for decades yet it would be hard to say that their ultimate performance has been achieved, even with the power of today’s fast digital processors. There are many challenges ahead including the creation of homogeneous directionless diffuse field using a limited number of discrete loudspeaker sources. There is also a need to create soundfield having some localized directional properties mimicking the phenomena found in acoustics of musical instruments interfacing the acoustics of rooms. Some solutions to these challenges will be presented and discussed by the authors.
Active field control using sound field generation technique—Case study of a Live concert at a virtual Renaissance church
Authors: Watanabe Takayuki , Ikeda Masahiro , Kim Sungyoung
Published: May 2013

In the Renaissance area, musical culture was centered at and spread across churches. In order to appreciate Renaissance music, therefore, it is important to account for the influence of acoustics of churches at that era so that audiences today can experience homogeneous musical appreciation. We used an Active Field Control system to create the acoustics using measured the impulse responses (IRs) of a church. The system consists of directional microphones, head amps, a convolution engine, a matrix processor, amplifiers, and loudspeakers. And it picks up the direct response of performance, convolves it with the measured IRs, and reproduces the resulting sound using loudspeakers around the room. Loudspeaker positions in the performance hall are equivalent to the positions where the IRs had been measured at the church. This technique allows us to convincingly recreate not only the reverberation time but also the spatial impressions of the church at the performance hall. In addition, we modified the IRs so that the inherent acoustical characters of the performance space would less influence to recreation of the target church acoustics. We used a 6-channel recording/reproduction system to evaluate the modification to the IRs and the impressions of the recreated acoustics in the room. This paper presents the results of the experiment.
Design and experience with subterranean installation of a fully anechoic acoustical testing chamber
Authors: Ellingson Roger M. , Helt Patrick V.
Published: May 2013

One of the purposes of a fully anechoic chamber is to provide a very quiet, near-echo-free environment simulating free-field acoustical conditions. From design specification to completion, installation of a fully anechoic chamber can be an enormous undertaking as compared to installation of conventional sound-attenuated acoustical test rooms, which generally are smaller in physical size and have less stringent sound attenuation requirements. The authors had the opportunity to specify the requirements and oversee the installation of a fully anechoic chamber designed to support near-full-frequency, human-hearing range acoustical experiments at the VA National Center for Rehabilitative Auditory Research (NCRAR), located in Portland, OR, USA. Design and installation of NCRAR’s chamber to support entry at laboratory floor level was complicated by the job site location, a subterranean area beneath a structure that accommodates clinical and research offices on the upper floors, and serves as a parking garage on the lower floors. In addition to ambient noise considerations, existing architectural restrictions included parking traffic patterns, below-grade earthquake beams, limited overhead clearance, ground water seepage, and storm water flow patterns. The purpose of this paper is to share pictorial-illustrated experiential results on specification, design, installation, and chamber operation as well as architectural considerations, site preparation, and construction detail.
Human performance in aural classification of sonar echoes
Authors: Allen Nancy , Hines Paul C.
Published: May 2013

Active sonar echoes from man-made objects and those from naturally occurring features in certain coastal environments can be difficult to distinguish when using traditional sonar processing techniques and visual displays. An approach being investigated at Defence R&D Canada-Atlantic for addressing this challenge is to exploit the capability of human hearing for discriminating between these two classes of echoes. Part of the work consisted in producing a human-performance baseline. Two human listening tests were designed and carried out. Both used stimuli that were generated from a sample of echoes recorded during actual sonar experiments at sea. In the second test, a 500-Hz high-pass filter was applied to the stimuli. Quantitative data from the rating-exercise portion of the tests were used to produce receiver-operating-characteristic (ROC) curves for modeling how well the participants could distinguish the two classes of echoes. For both tests, the results show that the listeners could hear differences, but performance was significantly better in the first test. Qualitative data collected from the questionnaire portion of the tests helped to interpret some of the performance results.
Temporal signal processing of dolphin biosonar echoes from fish prey
Authors: Au Whitlow , Ou Hui
Published: May 2013

