bluuebunny/crossref_metadata_2025 huggingface dataset:
Ray-based acoustic localization of cavitation in a highly reverberant environment
Authors: Chang Natasha A. , Dowling David R.
Published: May 2009

Acoustic detection and localization of cavitation have inherent advantages over optical techniques because cavitation bubbles are natural sound sources, and acoustic transduction of cavitation sounds does not require optical access to the region of cavitating flow. In particular, near cavitation inception, cavitation bubbles may be visually small and occur infrequently, but may still emit audible sound pulses. In this investigation, direct-path acoustic recordings of cavitation events are made with 16 hydrophones mounted on the periphery of a water tunnel test section containing a low-cavitation-event-rate vortical flow. These recordings are used to localize the events in three dimensions via cross correlations to obtain arrival time differences. Here, bubble localization is hindered by reverberation, background noise, and the fact that both the pulse emission time and waveform are unknown. These hindrances are partially mitigated by a signal-processing scheme that incorporates straight-ray acoustic propagation and Monte-Carlo techniques for compensating ray-path, sound-speed, and hydrophone-location uncertainties. The acoustic localization results are compared to simultaneous optical localization results from dual-camera high-speed digital-video recordings. For 53 bubbles and a peak-signal to noise ratio frequency of 6.7 kHz, the root-mean-square spatial difference between optical and acoustic bubble location results was 1.94 cm. Parametric dependences in acoustic localization performance are also presented.
Enhanced channel estimation and symbol detection for high speed multi-input multi-output underwater acoustic communications
Authors: Ling Jun , Yardibi Tarik , Su Xiang , He Hao , Li Jian
Published: May 2009

The need for achieving higher data rates in underwater acoustic communications leverages the use of multi-input multi-output (MIMO) schemes. In this paper two key issues regarding the design of a MIMO communications system, namely, channel estimation and symbol detection, are addressed. To enhance channel estimation performance, a cyclic approach for designing training sequences and a channel estimation algorithm called the iterative adaptive approach (IAA) are presented. Sparse channel estimates can be obtained by combining IAA with the Bayesian information criterion (BIC). Moreover, the RELAX algorithm can be used to improve the IAA with BIC estimates further. Regarding symbol detection, a minimum mean-squared error based detection scheme, called RELAX-BLAST, which is a combination of vertical Bell Labs layered space-time (V-BLAST) algorithm and the cyclic principle of the RELAX algorithm, is presented and it is shown that RELAX-BLAST outperforms V-BLAST. Both simulated and experimental results are provided to validate the proposed MIMO scheme. RACE’08 experimental results employing a 4×24 MIMO system show that the proposed scheme enjoys an average uncoded bit error rate of 0.38% at a payload data rate of 31.25kbps and an average coded bit error rate of 0% at a payload data rate of 15.63kbps.
Relations between frequency selectivity, temporal fine-structure processing, and speech reception in impaired hearing
Authors: Strelcyk Olaf , Dau Torsten
Published: May 2009

Frequency selectivity, temporal fine-structure (TFS) processing, and speech reception were assessed for six normal-hearing (NH) listeners, ten sensorineurally hearing-impaired (HI) listeners with similar high-frequency losses, and two listeners with an obscure dysfunction (OD). TFS processing was investigated at low frequencies in regions of normal hearing, through measurements of binaural masked detection, tone lateralization, and monaural frequency modulation (FM) detection. Lateralization and FM detection thresholds were measured in quiet and in background noise. Speech reception thresholds were obtained for full-spectrum and lowpass-filtered sentences with different interferers. Both the HI listeners and the OD listeners showed poorer performance than the NH listeners in terms of frequency selectivity, TFS processing, and speech reception. While a correlation was observed between the monaural and binaural TFS-processing deficits in the HI listeners, no relation was found between TFS processing and frequency selectivity. The effect of noise on TFS processing was not larger for the HI listeners than for the NH listeners. Finally, TFS-processing performance was correlated with speech reception in a two-talker background and lateralized noise, but not in amplitude-modulated noise. The results provide constraints for future models of impaired auditory signal processing.
Release and re-buildup of listeners’ models of auditory space
Authors: Keen Rachel , Freyman Richard L.
Published: May 2009

When listeners hear sound presented repeatedly in a room with reflections, echo threshold rises. The current experiments tested how long this buildup in echo threshold would last when exposure to a different simulated space (designated as room B) intervened before returning to the original space (designated room A). Stimuli were trains of lead–lag click pairs (room A) and trains of clicks with no reflections (room B) in an ABA sequence. After buildup in room A, echo threshold for click pairs in room A decreased in direct relation to amount of intervening exposure to room B. After 11 click pairs of room B, the effect of exposure to room A was gone. A second buildup in echo threshold in room A was not differentially affected by prior exposure to room A or a different simulated room, room C. Listeners appear to form a model when exposed to sound in a particular space, which is lost quickly upon hearing sound in a different space. Storing previous models is inefficient because the processes of buildup and breakdown occur quickly to sound in a new space.
Passive acoustic detection of schools of herring
Authors: Hahn Thomas R. , Thomas Gary
Published: May 2009

