bluuebunny/crossref_metadata_2025 huggingface dataset:
Frequency and time domain Bragg-modulated ray acoustics for truncated periodic arrays
Authors: Felsen Leopold B. , Carin Lawrence
Published: October 2005

Many scenarios in underwater acoustics involve radiation from, or scattering by, configurations with periodic or quasiperiodic features. Depending on the operating conditions, the acoustic field generated by these processes carries the gross imprint of periodicity or quasiperiodicity, without or with effects of truncation, regardless of the detailed structure in each unit cell. To understand and quantify the phenomenology under time harmonic and pulsed excitation, especially in the mid- to high-frequency regime where the width of each cell can cover many wavelengths, it is instructive to explore alternative parametrizations that emphasize, respectively, the nondispersive direct radiation from each cell and the dispersive collective treatment of strict or weakly perturbed periodicity, without and with truncations. The prototype configuration for this two-dimensional study is a finite periodic array of linearly phased parallel filamentary elements. The formulation makes use of Poisson summation and subsequent asymptotics applied to the finite array, and is parametrized in the configuration-spectrum phase space; it highlights the connection between local and global phenomena in both the space-time and wave-number-frequency domains, with a view toward phase-space data processing. Formal aspects and general principles are presented in this paper and are applied to infinite and truncated periodic arrays to illustrate how known results obtained by other methods are recovered with the Poisson-based algorithm. The outcome is a new frequency and time domain Bragg-modulated ray acoustic model that generalizes the nonuniform and uniform ray fields of the geometrical theory of diffraction, so as to include effects of periodic dispersion, with truncations. Under transient conditions, such dispersion gives rise to new time domain Bragg modes.
Travel time surface of a transverse cusp caustic produced by reflection of acoustical transients from a curved metal surface in water
Authors: Frederickson Carl K. , Marston Philip L.
Published: October 2005

The transient response of a wave front that forms a transverse cusp caustic is studied. The travel time surface of a simple cuspoid caustic is known to have the general shape of the singular surface of the next higher order cuspoid catastrophe. Though the transverse cusp wave front considered here is curved in two dimensions, it is also shown to have a travel time surface with the general shape of the singular surface of the swallow tail catastrophe. Ultrasonic experiments to image the travel time surface of the transverse cusp caustic were performed. The imaged travel time surface has the shape of the swallow tail singular surface imposed on a slowly varying travel time surface due to the spherical nature of the source. Imaged cuts through the travel time surface compare very well to calculated travel time curves. The calculated curves are a superposition of a smooth background contribution and a contribution due to the shape of the reflecting surface. It is the second contribution that describes the merging and disappearance of rays as the cusp caustic is crossed. There are no adjustable parameters used in the comparison. It is possible to infer at least qualitative information about the reflecting surface from the imaged travel time surface.
Numerical simulation of interface waves by high-order spectral modeling techniques
Authors: Priolo Enrico , Carcione José M. , Seriani Géza
Published: October 2005

Few problems in elastodynamics have a closed-form analytical solution. The others can be investigated with semianalytical methods, but in general one is not sure whether these methods give reliable solutions. The same happens with numerical techniques: for instance, finite difference methods solve, in principle, any complex problem, including those with arbitrary inhomogeneities and boundary conditions. However, there is no way to verify the quantitative correctness of the solutions. The major problems are stability with respect to material properties, numerical dispersion, and the treatment of boundary conditions. In practice, these problems may produce inaccurate solutions. In this paper, the study of complex problems with two different numerical grid techniques in order to cross-check the solutions is proposed. Interface waves, in particular, are emphasized, since they pose the major difficulties due to the need to implement boundary conditions. The first method is based on global differential operators where the solution is expanded in terms of the Fourier basis and Chebyshev polynomials, while the second is the spectral element method, an extension of the finite element method that uses Chebyshev polynomials as interpolating functions. Both methods have spectral accuracy up to approximately the Nyquist wave number of the grid. Moreover, both methods implement the boundary conditions in a natural way, particularly the spectral element algorithm. We first solve Lamb’s problem and compare numerical and analytical solutions; then, the problem of dispersed Rayleigh waves, and finally, the two-quarter space problem. We show that the modeling algorithms correctly reproduce the analytical solutions and yield a perfect matching when these solutions do not exist. The combined modeling techniques provide a powerful tool for solving complex problems in elastodynamics.
Leaky torsional acoustic modes in infinite clad rods
Authors: Viens M. , Tsukahara Y. , Jen C. K. , Cheeke J. D. N.
Published: October 2005

Dispersion curves, leakage loss, and displacement profiles obtained from the exact analysis of leaky torsional acoustic modes in infinitely thick cladded rods are presented. Two different structures, in which both the longitudinal and the shear velocities of the core are either larger or smaller than those of the cladding, are considered. Similarities are observed between results obtained and dispersion curves which have been reported previously by other workers. Analytic loss relations are also reported for specific modes at either the low- or the high-frequency limit. Applications for this analysis are discussed.
Numerical simulations of piano strings. I. A physical model for a struck string using finite difference methods
Authors: Chaigne Antoine , Askenfelt Anders
Published: October 2005