Dolphins that emit whistle signals (except for sperm whales) project short broad- band biosonar clicks containing about 5 to 7 cycles with exponential decaying envelope and Q (center frequency over bandwidth) between 2 and 3. The broadband nature of the biosonar clicks allow for good temporal resolution of echo highlights, which in turn allows for the discriminations of different targets including fish prey. Most of the echoes from fish originate from signals reflecting off the swim bladder of fish. Different species of fish have swim bladders of different shape, size, and orientation so that echoes from these species can often be discriminated from temporal cues. The echoes contain many highlights as the signals reflect off different surfaces and parts of the fish body and swim bladder. This presentation will discuss the temporal characteristics of echoes from fish prey, which are highly aspect dependent and will discuss the six temporal parameters that were used in a support vector machine (SVM) to discriminate between species. Results suggest how dolphins can classify fish based on their echoes and provide some insight as to which features might enable the classification.
Auditory scene analysis in budgerigars (Melopsittacus undulatus) and zebra finches (Taeniopygia guttata)
Authors: Dent Micheal L. , Neilans Erikson G. , Flaherty Mary M. , Martin Amanda K.
Published: May 2013

Deciphering the auditory scene is a problem faced by humans and animals alike. However, when faced with overlapping sounds from multiple locations, listeners are still able to attribute the individual sound objects to their individual sound-producing sources. Here, we determined which characteristics of sounds are important for streaming versus segregating in birds. Budgerigars and zebra finches were trained using operant conditioning procedures on an identification task to peck one key when they heard a whole zebra finch song and to peck another when they heard a zebra finch song missing a middle syllable. Once the birds were trained to a criterion performance level on those endpoint stimuli, probe trials were introduced on a small proportion of all trials. The probe songs contained modifications of the incomplete training song’s missing syllable. When the bird responded as if the probe was a whole song, it suggests they streamed together the altered syllable and the rest of the song. When the bird responded non-whole song, it suggests they segregated the altered probe from the rest of the song. Results show that some features, such as spectrotemporal similarity and location, are more important for streaming than other features, such as timing.
Improving the acousto-optic detection of high-intensity focused ultrasound lesions
Authors: Adams Matthew T. , Wang Qi , Cleveland Robin O. , Roy Ronald A.
Published: May 2013

Real-time acousto-optic (AO) sensing has been shown to non-invasively detect changes in ex vivo tissue optical properties during high-intensity focused ultrasound (HIFU) exposures. Although proof-of-concept experiments have been successful, the underlying parameters and mechanisms affecting the AO detectability of HIFU lesion formation are not well understood. In this work, a modeling based approach is used to improve the AO sensing of lesion formation during HIFU therapy. The angular spectrum method is used to model the acoustic field from the HIFU source and the temperature field, due to the absorption of ultrasound, is modeled using a finite-difference time-domain solution to the Pennes bioheat equation. Wavelength specific changes in tissue optical properties are calculated using a thermal dose model, calibrated by experimental data. The diffuse optical field is modeled using an open-source graphics processing unit accelerated Monte Carlo algorithm. The Monte Carlo algorithm is modified to account for light-sound interactions, using the acoustic field from the angular spectrum method, and to account for AO signal detection. Results will demonstrate the important roles of optical wavelength selection, and illumination and detection configurations on the detectability of HIFU lesions by optical and AO sensing methods. [Work supported in part by NSF.]
Determination of tissue injury thresholds from ultrasound in a porcine kidney model
Authors: Wang Yak-Nam , Simon Julianna C. , Cunitz Bryan W. , Starr Frank L. , Paun Marla , Denny Liggitt , Evan Andrew P. , McAteer James A. , Williams James C. , Liu Ziyue , Kaczkowski Peter J. , Hsi Ryan S. , Sorensen Mathew D. , Harper Jonathan D. , Bailey Michael R.
Published: May 2013