Herring (Clupea pallasii and C. harengus) have been observed to release gas from their bladders during vertical migration likely to adjust buoyancy and also when under strong predation pressure. Based on recently measured and modeled sound for individual fish, spectral levels are estimated for entire herring schools in the ocean for both scenarios, and the feasibility of passive detection is explored. For a typical school of migrating herring near-surface spectral levels of about 50dB rel., 1μPa∕√Hz at 3–7kHz are predicted. If wind conditions are calm where migrating herring are found, such as for Pacific herring in Prince William Sound, Alaska, passive detection is very likely. For an exemplary 10 metric ton compact school, peak spectral source levels of about 80–90dB rel. 1μPa∕√Hz ref. 1m are predicted, yielding a range of detection against calm wind background of about 1000m. Field measurements of potential gas-release events agree with the predictions for the compact school scenario with regard to levels and spectral shape and indicate that passive acoustic monitoring is feasible and could be a prime tool to study predator-prey interactions.
Application of Renyi entropy for ultrasonic molecular imaging
Authors: Hughes M. S. , Marsh J. N. , Arbeit J. M. , Neumann R. G. , Fuhrhop R. W. , Wallace K. D. , Thomas L. , Smith J. , Agyem K. , Lanza G. M. , Wickline S. A. , McCarthy J. E.
Published: May 2009

Previous work has demonstrated that a signal receiver based on a limiting form of the Shannon entropy is, in certain settings, more sensitive to subtle changes in scattering architecture than conventional energy-based signal receivers [M. S. Hughes et al., J. Acoust. Soc. Am. 121, 3542–3557 (2007)]. In this paper new results are presented demonstrating further improvements in sensitivity using a signal receiver based on the Renyi entropy.
Evaluating the role of spectral and envelope characteristics in the intelligibility advantage of clear speech
Authors: Krause Jean C. , Braida Louis D.
Published: May 2009

In adverse listening conditions, talkers can increase their intelligibility by speaking clearly [Picheny, M.A., et al. (1985). J. Speech Hear. Res. 28, 96–103; Payton, K. L., et al. (1994). J. Acoust. Soc. Am. 95, 1581–1592]. This modified speaking style, known as clear speech, is typically spoken more slowly than conversational speech [Picheny, M. A., et al. (1986). J. Speech Hear. Res. 29, 434–446; Uchanski, R. M., et al. (1996). J. Speech Hear. Res. 39, 494–509]. However, talkers can produce clear speech at normal rates (clear/normal speech) with training [Krause, J. C., and Braida, L. D. (2002). J. Acoust. Soc. Am. 112, 2165–2172] suggesting that clear speech has some inherent acoustic properties, independent of rate, that contribute to its improved intelligibility. Identifying these acoustic properties could lead to improved signal processing schemes for hearing aids. Two global-level properties of clear/normal speech that appear likely to be associated with improved intelligibility are increased energy in the 1000–3000-Hz range of long-term spectra and increased modulation depth of low-frequency modulations of the intensity envelope [Krause, J. C., and Braida, L. D. (2004). J. Acoust. Soc. Am. 115, 362–378]. In an attempt to isolate the contributions of these two properties to intelligibility, signal processing transformations were developed to manipulate each of these aspects of conversational speech independently. Results of intelligibility testing with hearing-impaired listeners and normal-hearing listeners in noise suggest that (1) increasing energy between 1000 and 3000Hz does not fully account for the intelligibility benefit of clear/normal speech, and (2) simple filtering of the intensity envelope is generally detrimental to intelligibility. While other manipulations of the intensity envelope are required to determine conclusively the role of this factor in intelligibility, it is also likely that additional properties important for highly intelligible speech at normal speech at normal rates remain to be identified.
On the level-dependent attenuation of a perforated device
Authors: Chen Lan , Sang Jinqiu , Li Xiaodong
Published: May 2009

To investigate the physical principle governing the level-dependent attenuation of a perforated earplug, a mathematical model is first established with the transfer-matrix method to calculate the noise reduction through a simplified device, one perforated panel with back cavity, mounted in an impedance tube. The model prediction is compared with the measured noise reduction through two series of large-scale devices and one device with the dimensions of the ear canal under continuous noise and sinusoidal excitations. The model helps to improve significantly the level-dependent attenuation of the large-scale device. It also illustrates that the attenuation is not solely determined by the resistance of the orifice, which has been a well accepted design concept, but resulted from an incorporated effect of the acoustic filter comprised of the acoustic impedance of the orifice and other elements in the earplug-ear-canal system. This mechanism can interpret a resonance at low incident levels on improper design and reveal approaches to eliminate it. Finally, the model’s potential contributions to the design of a perforated earplug are discussed, along with the threshold of level-dependent attenuation supported with experimental evidence.
Scattering calculation and image reconstruction using elevation-focused beams
Authors: Duncan David P. , Astheimer Jeffrey P. , Waag Robert C.
Published: May 2009