The first attempt to musical by solving the of by means of (FDM) was made by Hiller and Ruiz [J. Audio Eng. Soc. 19, 462–472 (1971)]. It is shown here how this numerical approach and the underlying physical can be improved in order to simulate the motion of the string with a high degree of realism. Starting from the fundamental of a damped, stiff string interacting with a nonlinear hammer, a numerical finite difference scheme is derived, from which the time histories of string displacement and velocity for each point of the string are computed in the time domain. The interacting force between hammer and string, as well as the force acting on the bridge, are given by the same scheme. The performance of the is illustrated by a few examples of simulated string waveforms. A brief discussion of the aspects of numerical stability and dispersion with reference to the proper choice of sampling parameters is also included.
Ray theory applied to the internal caustic identification of liquid cavities acoustically insonified
Authors: de Billy M. , Piet J. F. , Mahaut S. , Quentin G.
Published: October 2005

The acoustic field in water-filled cylindrical cavities machined inside a duraluminum matrix is studied. The internal acoustic beam focused near the cusp points within the liquid cavity. An optical process (pulsed schlieren technique) is used to visualize the acoustic wave front as the wave propagates in the fluid cylinder. The axial positions of the p-ray cusp caustics are compared with the theoretical values calculated by the ray theory [J. A. Lock and E. A. Hovenac, J. Opt. Soc. Am. A 8, 1541–1552 (1991)].
Application of the Padé approximant technique to the perturbation theory for rough surface scattering
Authors: Kazandjian Luc
Published: October 2005

The Padé approximant technique is applied to the series yielded by the classical perturbation theory in the case of two-dimensional scattering of an acoustic plane wave from a sinusoidal boundary with either a Dirichlet condition or a Neumann condition. The method proposed is numerically compared with other implementations of the perturbation theory. It is found that this method enlarges the domain of validity of the perturbation method and can allow considerably faster computations. Comments are made about the applicability of the Padé approximant technique to three-dimensional scattering.
The effects of masking on vibrotactile temporal summation in the detection of sinusoidal and noise signals
Authors: Gescheider G. A. , Hoffman K. E. , Harrison M. A. , Travis M. L. , Bolanowski S. J.
Published: October 2005

Thresholds for the detection of vibrotactile signals of varied duration applied to the thenar eminence were measured in the absence of and in the presence of a masking stimulus. Signals were 250- and 500-Hz sinusoids and noise bursts with bandwidth limited to 250–1000 Hz. The masking stimulus was either a 250-Hz sinusoid, which was presented in phase with the signal when it was sinusoidal, or noise. Changes in threshold as a function of changes in signal duration were found which were predicted accurately from Zwislocki’s theory [J. Acoust. Soc. Am. 32, 1046–1060 (1960)] of temporal summation when the signal was detected by the Pacinian channel, but not when it was detected by a non-Pacinian channel (NP). However, when either the signal or the masking stimulus or both were noise, NP thresholds were affected by changes in signal duration. Only when the signal and masking stimuli were both sinusoids were NP thresholds independent of signal duration. It is concluded that signal duration effects in the NP channel are not due to temporal integration, but rather to increases in information about the signal content provided to the subject as exposure duration is increased.
A quantitative model of voice F control
Authors: Farley Glenn R.
Published: October 2005

A mathematical model of the larynx, based on biomechanical principles, is described. Components represented include two cartilage elements (cricoid with locked arytenoids, and thyroid), three muscles (thyroarytenoid [TA], cricothyroid pars rectus [CTr], and cricothyroid pars oblique [CTo]), and two ligaments (cricothyroid and vocal ligaments), as well as subglottal pressure (PS). For any combination of muscle activities and PS level, equilibrium positions and tensions could be calculated for components in the system. The tensions and lengths of vocal fold elements were then used to calculate fundamental frequency (F0) of vocal fold vibration. Systematic variation of model muscle activation and PS patterns allowed study of the behavior of the model. TA activity tended to shorten the vocal folds; increased levels of CTr and CTo activity, and PS, had the opposite effect. Increased activity of any muscle tended to increase vocal fold tension, while PS increases were mainly ineffective. F0 was generally increased by increased CTr, CTo, and PS values. However, TA activity had a strongly nonmonotonic effect on F0. Best control of F0 could be achieved only by a process of cocontraction of all muscles at low frequencies, followed by sustained contraction of CTr and CTo with decreasing TA activity for F0’s increasing above this low-frequency range. These results are discussed in terms of their possible implications for normal and abnormal voice production, and as a set of constraints for neural modeling efforts.
Effects of first formant onset frequency on [-voice] judgments result from auditory processes not specific to humans
Authors: Kluender Keith R. , Lotto Andrew J.
Published: October 2005