Therapeutic ultrasound has an increasing number of applications in urology, including shockwave lithotripsy, stone propulsion, tissue ablation, and hemostasis. However, the threshold of renal injury using ultrasound is unknown. The goal of this study was to determine kidney injury thresholds for a range of intensities between diagnostic and ablative therapeutic ultrasound. A 2 MHz annular array generating spatial peak pulse average intensities (ISPPA) up to 30,000 W/cm2 in water was placed on the surface of in vivo porcine kidneys and focused on the adjacent parenchyma. Treatments consisted of pulses of 100 μs duration triggered every 3 ms for 10 min at various intensities. The perfusion-fixed tissue was scored by three blinded independent experts. Above a threshold of 20,000 W/cm2, the majority of injury observed included emulsification, necrosis, and hemorrhage. Below this threshold, almost all injury presented as focal cell and tubular swelling and/or degeneration. These findings provide evidence for a wide range of potentially therapeutic ultrasound intensities that has a low probability of causing injury. While this study did not examine all combinations of treatment parameters of therapeutic ultrasound, tissue injury appears dose-dependent. [Work supported by NIH DK43881, DK092197, and NSBRI through NASA NCC 9-58.]
Electret microphone modeling and optimization by combined finite element analysis and lumped-element techniques
Authors: Schafer David
Published: May 2013

Electret microphone performance modeling must represent detailed coupled mechanical, electrostatic, and fluid dynamics of the moving diaphragm, stationary back electrode and intervening air layer (the “motor”) while also evaluating the net performance characteristics which result. We model electret microphone response and noise by combined FEA and lumped-element methods. In detail, the diaphragm is treated as a tensioned membrane. Electrostatics are evaluated in locally parallel-plate fashion. Air layer dynamics are treated by reducing the exact thermoacoustic parallel-plate 3D squeeze solution to a 2D “transmission sheet” differential equation. Analyses (COMSOL FEA 2D PDE-mode) include initial diaphragm deflection upon back electrode polarization, then the harmonic responses to changes in frontside pressure, backside pressure, and polarization. Motor responses (frontside/backside volume flows and output current) are reduced to impedance and transfer characteristics of a three-port (front and back acoustic, electrical) element, which is then embedded in a traditional host lumped-element equivalent circuit (including amplifier). FEA-based motor characteristics over an array of frequencies lead to microphone frequency response and noise spectrum. Performance optimization is done by scripting motor geometry generation and FEA analysis as a subroutine, defining a space of adjusted parameters (motor dimensions, etc.) and running recurring FEA analyses generated by a downhill simplex algorithm.
A numerical study of non-collinear mixing of three-dimensional nonlinear waves in an elastic half-space
Authors: Sun Zhenghao , Li Hongguang , Li Fucai
Published: May 2013

Interactions of two non-collinear nonlinear ultrasonic waves in an elastic half-space with quadratic and hysteretic nonlinearity are investigated in this paper. The numerical problem is solved by a semi-discrete central scheme as well as a finite element method in two and three dimensions, respectively. Features and intensity distribution of the resonant wave are analyzed both in time and frequency domains, and the method of non-collinear wave mixing is proved to be a promising method, which is both effective and sensitive in material characterization and structure damage detection.
Learning acoustic phonetics by listening, seeing, and touching
Authors: Arai Takayuki
Published: May 2013

There is a huge volume of written textbooks available in virtually every modern field, including acoustic phonetics. However, in areas dealing with acoustics, learners often face problems and limitations when they deal with only written material and no audio or visual information. As one response to this problem, we have developed several sets of physical models of the human vocal tract and have shown that they are extremely useful for intuitive understanding. In addition, we also developed a tool called “Digital Pattern Playback.” Another solution is an online version featuring demonstrations. We are currently collecting materials, mainly in the form of sounds, for educational purposes in acoustics and phonetics and are releasing them as “Acoustic-Phonetics Demonstrations” through our Web site. These demonstrations are designed for students in linguistics, phonetics and phonology, speech pathology, audiology, psychoacoustics, speech engineering, and others. However, potential users are not limited to these groups, as we feel that a wide range of learners can obtain tremendous benefits from the demonstrations, including those who are studying foreign languages or patients undergoing speech articulation therapy. [Work partially supported by a Grant-in-Aid for Scientific Research (24501063) from the Japan Society for the Promotion of Science.]
Adaptive stabilized finite element framework for simulation of vocal fold turbulent fluid-structure interaction and towards aeroacoustics
Authors: Jansson Johan
Published: May 2013

