bluuebunny/crossref_metadata_2025 huggingface dataset:
Tempo, stress, and vowel reduction in American English
Authors: Fourakis Marios
Published: October 2005

Two processes that affect the acoustic characteristics of vowels, namely, phonological and phonetic vowel reduction are discussed. Phonological vowel reduction applies to unstressed vowels. Phonetic vowel reduction is supposed to apply to all vowels and be caused by fast speech rates, context, as well as lack of stress. In this experiment, the effects of changes in stress and in rate of speech (tempo) on the acoustic characteristics of American English monophthongal, nonretroflex vowels were examined. Four male and four female native speakers produced these vowels in two contexts, [h _d] and [b _d], in a carrier sentence, under four conditions of tempo stress (slow-stressed, slow-unstressed, fast-stressed, and fast-unstressed). Measurements of duration and fundamental frequency showed that the subjects did, in fact, vary tempo and stress as instructed. The effect of a change in stress on vowel duration was found to be slightly larger than that of a change in tempo. The putative vowel portion of each utterance was analyzed, formant tracks were obtained, and these were plotted in an auditory–perceptual space [J. D. Miller, J. Acoust. Soc. Am. 85, 2114–2134 (1989)]. These plots served to determine the part of the utterance that could, in most cases, be considered its steady state. For each utterance, an average of the coordinates of this steady-state portion was taken and was used to represent the utterance as a point in the auditory–perceptual space. The distance of these data points from the point representing the acoustic characteristics of a vowel produced by a neutral vocal tract was used to determine the magnitude of phonetic vowel reduction caused by faster tempo and less stress, relative to the slow-stressed condition. Although the results indicate that tempo and stress may not have a major influence on the distances of individual vowels from the neutral point, the size of the vowel space overall was affected. The vowel space was largest for the slow stressed condition and smallest for the fast unstressed condition. In addition, several vowel classifications schemes were tested using linear discriminant analysis, and the one proposed by Miller (1989) performed better than other combinations of fundamental frequency and the first three formants.
Gender recognition from speech. Part I: Coarse analysis
Authors: Wu Ke , Childers D. G.
Published: October 2005

The purpose of this research was to investigate the potential effectiveness of digital speech processing and pattern recognition techniques for the automatic recognition of gender from speech segments. In this paper ‘‘coarse’’ acoustic coefficients (autocorrelation, linear prediction, cepstrum, and reflection) were used to form test and reference templates for vowels, voiced fricatives, and unvoiced fricatives. The effects of different distance measures, filter orders, recognition schemes, and vowels and fricatives were comparatively assessed to determine their effectiveness for the task of gender recognition from speech segments. The results showed that most of the acoustic parameters worked well for gender recognition. A within-gender and within-subject averaging technique was important for generating appropriate test and reference templates. The Euclidean distance measure appeared to be the most robust as well as the simplest of the distance measures. The results from this study implied that the gender information is time invariant, phoneme independent, and speaker independent for a given gender. One recognition scheme achieved 100% correct speaker gender classification for a database of 52 talkers (27 male and 25 female). In part II of this paper [J. Acoust. Soc. Am. 90 (1991); hereafter referred to as paper II] the detailed features of ten vowels that appeared responsible for distinguishing a speaker’s gender were examined statistically. Included in paper II is a replication of part of the classical study of Peterson and Barney [J. Acoust. Soc. Am. 24, 175–184 (1952)] of vowel characteristics.
Stimulus features affecting psychophysical detection thresholds for electrical stimulation of the cochlea. I: Phase duration and stimulus duration
Authors: Pfingst Bryan E. , De Haan David R. , Holloway Lisa A.
Published: October 2005