Pressure scattered by cylindrical and spherical objects with elevation-focused illumination and reception has been analytically calculated, and corresponding cross sections have been reconstructed with a two-dimensional algorithm. Elevation focusing was used to elucidate constraints on quantitative imaging of three-dimensional objects with two-dimensional algorithms. Focused illumination and reception are represented by angular spectra of plane waves that were efficiently computed using a Fourier interpolation method to maintain the same angles for all temporal frequencies. Reconstructions were formed using an eigenfunction method with multiple frequencies, phase compensation, and iteration. The results show that the scattered pressure reduces to a two-dimensional expression, and two-dimensional algorithms are applicable when the region of a three-dimensional object within an elevation-focused beam is approximately constant in elevation. The results also show that energy scattered out of the reception aperture by objects contained within the focused beam can result in the reconstructed values of attenuation slope being greater than true values at the boundary of the object. Reconstructed sound speed images, however, appear to be relatively unaffected by the loss in scattered energy. The broad conclusion that can be drawn from these results is that two-dimensional reconstructions require compensation to account for uncaptured three-dimensional scattering.
Optimal simulations of ultrasonic fields produced by large thermal therapy arrays using the angular spectrum approach
Authors: Zeng Xiaozheng , McGough Robert J.
Published: May 2009

The angular spectrum approach is evaluated for the simulation of focused ultrasound fields produced by large thermal therapy arrays. For an input pressure or normal particle velocity distribution in a plane, the angular spectrum approach rapidly computes the output pressure field in a three dimensional volume. To determine the optimal combination of simulation parameters for angular spectrum calculations, the effect of the size, location, and the numerical accuracy of the input plane on the computed output pressure is evaluated. Simulation results demonstrate that angular spectrum calculations performed with an input pressure plane are more accurate than calculations with an input velocity plane. Results also indicate that when the input pressure plane is slightly larger than the array aperture and is located approximately one wavelength from the array, angular spectrum simulations have very small numerical errors for two dimensional planar arrays. Furthermore, the root mean squared error from angular spectrum simulations asymptotically approaches a nonzero lower limit as the error in the input plane decreases. Overall, the angular spectrum approach is an accurate and robust method for thermal therapy simulations of large ultrasound phased arrays when the input pressure plane is computed with the fast nearfield method and an optimal combination of input parameters.
Model predicts bat pinna ridges focus high frequencies to form narrow sensitivity beams
Authors: Kuc Roman
Published: May 2009

The pinnae of bats contain ridges whose function was previously thought to be structural. This paper suggests that ridges form a reflecting Fresnel lens that focuses high-frequency acoustic signals into the ear canal to form a narrow elevation sensitivity beam. E. fuscus ridges are modeled as a series of four paraboloidal strips and the tragus is considered to act as a secondary reflecting element analogous to a Cassegrain system. A diffraction grading having the equivalent spacing suggests frequencies above 150kHz. Using an example 170kHz, a random search for ridge dimensions that minimize side-lobes in the frequency magnitude response yields the tapered ridge structure observed in E. fuscus and produces an 18° (full width half energy) beam width. We speculate that the possible high-frequency sources are ecologically (prey) generated and/or the third harmonic of the call. The attenuation at such high frequencies requires that the source be close by. Passive prey localization in the postbuzz stage, when echoes overlap call transmissions and the prey is within 8cm, could improve prey capture efficiency. An experiment using 40kHz ultrasound with human observers verifies that frequencies beyond the audiometric range, when sufficiently intense, can still be perceived.
Long-term stability of spontaneous otoacoustic emissions
Authors: Burns Edward M.
Published: May 2009

Spontaneous otoacoustic emissions (SOAEs) were measured longitudinally for durations up to 19.5 years. Initial ages of the subjects ranged from 6 to 41 years. The most compelling finding was a decrease in frequency of all emissions in all subjects, which was approximately linear in %/year and averaged 0.25%/year. SOAE levels also tended to decrease with age, a trend that was significant, but not consistent across emissions, either within or across subjects. Levels of individual SOAEs might decrease, increase, or remain relatively constant with age. Several types of frequency/level instabilities were noted in which some SOAEs within an ear interacted such that their levels were negatively correlated. These instabilities often persisted for many years. SOAEs were also measured in two females over the course of their pregnancies. No changes in SOAE levels or frequencies were seen, that were larger than have been reported in females over a menstrual cycle, suggesting that levels of female gonadal hormones do not have a significant direct effect on SOAE frequencies or levels.
The impact of ocean sound speed variability on the uncertainty of geoacoustic parameter estimates
Authors: Jiang Yong-Min , Chapman N. Ross
Published: May 2009