As a step toward building a more complete model of voice production mechanics, we assess the feasibility of a fluid-structure simulation of the vocal fold mechanics in the Unicorn incompressible Unified Continuum framework. The Unicorn framework consists of conservation equations for mass and momentum, a phase function selecting solid or fluid constitutive laws, a convection equation for the phase function and moving mesh methods for tracking the interface, and discretization through an adaptive stabilized finite element method. The framework has been validated for turbulent flow for both low and high Reynolds numbers and has the following features: implicit turbulence modeling (turbulent dissipation only occurs through numerical stabilization), goal-oriented mesh adaptivity, strong, implicit fluid-structure coupling, and good scaling on massively parallel computers. We have applied the framework for turbulent fluid-structure interaction simulation of vocal folds, and present recent results. Initial steps toward aeroacoustics have been carried out in the framework (exhaust system and landing gear applications), and will be presented as well as the current state of aeroacoustic modeling for the human voice in the framework.
Can wind-turbine sound that is below the threshold of hearing be heard?
Authors: Schomer Paul
Published: May 2013

This paper is geared toward wind-turbine sound, but it is really a simple variation on the basic concepts that this author used in the development of loudness-level-weighted sound exposure (Schomer et al., J. Acoust. Soc. Am. 110(5, Pt. 1), 2390–2397 (2001)] and of Rating Noise Curves (RNC) [Schomer, Noise Control Eng. J. 48(3), 85–96 (2000)], which are used in our Standard, ANSI/ASA S12.2 Criteria for evaluating room noise. The fundamental issue is: Can we hear slowly surging or pulsating sounds for which the LEQ spectrum is below the threshold of hearing, where “slowly” means that the pulses come at a rate that is no faster than about 4 pulses per second? The short answer is yes, and the longer answer is that this effect is a function of the spectral content and becomes more-and-more prominent as the spectral content goes lower-and-lower in the audible frequency range. So surging or pulsing sound that is primarily in the 16 or 31 Hz octave bands will show the greatest effect. This paper shows the applicability of these results to wind-turbine sound. Variation in the threshold of hearing at low frequencies is an additional factor that also is discussed in this paper.
Autocorrelation analysis of military jet aircraft noise
Authors: Harker Blaine M. , Gee Kent L. , Neilsen Tracianne B. , McInerny Sally A. , James Michael M.
Published: May 2013

Jet noise research has seen increased use of autocorrelation analyses to glean physical insight about the source and its radiation properties. Length scales and other features have been identified in support of models incorporating large-scale (LSS) and fine-scale (FSS) turbulent structures. In this paper, the meaningful use of autocorrelation in jet noise analysis is further examined. A key finding is that the effect of the peak frequency on autocorrelation width needs to be removed prior to making conclusions about the relative LSS and FSS contributions. In addition, the Hilbert transform is applied to create an envelope of the autocorrelation function to more consistently define a characteristic time scale. These methods are first applied to the analytical LSS and FSS similarity spectra, previously developed by Tam et al. [AIAA 96-1716, 1996]. It is found that the envelope of the FSS similarity autocorrelation function is more similar to that of a delta function than the LSS envelope. These curves are used to more effectively quantify FSS and LSS features in noise spectra from the F-22A Raptor. [Work supported by ONR.]
Evaluating the role of efferent inhibition on cochlear responses: Simultaneous psychophysical and otoacoustic emission measurements
Authors: Henin Simon , Long Glenis
Published: May 2013

The auditory system continuously adapts to changes in the acoustic environment. Behavioral experiments in humans have demonstrated that changes in the acoustic environment produce dynamic changes in perception, for example, increases in thresholds in the presence of background noise. This dynamic change in the auditory system is hypothesized to be mediated by efferent feedback from the olivocochlear system. The effect of efferent inhibition on cochlear mechanics was investigated using a simultaneous psychoacoustics and otoacoustic emissions (OAEs) task using identical stimulus conditions. Cochlear responses to short tone-burst stimuli were analyzed under various masking conditions. Robust modification of cochlear responses to the short tone-burst stimuli was observed during contralateral acoustic stimulation and during long-duration ipsilateral masking. Concomitant changes in perceptual thresholds and OAEs are consistent with the hypothesis that both stem from efferent activation. This novel paradigm provides simultaneous perceptual and physiological estimates of cochlear-based efferent activation in the same human subjects.
Total and component distortion product otoacoustic emission analysis in persons with induced negative middle ear pressure
Authors: Thompson Suzanne , Long Glenis , Henin Simon
Published: May 2013