When F1-onset frequency is lower, longer F1 cutback (VOT) is required for human listeners to perceive synthesized stop consonants as voiceless. K. R. Kluender [J. Acoust. Soc. Am. 90, 83–96 (1991)] found comparable effects of F1-onset frequency on the ‘‘labeling’’ of stop consonants by Japanese quail (coturnix coturnix japonica) trained to distinguish stop consonants varying in F1 cutback. In that study, CVs were synthesized with naturallike rising F1 transitions, and endpoint training stimuli differed in the onset frequency of F1 because a longer cutback resulted in a higher F1 onset. In order to assess whether earlier results were due to auditory predispositions or due to animals having learned the natural covariance between F1 cutback and F1-onset frequency, the present experiment was conducted with synthetic continua having either a relatively low (375 Hz) or high (750 Hz) constant-frequency F1. Six birds were trained to respond differentially to endpoint stimuli from three series of synthesized /CV/s varying in duration of F1 cutback. Second and third formant transitions were appropriate for labial, alveolar, or velar stops. Despite the fact that there was no opportunity for animal subjects to use experienced covariation of F1-onset frequency and F1 cutback, quail typically exhibited shorter labeling boundaries (more voiceless stops) for intermediate stimuli of the continua when F1 frequency was higher. Responses by human subjects listening to the same stimuli were also collected. Results lend support to the earlier conclusion that part or all of the effect of F1 onset frequency on perception of voicing may be adequately explained by general auditory processes. Implications concerning VOT variation for stop consonants at different places of articulation or preceding different vowels are discussed.
Auditory supplements to speechreading: Combining amplitude envelope cues from different spectral regions of speech
Authors: Grant Ken W. , Braida Louis D. , Renn Rebecca J.
Published: October 2005

Many listeners with severe-to-profound hearing losses perceive only a narrow range of low-frequency sounds and must rely on speechreading to supplement the impoverished auditory signal in speech recognition. Previous research with normal-hearing subjects [Grant et al., J. Exp. Psychol. 43A, 621–645 (1991)] demonstrated that speechreading was significantly improved when supplemented by amplitude-envelope cues that were extracted from various spectral regions of speech and presented as amplitude modulations of carriers with frequencies at or below the speech band from which the envelope was derived. This experiment assessed the benefit to speechreading provided by pairs of such envelope cues presented simultaneously. In general, greater improvements in speechreading scores were observed for pairs than for single envelopes when the carrier signals were chosen appropriately. However, when pairs of envelope signals were transposed to low frequencies, the benefit to speechreading was no better than the most effective single-band envelope signal tested, or for a low-pass-filtered speech signal with the same overall bandwidth. Suggestions for improving the efficacy of frequency-lowered envelope cues for hearing-impaired listeners are discussed.
On eigenmodes of the violin—Electronic holography and admittance measurements
Authors: Jansson Erik V. , Molin Nils-Erik , Saldner Henrik O.
Published: October 2005

The present experimental investigation, using recently assembled advanced electro-optical equipment for vibration analysis of three violins, was conducted to seek answers to three questions. A general or global question: Which parts of the violin body are vibrating the most? And two questions related to tonal quality: Are basic low-frequency vibration modes of a musically superior instrument different from those of an inferior violin? Can some special vibration properties be found to support the ‘‘bridge hill?’’ Optically obtained vibration modes were recorded as well as frequency responses in the form of admittance measurements. The investigation showed that the vibration modes found earlier are representative both for the inferior violin and the musically superior instruments, although discrepancies can be seen, both in eigenmode shapes and admittance responses. The experimental results are also in quite good agreement with published results of the modal analysis of a violin. Further, the experimental results indicate that the transversal vibrations are mainly within the plates, but at low frequencies, the vibrations of the edges and of the ribs can be large and in-plane as well as transversal. At higher frequencies, the transversal vibration amplitudes are small at the plate edges and larger inside. The top plate tends to have the largest amplitude of vibrations. In the 2.5-kHz range the violin with the most clear ‘‘bridge hill’’ tends to have the largest vibrations of the plates.
Surface acoustic waves as high-frequency sound transducers in gases
Authors: Stanzel Richard , von Schickfus Manfred
Published: October 2005

In order to increase the frequency range of acoustic measurements in gases surface acoustic waves have been used for excitation and detection of sound waves. With this technique the acoustic attenuation in He, Ar, and He–Ar mixtures at 180 MHz was measured. The results will be compared with data from the literature, obtained with piston-like moving transducers (x-cut quartz or Sell transducers) at frequencies below 10 MHz.
Reply to: ‘‘Comments on ‘Predicting theatre chair absorption from reverberation chamber measurements’ ’’ [J. Acoust. Soc. Am. 93, 2238–2240 (1993)]
Authors: Bradley J. S.
Published: October 2005