This paper investigates the influence of water column variability on the estimates of geoacoustic model parameters obtained from matched field inversions. The acoustic data were collected on the New Jersey continental shelf during shallow water experiments in August 2006. The oceanographic variability was evident when the data were recorded. To quantify the uncertainties of the geoacoustic parameter estimates in this environment, Bayesian matched field geoacoustic inversion was applied to multi-tonal continuous wave data. The spatially and temporally varying water column sound speed is parametrized in terms of empirical orthogonal functions and included in the inversion. Its impact on the geometric and geoacoustic parameter estimates is then analyzed by the inter-parameter correlations. Two different approaches were used to obtain information about the variation of the water sound speed. One used only the profiles collected along the experimental track during the experiment, and the other also included observations collected over a larger area. The geoacoustic estimates from both the large and small sample sets are consistent. However, due to the diversity of the oceanic sound speed, more empirical orthogonal functions are needed in the inversion when more sound speed profile samples are used.
Some spatial and temporal effects on the speech privacy of meeting rooms
Authors: Bradley John S. , Apfel Marina , Gover Bradford N.
Published: May 2009

This paper reports on initial experiments concerning how key spatial and temporal effects in rooms influence the speech privacy provided by enclosed rooms. The first part of the work demonstrates that for the same signal-to-noise ratio, the intelligibility of speech and the threshold of intelligibility are significantly different for transmission between real rooms than in the previous results in approximately free-field conditions [B. N. Gover and J. S. Bradley, J. Acoust. Soc. Am. 116, 3480–3490 (2004)]. The second part investigates the influence of aspects of the spatial and temporal components of sound fields in typical rooms, to explain these differences for transmission between real rooms. These components included the separate effects of early-arriving and later-arriving reflected speech sounds. They also included the effects of spatially separated speech and noise sources as well as more diffuse noise representative of typical meeting rooms. In realistic combinations these effects are of practical importance and can change privacy criteria by 5dB or more. Ignoring them could lead to costly over-design of the sound insulation required to achieve adequate speech privacy.
Cues to perception of reduced flaps
Authors: Warner Natasha , Fountain Amy , Tucker Benjamin V.
Published: May 2009

Natural, spontaneous speech (and even quite careful speech) often shows extreme reduction in many speech segments, even resulting in apparent deletion of consonants. Where the flap ([ɾ]) allophone of /t/ and /d/ is expected in American English, one frequently sees an approximant-like or even vocalic pattern, rather than a clear flap. Still, the /t/ or /d/ is usually perceived, suggesting the acoustic characteristics of a reduced flap are sufficient for perception of a consonant. This paper identifies several acoustic characteristics of reduced flaps based on previous acoustic research (size of intensity dip, consonant duration, and F4 valley) and presents phonetic identification data for continua that manipulate these acoustic characteristics of reduction. The results indicate that the most obvious types of acoustic variability seen in natural flaps do affect listeners’ percept of a consonant, but not sufficiently to completely account for the percept. Listeners are affected by the acoustic characteristics of consonant reduction, but they are also very skilled at evaluating variability along the acoustic dimensions that realize reduction.
The use of phrase-level prosodic information in lexical segmentation: Evidence from word-spotting experiments in Korean
Authors: Kim Sahyang , Cho Taehong
Published: May 2009

This study investigated the role of phrase-level prosodic boundary information in word segmentation in Korean with two word-spotting experiments. In experiment 1, it was found that intonational cues alone helped listeners with lexical segmentation. Listeners paid more attention to local intonational cues (…H#L…) across the prosodic boundary than the intonational information within a prosodic phrase. The results imply that intonation patterns with high frequency are used, though not exclusively, in lexical segmentation. In experiment 2, final lengthening was added to see how multiple prosodic cues influence lexical segmentation. The results showed that listeners did not necessarily benefit from the presence of both intonational and final lengthening cues: Their performance was improved only when intonational information contained infrequent tonal patterns for boundary marking, showing only partially cumulative effects of prosodic cues. When the intonational information was optimal (frequent) for boundary marking, however, poorer performance was observed with final lengthening. This is arguably because the phrase-initial segmental allophonic cues for the accentual phrase were not matched with the prosodic cues for the intonational phrase. It is proposed that the asymmetrical use of multiple cues was due to interaction between prosodic and segmental information that are computed in parallel in lexical segmentation.
Real-time algorithm for acoustic imaging with a microphone array
Authors: Huang Xun
Published: May 2009

Acoustic phased array has become an important testing tool in aeroacoustic research, where the conventional beamforming algorithm has been adopted as a classical processing technique. The computation however has to be performed off-line due to the expensive cost. An innovative algorithm with real-time capability is proposed in this work. The algorithm is similar to a classical observer in the time domain while extended for the array processing to the frequency domain. The observer-based algorithm is beneficial mainly for its capability of operating over sampling blocks recursively. The expensive experimental time can therefore be reduced extensively since any defect in a testing can be corrected instantaneously.
Perceptual learning of systematic variation in Spanish-accented speech
Authors: Sidaras Sabrina K. , Alexander Jessica E. D. , Nygaard Lynne C.
Published: May 2009