Distortion product otoacoustic emissions (DPOAEs) are generated when two primary tones (f1, f2 with f2 > f1) are presented simultaneously to the ear. Inter-modulation between primary tones produces distortion products at predictable frequencies not present in the original signal (e.g., 2f1 − f2). In persons with negative middle ear pressure (NMEP), the tympanic membrane is retracted and pulled inward, compressing structures in the middle ear. NMEP is expected to modify DPOAE level and phase. Performing the Toynbee maneuver can artificially induce NMEP. Changes in DPOAE primary f1 and f2 level and phase and energy reflectance measures were used to confirm that 8 subjects had artificially induced NMEP using the Toynbee maneuver. DPOAEs were obtained using 1 s/octave duration logarithmic frequency sweeping primaries, f2/f1 = 1.22 producing 2f1 − f2 from 320–2560 Hz, L1 = L2 = 65, 70, 75 dB SPL. There was a significant effect of condition, primary level, and frequency on total DPOAE and component amplitude. Separation of DPOAE components allowed improved detection of NMEP effects on DPOAE amplitude.
Correlations between noninvasive and direct physiological metrics of auditory function in chinchillas with noise-induced hearing loss
Authors: Henry Kenneth S. , Snyder Sandra F. , Heinz Michael G.
Published: May 2013

Noninvasive physiological tools for assessing auditory function in humans can provide valuable information when behavioral tests are not possible. Furthermore, these tools hold promise to provide greater insight into underlying cochlear pathologies. In this study, we used noninvasive distortion product otoacoustic emissions (DPOAEs) and auditory brainstem responses (ABRs) to estimate changes in auditory function in chinchillas with noise-induced hearing loss. Two aspects of cochlear function, sensitivity (threshold) and frequency selectivity (tuning), were measured directly using neurophysiological recordings from auditory-nerve (AN) fibers to assess the predictive value of DPOAEs and ABRs. Both DPOAE amplitude and ABR threshold were well correlated (R-square ~0.5) with AN fiber threshold near stimulus frequency. For DPOAEs, the correlation was strongest for cochlear function near F2. Correlations of both noninvasive metrics with AN tuning were weaker but statistically significant. The relatively weak correlation between DPOAE amplitude and AN tuning was unexpected because both measures are tied to the integrity of outer hair cells. Alternatively, previous results suggest ABR latency may be more predictive of AN tuning. Ultimately, DPOAEs may be a more practical clinical and research screening tool for hearing loss than ABRs due to shorter recording time. [Research supported by NIH (NIDCD) F32-DC012236 and R01-DC009838.]
How broadband speech may avoid neural firing rate saturation at high intensities and maintain intelligibility
Authors: Warren Richard , Bashford James , Lenz Peter
Published: May 2013

While broadband speech may remain perfectly intelligible at levels exceeding 90 dB, narrowband speech intelligibility (e.g., 2/3-octave passband centered at 1.5 kHz) may decline by 25% or more at moderate intensities (e.g., 75 dB). This “rollover” effect is substantially reduced, however, when a speech band is accompanied by flanking bands of white noise [Bashford et al., J. Acoust. Soc. Am. 117, 365–369 (2005)], suggesting that lateral suppression helps preserve broadband speech intelligibility at high levels. The present study found that when noise flankers were presented individually at a low spectrum level (−30 dB relative to the speech) only the higher-frequency flanker produced a significant intelligibility increase. However, the lower-frequency flanking noise did produce an equivalent increase when its spectrum level was raised 10 dB. This asymmetrical intensity requirement for noise flankers links the effective dynamic range of speech intelligibility to reported characteristics of both lateral (two-tone) suppression of auditory nerve (AN) fiber activity and lateral inhibition of secondary cells of the cochlear nucleus. These and other observations will be discussed in the broader context of how various auditory mechanisms help preserve speech intelligibility at high intensities by reducing firing rate saturation. [Work supported by NIH.]
A simulation of neural coding and auditory frequency analysis
Authors: Houde Robert A. , Hillenbrand James M. , Gayvert Robert T. , Houde John F.
Published: May 2013