Sound absorption coefficients of samples of theatre chairs are linearly related to the sample perimeter/area ratio due to both edge absorption and diffraction effects. The method of using screens to minimize the edge absorption of samples of theatre chairs may be practically useful, but is subject to experimental uncertainties. Predicting the effect of theatre chairs in an auditorium by extrapolating from the measurements of several different sized samples of chairs is still thought to be the most accurate approach.
Precision measurement of the surface acoustic wave velocity on silicon single crystals using optical excitation and detection
Authors: Coufal Hans , Meyer Kevin , Grygier Robert K. , Hess Peter , Neubrand Achim
Published: October 2005

Surface acoustic wave (SAW) and pseudo surface acoustic wave (PSAW) pulses were excited on the (001) and (111) planes of silicon single crystals using ps laser pulses of 248-nm wavelength. An optical beam deflection technique with a bandwidth of 600 MHz was developed for the detection of SAW with high spatial resolution. With this technique the SAW velocities on these silicon planes were determined with an accuracy of 0.1% in various directions of propagation.
Least-squares target detection for a twin-line towed array
Authors: Bucker Homer
Published: October 2005

A least-squares algorithm for detecting acoustic targets with a twin-line towed array is described. Target parameters, the self-noise level, and the line shapes are adjusted for a minimum difference between measured and calculated elements of the covariance matrix.
Acoustic pulse compression using passive phase-conjugate processing
Authors: Dowling David R.
Published: October 2005

Passive phase-conjugate (PPC) processing consists of using the first arrival, from a stream of pulses that have traversed a complex refractive medium, as a filter for later pulse arrivals. Its connection with true ‘‘active’’ phase conjugation lies in invoking reciprocity for the backpropagation. When the intervening acoustic medium changes slowly, the PPC processor creates an unambiguous temporal peak even when a complex multipath environment separates the acoustic source and receiver. Temporal sidelobes in the PPC processor output can be suppressed by increasing the number of receivers and coherently summing their output. Recent deep-ocean acoustic propagation measurements (IWAC ’90) are used to assess PPC processing in a fading multipath acoustic channel. For a range of 272.3 km at 460 Hz, the compressed peak persists on average for about half an hour. Fading of the PPC processor peak is found to be consistent with parabolic-equation second-moment theory for wave propagation in a random medium.
Resonance frequencies of the human skull in vivo
Authors: Håkansson Bo , Brandt Anders , Carlsson Peder , Tjellström Anders
Published: October 2005

Patients with skin penetrating titanium implants in the temporal bone, for attachment of bone-anchored hearing aids, have made it possible to investigate the free-damped natural frequencies (resonance frequencies) of the human skull in vivo. The resonance frequencies of the skull of six subjects were investigated. The resonance frequencies were extracted from two frequency response functions (acceleration/force) measured on each subject: One point measurement where the force and acceleration were both measured at the same point, and one transcranial measurement where the acceleration was measured contralaterally. Between 14 and 19 resonance frequencies were identified for each subject in the frequency range 500 Hz to 7.5 kHz. The two lowest resonance frequencies were found to be on the average 972 (range 828–1164) and 1230 (range 981–1417) Hz. The relative damping coefficients of all resonances were found to be between 2.6 and 8.9%. Due to the relatively high damping coefficients, it is assumed that the resonance frequencies do not significantly affect bone conducted sound. In the transcranial measurements, however, a few large antiresonances were found which may affect bone-conducted sound. Intersubject variations were large, probably due to individual variations in skull geometry and in mechanical parameters. The results were shown to be consistent with previous results obtained on dry skulls. No obvious correlation between lowest resonance frequency and skull size was found.
Discrimination of random, time-varying spectra with statistical constraints
Authors: Lutfi Robert A.
Published: October 2005

In the present study, a sample-discrimination procedure is used to examine how various statistical constraints on random, time-varying spectra might aid listeners in discriminating among these signals. Time-varying spectra were constructed by adding five sequences of tones, five tones in each sequence, centered at the octave frequencies from 250 to 4000 Hz. The levels, frequencies, and durations of the tones varied randomly from one presentation to the next. The signal to be detected was an increment in the level of tones comprising the sequence centered at 1000 Hz (the target sequence). Three main conditions were examined in which the parameter values of tones varied independently of one another (equal-variance condition), covaried across tone sequences (cross-channel covariance condition), or covaried among themselves for each tone (cross-parameter covariance condition). For the latter two conditions, two special cases were examined in which the target sequence varied independently of nontarget sequences or varied according to the same statistical rules. For five of six listeners, comparisons of performance to that of an ideal, within-channel observer gave clear evidence of an ability to take advantage of cross-channel statistical constraints, but only when the target sequence varied independently of its context. Results offer no evidence for listeners’ ability to take advantage of cross-parameter constraints. Trial-by-trial analyses in the equal-variance condition identify factors other than the appropriate weighting of information as primarily responsible for suboptimal performance, though individual differences complicate this interpretation. A failure of simple detection-theoretic models to account for these results is discussed in terms of possible perceptual rules governing sound source identification of real-world objects.
Annoyance perception of sound and information extraction
Authors: Berglund Birgitta , Harder Kathleen , Preis Anna
Published: October 2005