Spoken language is characterized by an enormous amount of variability in how linguistic segments are realized. In order to investigate how speech perceptual processes accommodate to multiple sources of variation, adult native speakers of American English were trained with English words or sentences produced by six Spanish-accented talkers. At test, listeners transcribed utterances produced by six familiar or unfamiliar Spanish-accented talkers. With only brief exposure, listeners perceptually adapted to accent-general regularities in spoken language, generalizing to novel accented words and sentences produced by unfamiliar accented speakers. Acoustic properties of vowel production and their relation to identification performance were assessed to determine if the English listeners were sensitive to systematic variation in the realization of accented vowels. Vowels that showed the most improvement after Spanish-accented training were distinct from nearby vowels in terms of their acoustic characteristics. These findings suggest that the speech perceptual system dynamically adjusts to the acoustic consequences of changes in talker’s voice and accent.
Scattering effect on the sound focused personal audio system
Authors: Chang Ji-Ho , Park Jin-Young , Kim Yang-Hann
Published: May 2009

Recently, a personal audio system has been studied that uses an array of loudspeakers to localize sound to only the area around a user. To realize this system, beamforming or acoustic contrast control has been applied on the assumption that sources radiate sound in a free-field. This means that not only reflection by walls, but also the scattering effect by the user’s head is neglected. Reflection by walls is negligible because personal devices are usually used in a short distance so that direct sound is dominant over reverberant sound. However, the scattering effect by the user’s head has a considerable effect on the focused sound field. For example, the region where sound energy is not focused becomes louder when a user is actually in the focused region due to the scattered sound by the user’s head in the focused region. In this paper, the scattering effect is shown computationally on the simple assumption that the user’s head is a rigid sphere. Then, an improving control method, which overcomes this effect, is proposed. The method is shown to outperform the previous method in terms of lowering the sound level in the side regions when a user is in the bright zone.
How sensitivity to ongoing interaural temporal disparities is affected by manipulations of temporal features of the envelopes of high-frequency stimuli
Authors: Bernstein Leslie R. , Trahiotis Constantine
Published: May 2009

This study addressed how manipulating certain aspects of the envelopes of high-frequency stimuli affects sensitivity to envelope-based interaural temporal disparities (ITDs). Listener’s threshold ITDs were measured using an adaptive two-alternative paradigm employing “raised-sine” stimuli [John, M. S., et al. (2002). Ear Hear. 23, 106–117] which permit independent variation in their modulation frequency, modulation depth, and modulation exponent. Threshold ITDs were measured while manipulating modulation exponent for stimuli having modulation frequencies between 32 and 256 Hz. The results indicated that graded increases in the exponent led to graded decreases in envelope-based threshold ITDs. Threshold ITDs were also measured while parametrically varying modulation exponent and modulation depth. Overall, threshold ITDs decreased with increases in the modulation depth. Unexpectedly, increases in the exponent of the raised-sine led to especially large decreases in threshold ITD when the modulation depth was low. An interaural correlation-based model was generally able to capture changes in threshold ITD stemming from changes in the exponent, depth of modulation, and frequency of modulation of the raised-sine stimuli. The model (and several variations of it), however, could not account for the unexpected interaction between the value of raised-sine exponent and its modulation depth.
Vowel-pitch matching in Wagner’s operas: Implications for intelligibility and ease of singing
Authors: Smith John , Wolfe Joe
Published: May 2009

European vowels are mainly distinguished by the two lowest resonance frequencies (R1 and R2) of the vocal tract. Once the pitch frequency f0 exceeds the value of R1 in normal speech, sopranos can deliberately “tune” R1 to match f0. This increases loudness, uniformity of tone, and ease of singing, at some cost to intelligibility. Resonance tuning would be assisted if the pitch of the note written for a vowel corresponded with its usual range of R1. Analysis of several soprano roles indicates that Wagner aided the acoustics of the soprano voice at high pitch when setting text to music.
Sound speed and density characterization of milk adulterated with melamine
Authors: Elvira Luis , Rodríguez Jaime , Lynnworth Lawrence C.
Published: May 2009

Milk contaminated with melamine resulted in an important health hazard that affected many babies in China recently. Ultrasonic characterization of adulterated milk may detect gross levels of melamine contamination. Sound speed and density measurements were made in skim milk as a function of melamine adulteration. An ultrasonic measurement technique to implement milk quality control is discussed.
Correction for partial reflection in ultrasonic attenuation measurements using contact transducers
Authors: Treiber Martin , Kim Jin-Yeon , Jacobs Laurence J. , Qu Jianmin
Published: May 2009