Our understanding of the neural mechanisms underlying the very fine auditory frequency discrimination exhibited by listeners remains far from complete. To investigate this question we developed a functional model of the cochlear process in sufficient detail to allow the simulation of the principal characteristics of the cochlea's response to multi-tone and noise stimuli over a wide range of input levels. The model simulates level-dependent changes in frequency selectivity, combination-tone distortion, tone-on-tone suppression and masking, adaptation, and critical-band masking. The model is structured as 3000 channels, each consisting of a basilar membrane bandpass filter and inner-hair cell assembly. Input to each channel is the stapes displacement signal, and the output consists of ten independent stochastic point processes that are transmitted to the CNS on auditory-nerve fibers (ANFs). Our main purpose is to address these questions: (1) What narrowband spectrum information is available in the cochlea output? (2) How is this information encoded on the ANFs? (3) How might it be decoded in the CNS? An analysis of ensemble coding of the cochlear output showed that the precision (signal-to-noise ratio) of the information decoded by the CNS frequency analysis is directly related to the bandwidth of the basilar membrane filters.
A physiology-based auditory model elucidating the function of the cochlear amplifier and related phenomena. Part I: Model structure and computational method
Authors: Hudde Herbert , Becker Sebastian
Published: May 2013

An auditory model “PhyBAM” is presented, which in the long term aims at reproducing the human auditory perception. In recent years, the awareness has grown that many perceptive features have their origin in the peripheral ear, above all in the cochlea. In the present stage, PhyBAM is actually just a model of the peripheral ear. To simulate the perception of arbitrary sound signals, the signal processing occurring in the cochlea has to be formulated close to the physiological basis. Even so the model must be kept as simple as possible for the given aim. As a compromise, PhyBAM is set up as an ordinary circuit model. In this first of two associated papers, the model structure and the computational methods are presented. The model covers ear canal, middle ear, and cochlea. The cochlea model is by far the most sophisticated part. To include the unsymmetrical conditions at both cochlear windows and the resulting common and differential modes, a two-canal circuit is used. The main challenge is the implementation of the cochlear amplifier on the basis of measured tuning curves and otoacoustic emissions. Finding an appropriate model structure and proper parameters turns PhyBAM into an instrument of cochlear research.
Application of the spectral method for computation of the spectrum of anisotropic waveguides
Authors: Zharnikov Timur , Syresin Denis , Hsu Chaur-Jian
Published: May 2013

Spectral method is formulated in cylindrical coordinates for the general case of waveguide with arbitrary anisotropy with the spatial dependence. According to the idea of this approach, matrix representation of operator in the right-hand side of governing equations is considered. As a result, the latter are cast into exact infinite set of integro-differential equations. Explicit expressions for their kernels expose coupling between axial and azimuthal harmonics. Coupling of axial harmonics vanishes in important case of waveguide with translational invariance in axial direction. It results in the set of differential equations, which is used to introduce practical approximation procedure. The latter yields generalized eigenvalue problem, which can be solved numerically for the spectrum of the operator. The spectrum is sorted according to eigenmodes’ properties. Thus dispersion curves of eigenmodes are constructed. Presented consideration can be adapted for waveguides of different physical nature (elastic, electromagnetic, etc.) and different geometry (rectangular, elliptical, etc.). Developed technique is verified by comparison with results of controlled laboratory measurements on anisotropic sample. Monopole, dipole and quadrupole normal modes for scaled borehole in anisotropic rock sample with TTI symmetry are considered. The comparison of spectral method results with the dispersion analysis of synthetic data is provided as well.
Active detection of a moving target in a waveguide with strong masking echoes
Authors: Benoit Yoann , Prada Claire
Published: May 2013

In shallow water, active detection of a small moving target can be difficult because of strong echoes from large fixed obstacles. To cancel strong unwanted echoes, differences between successive acquisitions can be achieved, however they are very sensitive to fluctuations. A projection method combined with a fast acquisition technique is proposed as a robust alternative. An ultrasonic experiment is presented: a 64 transducers linear vertical array is used to detect a small target moving above a large obstacle in a waveguide. To reduce acquisition time, eight groups of adjacent elements transmit linear frequency modulations with increasing delays in a single emission. The 8x64 array response matrix is then obtained by correlations and time windowing. The projection is achieved between two acquisitions obtained while the target is moving, in order to remove the obstacle’s contribution. Namely, the second acquired matrix is projected on the space orthogonal to the eight singular vectors of the first acquired matrix. Then, it is shown that the first singular vector of the projected matrix focuses on the second target’s position. Comparisons are made with the decomposition of the time reversal operator in differential mode and conventional beamforming.
Experimental verification of enhanced sound transmission from water to air at low frequencies
Authors: Calvo David C. , Nicholas Michael , Orris Gregory J.
Published: May 2013