The judgment of annoyance of distorted speech differs radically for different language groups. The results show that those who do comprehend a spoken language, base their annoyance judgments on the informational content extracted while those who do not base it on the perceptual characteristics of meaningless sound (particularly loudness). A series of distorted German speech sounds were presented to two subject groups consisting of native Swedish and English speakers, and the results were compared with earlier results from groups of native German and Polish subjects. The 50 stimuli were generated from the very same speech signal distorted in two principle ways, either with repeated silent gaps or superimposed noise impulses. The perceived annoyance of the distorted speech was judged both by category scaling for all subject groups, and as a control for ‘‘ceiling’’ effects, also by magnitude estimation for the Swedish and the English subjects. There is a pronounced tendency for German subjects to judge the German speech distorted with silent gaps as more annoying than that distorted with superimposed noise impulses. In contrast, the Swedish, English, and Polish subjects judged the two German-speech distortions in reversed order with regard to annoyance. Thus for noncomprehending listeners, noise-distorted speech is more annoying but for comprehending listeners it is speech distorted by gaps. This means that impaired communication intrusiveness rather than loudness predominates in annoyance judgements from comprehending listeners.
Psychoacoustic equivalence of frequency modulation and quasi-frequency modulation
Authors: Edwards Brent W. , Viemeister Neal F.
Published: October 2005

Frequency modulation (FM) is known to be reasonably approximated by quasi-frequency modulation (QFM) for small modulation indices, β, but the range of β for which this approximation is appropriate is unclear. Thresholds for discrimination between FM and QFM with equal β’s are obtained in order to estimate an upper bound on β for determining when this approximation is valid psychoacoustically, i.e., when FM and QFM are indiscriminable. At low modulation frequencies (fm≤4 Hz), QFM is never a valid approximation to FM at any detectable modulation level since discrimination thresholds are below FM detection thresholds. For modulation frequencies between 8 and 32 Hz, discrimination thresholds are approximately −2.5 dB (20 log β) and can be accounted for by detection of envelope fluctuations in the QFM signal. For modulation frequencies at 64 Hz and above, discrimination thresholds improve with increasing modulation frequency in the same manner as FM and QFM detection thresholds. Discrimination in this frequency region seems to be mediated by detection of the component 2fm Hz below the carrier, i.e., by the most detectable component in the FM signal which does not occur in the QFM signal.
Level-dependent critical bandwidth for phase discrimination
Authors: Nelson David A.
Published: October 2005

Monaural phase discrimination was evaluated at 1000 Hz in six normal-hearing listeners as a function of the frequency difference between components in three-tone complexes at 40, 60, 80, and 100 dB SPL. The phase of the center component of 100% sinusoidally amplitude-modulated (SAM) waveforms was shifted by 90° to produce quasi-frequency-modulated (QFM) waveforms that had identical long-term power spectra to the SAM waveforms but with different amplitude envelopes and temporal fine structure. At low modulation frequencies, where spectral components were close together and presumably all well within a single auditory filter, normal-hearing listeners could easily discriminate QFM from SAM waveforms. As modulation frequency increased, a point was reached where listeners could no longer distinguish QFM from SAM waveforms, referred to here as the critical bandwidth for phase discrimination (CBphs). Discrimination performance (d′) was measured as a function of modulation frequency to yield a psychometric function for phase discrimination. From that psychometric function, CBphs was defined as the modulation frequency corresponding to d′=1.0. At a carrier frequency of 1000 Hz, CBphs increased with level between 40 and 80 dB SPL according to the relation: CBphs=34⋅I(0.136). Above 80 dB SPL, very little change in CBphs with level was seen. The increase with level of the geometric mean CBphs was predicted from level-dependent auditory filter slopes inferred from forward-masked tuning curves, as was the tendency to reach an asymptote above 80 dB SPL. Comparisons with previous work indicate that CBphs values from the best performing subjects were well within the audibility region for cubic difference tones. It is proposed that internally generated cubic difference tones interact with externally generated acoustic components, both limited by a level-dependent auditory filter, to produce an internal excitation envelope that is the basis for discriminating between SAM and QFM waveforms. It is also suggested that individual differences at low levels may be due to internal phase ambiguities.
The measurement of the lateralization of narrow bands of noise using an acoustic pointing paradigm: The effect of sound-pressure level
Authors: Simon Helen J. , Collins Carter C. , Jampolsky Arthur , Morledge David E. , Yu Jenny
Published: October 2005