This research investigates the influence of partial reflection on the measurement of the absolute ultrasonic attenuation coefficient using contact transducers. The partial, frequency-dependent reflection arises from the thin fluid-layer interface formed between the transducer and specimen surface. It is experimentally shown that neglecting this reflection effect leads to a significant overestimation in the measured attenuation coefficient. A systematic measurement procedure is proposed that simultaneously obtains the ultrasonic signals needed to calculate both the reflection coefficient of the interface and the attenuation coefficient, without disturbing the existing coupling conditions. The true attenuation coefficient includes a correction based on the measured reflection coefficient—this is called the reflection correction. It is shown that including the reflection correction also reduces the variation (random error) in the measured attenuation coefficient. The accuracy of the proposed method is demonstrated for a material with a known attenuation coefficient. The proposed method is then used to measure the high attenuation coefficient of a cement-based material.
Noise reduction utilizing cross time-frequency ε-filter
Authors: Abe Tomomi , Matsumoto Mitsuharu , Hashimoto Shuji
Published: May 2009

A time-frequency ε-filter (TF ε-filter) is an advanced ε-filter applied to complex spectra along the time axis. It can reduce most kinds of noise while preserving a signal that varies frequently such as a speech signal. The filter design is simple and it can effectively reduce noise. It is applicable not only to small amplitude stationary noise but also to large amplitude nonstationary noise. However, when the filter is applied to the noise that varies much frequently along the time axis, TF ε-filter cannot reduce noise without the signal distortion. This paper introduces an advanced method for noise reduction that applies ε-filter to complex spectra not only along the time axis but also along the frequency axis labeled cross TF ε-filter. It can reduce the noise where the neighboring frequency bins have similar powers. To show the effectiveness of the proposed method, some comparative experiments are also given, such as the performance of noise reduction and the robustness concerning input signal-to-noise ratio and parameter changes.
Material properties from acoustic radiation force step response
Authors: Orescanin Marko , Toohey Kathleen S. , Insana Michael F.
Published: May 2009

An ultrasonic technique for estimating viscoelastic properties of hydrogels, including engineered biological tissues, is being developed. An acoustic radiation force is applied to deform the gel locally while Doppler pulses track the induced movement. The system efficiently couples radiation force to the medium through an embedded scattering sphere. A single-element, spherically-focused, circular piston element transmits a continuous-wave burst to suddenly apply and remove a radiation force to the sphere. Simultaneously, a linear array and spectral Doppler technique are applied to track the position of the sphere over time. The complex shear modulus of the gel was estimated by applying a harmonic oscillator model to measurements of time-varying sphere displacement. Assuming that the stress-strain response of the surrounding gel is linear, this model yields an impulse response function for the gel system that may be used to estimate material properties for other load functions. The method is designed to explore the force-frequency landscape of cell-matrix viscoelasticity. Reported measurements of the shear modulus of gelatin gels at two concentrations are in close agreement with independent rheometer measurements of the same gels. Accurate modulus measurements require that the rate of Doppler-pulse transmission be matched to a priori estimates of gel properties.
How to stretch and shrink vowel systems: Results from a vowel normalization procedure
Authors: Geng Christian , Mooshammer Christine
Published: May 2009

One of the goals of phonetic investigations is to find strategies for vowel production independent of speaker-specific vocal-tract anatomies and individual biomechanical properties. In this study techniques for speaker normalization that are derived from Procrustes methods were applied to acoustic and articulatory data. More precisely, data consist of the first two formants and EMMA fleshpoint markers of stressed and unstressed vowels of German from seven speakers in the consonantal context /t/. Main results indicate that (a) for the articulatory data, the normalization can be related to anatomical properties (palate shapes), (b) the recovery of phonemic identity is of comparable quality for acoustic and articulatory data, (c) the procedure outperforms the Lobanov transform in the acoustic domain in terms of phoneme recovery, and (d) this advantage comes at the cost of partly also changing ellipse orientations, which is in accordance with the formulation of the algorithms.
Generalized shot noise model for time-reversal in multiple-scattering media allowing for arbitrary inputs and windowing
Authors: Haworth Kevin J. , Fowlkes J. Brian , Carson Paul L. , Kripfgans Oliver D.
Published: May 2009

A theoretical shot noise model to describe the output of a time-reversal experiment in a multiple-scattering medium is developed. This (non-wave equation based) model describes the following process. An arbitrary waveform is transmitted through a high-order multiple-scattering environment and recorded. The recorded signal is arbitrarily windowed and then time-reversed. The processed signal is retransmitted into the environment and the resulting signal recorded. The temporal and spatial signal and noise of this process is predicted statistically. It is found that the time when the noise is largest depends on the arbitrary windowing and this noise peak can occur at times outside the main lobe. To determine further trends, a common set of parameters is applied to the general result. It is seen that as the duration of the input function increases, the signal-to-noise ratio (SNR) decreases (independent of signal bandwidth). It is also seen that longer persisting impulse responses result in increased main lobe amplitudes and SNR. Assumptions underpinning the generalized shot noise model are compared to an experimental realization of a multiple-scattering medium (a time-reversal chaotic cavity). Results from the model are compared to random number numerical simulation.
Discrimination of complex tones with unresolved components using temporal fine structure information
Authors: Moore Brian C. J. , Hopkins Kathryn , Cuthbertson Stuart
Published: May 2009