Laboratory-tank measurement of enhanced sound transmission from water to air at low frequencies is presented. Findings are consistent with the theory of anomalous transparency of the water-air interface in which almost all of the acoustic energy emitted by a shallow submerged source is emitted into the air [Godin, Phys. Rev. Lett. 97 (2006)]. The classical picture in the water remains very much the same: the monopole source suffers a radiation efficiency decrease due to interference with the strong surface reflection [McDonald and Calvo, J. Acoust. Soc. Am. 122 (2007)]. For source depths progressively less than a fraction of an acoustic wavelength in water, the measured radiation pattern in the air becomes progressively omnidirectional. The wider radiation pattern owes itself to the conversion of inhomogeneous (evanescent waves) into propagating waves that fill the angular space outside the usual 13-degree cone. On-axis point measurements using a microphone in air and a hydrophone in water, along with the measured directivities, are consistent with previously published power transmission ratios from water to air. Asymptotic expressions for the radiated field in the air are also presented. [Work sponsored by the Office of Naval Research.]
The rate of convergence and error distribution of Galerkin approximations to eigenvalues in underwater acoustics
Authors: Evans Richard B.
Published: May 2013

The eigenfunctions of the depth separated wave equation can be expanded in terms of a known finite basis set. The expansion coefficients are found by requiring that the error is orthogonal to each of the M members of the basis. This is the Galerkin approximation, wherein one obtains an MxM matrix eigenvalue problem that can be solved by existing software packages. The convergence of the matrix eigenvalues depends on the suitability of the chosen basis set. Typically, the errors in the matrix eigenvalues are bounded by 1/(Mr), for large M, where the exponent r > 0 is the rate of convergence: Consider basis sets consisting of trigonometric (Fourier) or orthogonal (Legrendre) polynomials. The density discontinuity at the bottom of the ocean creates a corner in the eigenfunctions that should be built into the basis sets. A corner in the sound speed profile (e.g., at the bottom of the mixed-layer) yields r = 3/2, which assures convergence, but is still a practical consideration. The distribution of errors is a determining factor in the choice between Fourier-Galerkin and Legendre-Galerkin. The errors in the first (2/pi)M of the eigenvalues are orders of magnitude smaller with Legendre-Galerkin, in the problem presented.
A hybrid acoustic model for room impulse response synthesis
Authors: Southern Alexander , Siltanen Samuel
Published: May 2013

The prediction and synthesis of room impulse responses (RIR) has wide application from computer gaming to architectural acoustics. When a level of physical accuracy is important, a single acoustic modeling technique is usually limited by its computational load. Hybrid acoustic models target different time/frequency regions of the RIR with different modeling techniques. This paper introduces a hybrid acoustic model consisting of a physical FDTD model for low-mid frequencies, beam-tracing, and the acoustic radiance transfer method in the early part and late parts at high frequencies respectively. In this work, attention is given to establishing the equivalence of the boundary characteristics in each modeling domain. Good agreement is demonstrated indicating that mixing the separate model responses leads to an energetically consistent RIR.
Global understanding of important parameters for improvement of impact insulation
Authors: Migneron Jean-Philippe , Migneron Jean-Gabriel
Published: May 2013

Floors impact insulation performances can be very different from one assembly to another. Many years of research and development have been done in this area. Now, it seems that a new solution or another product is commercialized every week. From buyers’ point of view, there is a need to decide which topping and underlay will suit some noise requirement to the lowest cost. However, acousticians and specialists in noise control might consider a more complex problem, especially in multi-family dwellings. In lightweight construction, the relation between the floor and the ceiling underneath also affect the overall performance in terms of IIC, or even in risk of complaints. Knowing that it is often difficult to compare a real situation to a datasheet from a manufacturer or to building codes, few key ideas should be remembered. This paper aims to briefly review some conclusions of previous works done in impact sound insulation and to analyze how fundamental parameters can be applied to real installations. An example of variable modification on a topping sample also tries to demonstrate the influence of basic aspects without according most attention to single number ratings.