The effects of varying interaural time delay (ITD) and interaural intensity difference (IID) were measured in normal-hearing subjects as a function of eleven frequencies and at sound-pressure levels (SPL) from 60 to 90 dB SPL and at 25-dB sensation level. Using an ‘‘acoustic’’ pointing paradigm, the IID of a 500-Hz narrow-band (100 Hz) noise (the ‘‘pointer’’) was varied by the subject to coincide with that of a ‘‘target’’ ITD stimulus. ITDs of 0, ±200, and ±400 μs were obtained through total waveform delays of narrow-band noise (NBN), including envelope and fine structure. The results of this experiment confirm the traditional view of binaural hearing for like stimuli: There is little perceived displacement away from 0 IID at frequencies of 1250 Hz and above. In the low frequencies, subjects required IIDs greater than the expected 10 dB to perceive a fully lateralized image, and they varied in the maximum value of IID that they required, regardless of frequency. Our subjects did not always perceive the intracranial locations of ITD targets symmetrically: When the signal was delayed to one ear, the resultant matching IID was often different in magnitude than for the same ITD target delayed to the opposite ear for the identical frequency. The results of two subjects suggested that people with asymmetric normal hearing have adapted to their asymmetry for lateralization tasks: The subjects were found to lateralize toward the ear with the greater SPL stimulus, regardless of the ear to which the signal was delayed, when signals of equal SL were presented, and toward the leading ear when signals of equal SPL were presented (unequal SL). Increasing the presentation levels above 60 dB SPL had an effect on the perception of high-frequency ITD targets: As the intensity level increased, the slopes of the IID versus ITD functions increased indicating better discrimination of ITD. This study is in agreement with other studies in providing strong evidence of individual differences in lateralization experiments. These individual differences might be attributable to differential sensitivity to ambiguous time stimulus cues, differential task sensitivity, age effects, threshold asymmetries, or criterion variability.
Vibrotactile and electrotactile perception of time-varying pulse trains
Authors: Summers Ian R. , Dixon Philip R. , Cooper Philip G. , Gratton Denise A. , Brown Brian H. , Stevens John C.
Published: October 2005

To establish the best strategy for transmitting speech-derived information via a single tactile channel, measurements were made on the perception of frequency- and/or amplitude-modulated pulse-train stimuli, with a comparison of the electrotactile and vibrotactile modalities. In one experiment, vibrotactile perception of 2-oct step changes in stimulus frequency was found to be significantly better than electrotactile on a timescale appropriate for the transmission of speech features (e.g., with practiced subjects, information transfer of 69% with 200-ms vibrotactile stimuli, 32% with 200-ms electrotactile stimuli). Perception of step changes in stimulus amplitude was similar in the two modalities when changes in amplitude were tailored to match the different dynamic ranges available. In a second experiment, vibrotactile perception of voice fundamental frequency with various codings was investigated. Both experiments showed information transfer for vibrotactile stimuli to be greater when frequency and amplitude modulation were used together rather than with one or the other in isolation (sentence-stress identification scores: 66% for FM stimuli, 69% for AM stimuli, 80% for FM/AM stimuli). It is concluded that frequency- and amplitude-modulated vibratory stimulation is a good choice in a practical device for the profoundly hearing impaired.
Perceptual and computational separation of simultaneous vowels: Cues arising from low-frequency beating
Authors: Culling John F. , Darwin C. J.
Published: October 2005

Identification of simultaneous speech sounds, such as pairs of steady-state vowels (double vowels), is more accurate when there is a difference in fundamental frequency (F0). Accuracy of identification for double vowels increases with increasing F0 difference (ΔF0) asymptoting above 1 semitone. The experiment described here attempts to distinguish two mechanisms underlying this effect: first, perceptual separation by grouping together harmonic components of a common F0; and, second, exploitation of the fluctuations in the spectral envelope of the composite stimulus that result from beating between unresolved components. The beating is mainly caused by interactions between corresponding harmonics of the two vowels with a small ΔF0. Identification accuracy for normal, harmonically excited double vowels was compared with that for double vowels composed from the same components, but whose constituent vowels were excited by a mixture of the two harmonic series. These double vowels were designed to produce similar beating patterns to the normal double vowels. Both harmonically and inharmonically excited constituents improved identification with increasing ΔF0, but the increase was larger for harmonically excited vowels. A computational model based upon psychophysical measurements of auditory frequency and temporal resolution correctly predicted an increase in accuracy of identification with increasing ΔF0 which was attributable to beating. The results are interpreted in terms of a spectral change cue in the identification of double vowels with ΔF0’s which complements grouping by F0, and which plays a dominant role for ΔF0’s smaller than 1 semitone.
Intelligibility of conversational and clear speech in noise and reverberation for listeners with normal and impaired hearing
Authors: Payton K. L. , Uchanski R. M. , Braida L. D.
Published: October 2005

The effect of articulating clearly on speech intelligibility is analyzed for ten normal-hearing and two hearing-impaired listeners in noisy, reverberant, and combined environments. Clear speech is more intelligible than conversational speech for each listener in every environment. The difference in intelligibility due to speaking style increases as noise and/or reverberation increase. The average difference in intelligibility is 20 percentage points for the normal-hearing listeners and 26 percentage points for the hearing-impaired listeners. Two predictors of intelligibility are used to quantify the environmental degradations: The articulation index (AI) and the speech transmission index (STI). Both are shown to predict, reliably, performance levels within a speaking style for normal-hearing listeners. The AI is unable to represent the reduction in intelligibility scores due to reverberation for the hearing-impaired listeners. Neither predictor can account for the difference in intelligibility due to speaking style.
Speech enhancement based on physiological and psychoacoustical models of modulation perception and binaural interaction
Authors: Kollmeier Birger , Koch René
Published: October 2005