The information used to discriminate complex tones with (largely) unresolved components was assessed. In experiment 1, subjects discriminated a harmonic complex tone, H, with fundamental frequency F0, from an inharmonic tone, I, in which all components were shifted upwards by the same amount in hertz. Tones H and I had the same envelope repetition rate but different temporal fine structure (TFS). The tones were passed through a fixed bandpass filter centered on harmonic N, to reduce excitation pattern cues. For all F0s (35–400 Hz), performance decreased as N was increased from 11 to 15, but, except for F0=35 Hz, remained above chance for N=15, where all harmonics should be unresolved. This suggests that discrimination can be based on TFS rather than on partially resolved components. In experiment 2, subjects discriminated the F0 of complex tones filtered as in experiment 1. Here, both envelope rate and TFS cues were available. Except for F0=35 Hz, discrimination thresholds, expressed as the Weber fraction for a change in time interval, were similar to those measured in experiment 1, suggesting that performance in experiment 2 was dominated by the use of TFS rather than envelope cues.
Quantitative evaluation of fracture healing process of long bones using guided ultrasound waves: A computational feasibility study
Authors: Guo Xiasheng , Yang Di , Zhang Dong , Li Weiguo , Qiu Yong , Wu Junru
Published: May 2009

The feasibility of monitoring changes in guided waves’ characteristics in a fractured long bone as modeled by a hollow cylinder and a callus at different healing stages is studied. Various guided wave modes are detected and extracted from a broadband signal at several discrete locations. The energy-spectrum and vFEP (effective velocity of the first energy peak in callus region) of guided modes are found sensitive to the healing process in different aspects and stages. The healing process may be divided into several sub-courses, each of which can be evaluated by different combinations of guided wave modes. The energy-spectrum indicates that the longitudinal tube modes L(0,1) and L(0,2) are suitable for early healing; L(0,1), L(0,2), L(0,3), and L(0,5) for midway-course; and L(0,1) and L(0,3) for late consolidation, while L(0,2), L(0,5), and L(0,8) are suitable for detecting the change in callus geometrics. The vFEP results suggest that L(0,5) for monitoring early-course; L(0,3) and L(0,7) for midway process; L(0,2) for later consolidation, and L(0,7) for monitoring geometrical variation.
Relative-pitch tracking of multiple arbitrary sounds
Authors: Smaragdis Paris
Published: May 2009

Perceived-pitch tracking of potentially aperiodic sounds, as well as pitch tracking of multiple simultaneous sources, is shown to be feasible using a probabilistic methodology. The use of a shift-invariant representation in the constant-Q domain allows the modeling of perceived pitch changes as vertical shifts of spectra. This enables the tracking of these changes in sounds with an arbitrary spectral profile, even those where pitch would be an ill-defined quantity. It is further possible to extend this approach to a mixture model, which allows simultaneous tracking of varying mixed sounds. Demonstrations on real recordings highlight the robustness of such a model under various adverse conditions, and also show some of its unique conceptual differences when compared to traditional pitch tracking approaches.
Absolute calibration of hydrophones immersed in sandy sediment
Authors: Robb Gary B. N. , Robinson Stephen P. , Theobald Pete D. , Hayman Gary , Humphrey Victor F. , Leighton Timothy G. , Wang Lian Sheng , Dix Justin K. , Best Angus I.
Published: May 2009

An absolute calibration method has been developed based on the method of three-transducer spherical-wave reciprocity for the calibration of hydrophones when immersed in sandy sediment. The method enables the determination of the magnitude of the free-field voltage receive sensitivity of the hydrophone. Adoption of a co-linear configuration allows the acoustic attenuation within the sediment to be eliminated from the sensitivity calculation. Example calibrations have been performed on two hydrophones inserted into sandy sediment over the frequency range from 10to200kHz. In general, a reduction in sensitivity was observed, with average reductions over the frequency range tested of 3.2 and 3.6dB with respect to the equivalent water-based calibrations for the two hydrophones tested. Repeated measurements were undertaken to assess the robustness of the method to both the influence of the sediment disturbance associated with the hydrophone insertion and the presence of the central hydrophone. A simple finite element model, developed for one of the hydrophone designs, shows good qualitative agreement with the observed differences from water-based calibrations. The method described in this paper will be of interest to all those undertaking acoustic measurements with hydrophones immersed in sediment where the absolute sensitivity is important.
A stochastic model for the noise levels
Authors: Giménez A. , González M.
Published: May 2009

Accurate predictions of environmental noise levels are necessary to implement noise reduction strategies in urban areas. In this paper, a stochastic model is introduced to describe and predict the Lden, Lday, Levening, and Lnight levels. A Gaussian Ornstein–Uhlenbeck model is used to represent the dynamics of the noise levels, where the mean-reversion properties and seasonal volatility for each day of the week are studied separately.
Effects of pulsing of the target tone on the audibility of partials in inharmonic complex tones
Authors: Moore Brian C. J. , Glasberg Brian R. , Jepsen Morten L.
Published: May 2009

The audibility of partials was measured for complex tones with partials uniformly spaced on an ERBN-number scale. On each trial, subjects heard a sinusoidal “probe” followed by a complex tone. The probe was mistuned downwards or upwards (at random) by 3% or 4.5% from the frequency of one randomly selected partial in the complex (the “target”). The subject indicated whether the target was higher or lower in frequency than the probe. The probe and the target were pulsed on and off and the ramp times and inter-pulse intervals were systematically varied. Performance was better for longer ramp times and longer inter-pulse intervals. In a second experiment, the ability to detect which of two complex tones contained a pulsed partial was measured. The pattern of results was similar to that for experiment 1. A model of auditory processing including an adaptation stage was able to account for the general pattern of the results of experiment 2. The results suggest that the improvement in ability to hear out a partial in a complex tone produced by pulsing that partial is partly mediated by a release from adaptation produced by the pulsing, and does not result solely from reduction of perceptual confusion.
Auditory and auditory-visual intelligibility of speech in fluctuating maskers for normal-hearing and hearing-impaired listeners
Authors: Bernstein Joshua G. W. , Grant Ken W.
Published: May 2009

Speech intelligibility for audio-alone and audiovisual (AV) sentences was estimated as a function of signal-to-noise ratio (SNR) for a female target talker presented in a stationary noise, an interfering male talker, or a speech-modulated noise background, for eight hearing-impaired (HI) and five normal-hearing (NH) listeners. At the 50% keywords-correct performance level, HI listeners showed 7–12 dB less fluctuating-masker benefit (FMB) than NH listeners, consistent with previous results. Both groups showed significantly more FMB under AV than audio-alone conditions. When compared at the same stationary-noise SNR, FMB differences between listener groups and modalities were substantially smaller, suggesting that most of the FMB differences at the 50% performance level may reflect a SNR dependence of the FMB. Still, 1–5dB of the FMB difference between listener groups remained, indicating a possible role for reduced audibility, limited spectral or temporal resolution, or an inability to use auditory source-segregation cues, in directly limiting the ability to listen in the dips of a fluctuating masker. A modified version of the extended speech-intelligibility index that predicts a larger FMB at less favorable SNRs accounted for most of the FMB differences between listener groups and modalities. Overall, these data suggest that HI listeners retain more of an ability to listen in the dips of a fluctuating masker than previously thought. Instead, the fluctuating-masker difficulties exhibited by HI listeners may derive from the reduced FMB associated with the more favorable SNRs they require to identify a reasonable proportion of the target speech.
Velocity dispersion of guided waves propagating in a free gradient elastic plate: Application to cortical bone
Authors: Vavva Maria G. , Protopappas Vasilios C. , Gergidis Leonidas N. , Charalambopoulos Antonios , Fotiadis Dimitrios I. , Polyzos Demosthenes
Published: May 2009

The classical linear theory of elasticity has been largely used for the ultrasonic characterization of bone. However, linear elasticity cannot adequately describe the mechanical behavior of materials with microstructure in which the stress state has to be defined in a non-local manner. In this study, the simplest form of gradient theory (Mindlin Form-II) is used to theoretically determine the velocity dispersion curves of guided modes propagating in isotropic bone-mimicking plates. Two additional terms are included in the constitutive equations representing the characteristic length in bone: (a) the gradient coefficient g, introduced in the strain energy, and (b) the micro-inertia term h, in the kinetic energy. The plate was assumed free of stresses and of double stresses. Two cases were studied for the characteristic length: h=10−4 m and h=10−5 m. For each case, three subcases for g were assumed, namely, g>h, g<h, and g=h. The values of g and h were of the order of the osteons size. The velocity dispersion curves of guided waves were numerically obtained and compared with the Lamb modes. The results indicate that when g was not equal to h (i.e., g≠h), microstructure affects mode dispersion by inducing both material and geometrical dispersion. In conclusion, gradient elasticity can provide supplementary information to better understand guided waves in bones.
Field recordings of Gervais’ beaked whales Mesoplodon europaeus from the Bahamas
Authors: Gillespie Douglas , Dunn Charlotte , Gordon Jonathan , Claridge Diane , Embling Clare , Boyd Ian
Published: May 2009

The first recordings from free-ranging Gervais’ beaked whale (Mesoplodon europaeus) are presented. Nine Gervais’ beaked whales were observed visually for over 6 h. Clicks were only detected over a 15 min period during the encounter, which coincided with an 88 min period during which no whales were observed at the surface. Click lengths were typically around 200 μS and their dominant energy was in the frequency range 30–50 kHz. While these clicks were broadly similar to those of Cuvier’s and Blainville’s beaked whales, the Gervais’ beaked whale clicks were at a slightly higher frequency than those of the other species.