A novel approach for analyzing and filtering speech is described and evaluated which utilizes the ‘‘modulation spectrogram,’’ i.e., the two-dimensional representation of modulation frequencies versus center frequency as a function of time. This approach is based on physiological findings of a tonotopical organization of modulation frequencies perpendicular to carrier frequencies as well as psychoacoustical findings of ‘‘modulation tuning curves.’’ In addition, an interaction is assumed between the representation of modulation frequencies and the representation of auditory space as described by physiological and psychological models of binaural hearing. A noise-reduction algorithm based on this approach was implemented and tested which enhances or suppresses each combination of modulation frequency and center frequency according to its phase and intensity relation between the two input signals (i.e., both stereo channels of a dummy-head recording). When tested in several situations with interfering speakers and background noise both in anechoic and reverberant environment, the algorithm provided a small but a very robust increase in speech intelligibility which corresponds to approximately 2 dB in signal-to-noise ratio. Possible applications of this algorithm are noise reduction in adverse acoustical situations, digital hearing aids, processing schemes and preprocessing for speech recognition.
Numerical simulations of piano strings. II. Comparisons with measurements and systematic exploration of some hammer-string parameters
Authors: Chaigne Antoine , Askenfelt Anders
Published: October 2005

A physical model of the piano string, using finite difference methods, has recently been developed. [Chaigne and Askenfelt, J. Acoust. Soc. Am. 95, 1112–1118 (1994)]. The model is based on the fundamental equations of a damped, stiff string interacting with a nonlinear hammer, from which a numerical finite difference scheme is derived. In the present study, the performance of the model is evaluated by systematic comparisons between measured and simulated piano tones. After a verification of the accuracy of the method, the model is used as a tool for systematically exploring the influence of string stiffness, relative striking position, and hammer-string mass ratio on string waveforms and spectra.
Shear Q estimates from interface waves in marine sediments
Authors: Broadhead Michael K. , Ali Hassan B. , Bibee L. Dale
Published: October 2005

Scholte wave arrivals at two diverse test sites were analyzed with the spectral ratio method to obtain estimates of average shear Q of deep sea bottom sediments. The Scholte waves were generated by bottom-located explosive charges and recorded on ocean bottom seismometers. The results obtained are compared to Q estimates reported in the literature.
Analysis of the initial buildup of oscillations in a thermoacoustic prime mover
Authors: Atchley Anthony A.
Published: October 2005

Previously, the evolution of the quality factor Q of a prime mover below onset of self-oscillation as a function of the temperature difference applied across the prime mover stack was discussed [A. A. Atchley, H. E. Bass, T. J. Hofler, and H. T. Lin, ‘‘Study of a thermoacoustic prime mover below onset of self-oscillation,’’ J. Acoust. Soc. Am. 91, 734–743 (1992) and A. A. Atchley, ‘‘Standing wave analysis of a thermoacoustic prime mover below onset of self-oscillation,’’ J. Acoust. Soc. Am. 92, 2907–2914 (1992)]. In this paper, these measurements are extended through onset. Above onset Q is determined from measurements of the rise time of the initial buildup of the acoustic oscillations. These measurements, coupled with those previously reported for below onset conditions, show the complete evolution of Q for an individual prime mover from below to above onset. The measurements are in good agreement with the predictions of a standing wave analysis of prime movers discussed in the earlier work. It is suggested that measurements of the buildup of oscillations might be used to study factors that determine steady-state oscillation amplitude in prime movers.
Wave-vector filter analysis of turbulent flow
Authors: Mellen R. H.
Published: October 2005

The Corcos model for convective turbulence is equivalent to the problem of an ensemble of plane waves passing through spatial windows with rhombic iso-amplitude contours. The sharp corners introduce spurious spatial effects analogous to transients in the time domain. The elliptic model was proposed to suppress these artifacts by smoothing the contours [R. H. Mellen, J. Acoust. Soc. Am. 88, 2891–2893 (1990)]. Comparison of the two models is extended to further illustrate some of the implications for wave-vector filter experiments.
Development of an Arctic ray model
Authors: Stotts Steven A. , Koch Robert A. , Bedford Nancy R.
Published: October 2005

An Arctic ray model ICERAY, which is based on a ray model previously developed for deep ocean environments, is described. An acoustic propagation description for Arctic regions is achieved by introducing top layering above the ocean environment and allowing receivers to be located in either the ocean volume or the top (ice) layers. Compressional and shear ray components are included. An evanescent field, in a ray form, is shown to account for the dominant component of the field for long-range propagation to receivers in the ice. Comparisons to an existing full wave model and corresponding analytic expressions demonstrate the accuracy of this ray approach if the evanescent field is included.
Scattering from a rough sedimental seafloor containing shear and layering
Authors: Essen H.-H.
Published: October 2005

A simple first-order perturbation approach is presented, which directly determines the acoustic field scattered by a rough seafloor. The theory is based on the Born approximation. The seafloor is allowed to be shear-supporting or to be layered. Scattering strengths are derived for both monostatic and bistatic scattering, and discussed with respect to their dependence on seafloor parameters and roughness spectrum. Power-law wave-number spectra are found to be appropriate for explaining important features of backscatter, which are the magnitude, the frequency, and grazing-angle dependence. For grazing angles below critical, shear-wave velocities exceeding some 350 m s−1 significantly modify the scattering strength. With respect to narrow-band acoustic signals layering of the seafloor may be of importance. Numerical results are presented for the frequency range 10–100 kHz.
Propagation of transient waves through a stratified fluid medium: Wavelet analysis of a nonasymptotic decomposition of the propagator. Part I. Spherical waves through a two-layered system
Authors: Saracco G.
Published: October 2005

The propagator for the problem of transient waves in a stratified fluid medium has a natural decomposition into three contributions. These contributions are studied in detail by analytic and numerical methods and the results are compared with experiment. Since the individual contributions have dispersive character, they are analyzed with the help of continuous wavelet transforms.
Guided waves in a jointed composite plate
Authors: Karunasena W. , Shah A. H. , Datta S. K.
Published: October 2005

Reflection of guided waves by a thin region of bonding material at the interface between two composite plates has been investigated in this paper. The study is motivated by the need to develop a quantitative ultrasonic technique to characterize properties of joints between two plates, which may be laminated and anisotropic. Although there have been some recent studies that have examined the reflection and transmission of surface waves at the interface between two plates, they consider the two plates to be semi-infinite in thickness. Thus the mode conversion of waves into higher guided modes of the plates are not taken into account. In this paper, a model study of the interaction of the A0 and S0 guided wave modes with the joint material is presented. It is shown that as the frequency increases the coefficient of reflection shows resonant peaks at the cutoff frequencies of higher guided modes. These peaks become increasingly pronounced as the thickness of the joint increases. Another interesting feature is that the reflection coefficients have minima at a frequency that depends on the thickness and the properties of the joint material. These features can be used to determine the relevant properties of the joint.
Cavity flow tones in water
Authors: Burroughs Courtney B. , Stinebring David R.
Published: October 2005

Measurements of tones generated by flow over cavities were conducted in the ARL Penn State small water tunnel. The frequency dependence of cavity tones on the flow velocity and the size of the cavity opening in the streamwise direction are presented and discussed. Measured cavity wall vibration levels, unsteady pressure levels inside the cavity, and flow visualization are used to analyze generating mechanisms for cavity tones and Helmholtz resonances.
Measuring velocity and location of a target by the variable domain transform
Authors: Roosnek Nico
Published: October 2005

In this article an application of the variable domain transform (VDT) method, a new analyzing technique [N. Roosnek, J. Acoust. Soc. Am. 92, 3207–3212 (1992)], is shown to be applicable for the simultaneous measurement of velocity and distance of a target using sonar. The technique is based on the change of the shape of a suitable chosen FM pulse, due to the velocity of the target with respect to the measuring platform. This change can be easily analyzed by the VDT. The resolution of the velocity and of the location is a function of the shape of the FM pulse.
An experimental investigation of the collective oscillations of bubble plumes entrained by breaking waves
Authors: Loewen M. R. , Melville W. K.
Published: October 2005

Laboratory measurements of the sound produced by mechanically generated two-dimensional (2-D) and three-dimensional (3-D) breaking waves are presented. In the 2-D breaking experiments it was observed that the mean-square pressure at frequencies below 1 kHz correlated strongly with the fractional energy dissipated by breaking and the volume of air entrained. In addition, the volume of air entrained was found to be proportional to the fractional energy dissipated. These results imply that measurements of the low-frequency sound may be useful for studying the dynamics of breaking waves in the field. It was found that 2-D plunging breakers produced significant increases in spectral levels at frequencies below approximately 500 Hz but that spilling breakers did not. Large-amplitude low-frequency signals were observed to begin up to 1/3 of a wave period after the start of active breaking in both the two- and three-dimensional experiments. It is believed that these signals were due to the collective oscillation of the entrained bubble plumes. A model of a cylindrically shaped plume located immediately below the free surface and the void-fraction measurements of Lamarre and Melville [J. Acoust. Soc. Am. 95, 1317–1328 (1994)] were used to compute the eigenfrequencies of the volume mode of collective oscillation. The computed eigenfrequencies closely matched the frequencies of the observed signals. This agreement between the experimental observations and theory provides considerable support for the hypothesis that the observed bubble plumes were oscillating collectively in their volume mode. This leads to the conclusion that the collective volume oscillations of bubble plumes entrained by breaking waves may be a source of low-frequency sound in the ocean.