The shapes and levels of psychophysical detection threshold versus frequency functions for sinusoidal electrical stimulation of the deafened cochlea vary from subject to subject. These variations have been shown previously to be correlated with nerve-survival patterns. The shapes of these functions are critical in the design and calibration of analog processors for auditory prostheses. This paper examines two stimulus features that may contribute to the shapes of these functions: phase duration and stimulus duration. Psychophysical experiments were performed with unilaterally deafened and implanted macaque monkeys. Effects of phase duration were studied by measuring thresholds for single symmetric biphasic pulses presented at a rate of 1 per trial. Psychophysical detection thresholds for these pulses decreased, in the region of maximum slope, at a rate averaging −6.2 dB per doubling of phase duration, which is steeper than the rate of decrease in thresholds reported for single auditory-nerve fibers. Slope was weakly correlated with threshold level. Thresholds for long-duration sinusoids were consistently lower and the slopes of the threshold versus phase-duration functions were consistently steeper than those for single pulses. Thresholds for sinusoids and pulse trains decreased as a function of stimulus duration for durations up to at least 300 ms. The rate of decrease as a function of stimulus duration depended on the phase duration of the stimulus, and for long phase-duration signals, it depended on frequency. The rate of decrease was correlated with the absolute-detection-threshold level. We conclude that phase duration contributes significantly to, but does not completely account for, the steep slopes of the threshold versus frequency functions for long-duration sinusoids. Temporal integration is greater for longer phase-duration signals, giving rise to steeper slopes of threshold functions for long-duration sinusoids as compared to those for single pulses in the 1- to 5-ms/phase range. Current neural data and models do not account for the steep slopes of psychophysical threshold functions.
Fringed correlation discrimination and binaural detection
Authors: Jain Manoj , Gallagher David T. , Koehnke Janet , Colburn H. Steven
Published: October 2005

Results are reported from a series of binaural detection and interaural correlation discrimination experiments at 500 Hz. The experiments include fringed correlation discrimination in which the correlation change is restricted to a narrow (38 Hz) target band of frequencies and the reference correlation is maintained in a fringe band of frequencies. This experiment is designed to be analogous to a detection experiment with a narrow-band target; in both cases the correlation changes only inside the target band. A simplified theoretical framework is used to compare the results of the detection and correlation discrimination experiments. Results are consistent with the notion that binaural detection and interaural correlation discrimination are effected by a common mechanism when the reference correlation is unity (as in the NoSπ case). When the reference correlation is zero (as in the NuSo case), detection performance is significantly better than predicted from the measured ability to discriminate interaural correlation.
Variations in the speed of sound and interaural differences of time: An additional source of jitter
Authors: Saberi Kourosh
Published: October 2005

Variations in the speed of sound result in interaural temporal uncertainty which may contribute to localization ambiguity at the extreme peripheral azimuths.
A note on identification of the acoustic ocean transfer function in the Tyrrhenian Sea
Authors: Fradkin Larissa Ju.
Published: October 2005

Recently, exploratory analysis of signatures and responses collected during the NAPOLI 85 trial was conducted [L. Ju. Fradkin, J. Acoust. Soc. Am. 87, 1569 and 1577 (1990)]. It was concluded that an ocean transfer function (OTF) relating the two was a non-negative function of time consisting of two narrow peaks. The second peak was included to compensate for the mismatch in the signature and response shapes. In the absence of an alternative explanation, the peak was attributed to a lenslike feature in the ocean. It transpires now that the responses were preprocessed with a different hardware filter to signatures. Recalibration removes most of the mismatch and therefore a need for a second peak. Significantly, other conclusions on the OTF behavior at high frequencies still stand.
Direct digital-to-analog conversion of acoustic signals using a solid dielectric transducer/filter system
Authors: Scott Morgan M. , Bolen Lee N.
Published: October 2005

This paper describes the theory, construction, and operation of a transducer that is capable of transmitting an analog acoustic signal when driven by a digital input signal. The digital signal drives a capacitive dielectric transducer whose active surface area is proportional to the magnitude of the digital signal. The output of the transducer contains the original digitized analog signal in addition to higher frequency Fourier components that result from the sampling process. The unwanted additional signal is attenuated by coupling the transducer output into an acoustic low-pass filter whose response is determined by using electrical circuit analysis techniques. The results indicate that the experimental transducer system is useful for generating acoustic signals that have an upper frequency limit of 8 kHz. With suitable modifications, wider frequency performance is attainable.
Studies of the speech transducing performance of carbon and electret telephone transmitters
Authors: James James H. , Maxwell James H. , Whitesell Stephen R.
Published: October 2005

Five studies examined the relative performance of traditional carbon-type telephone transmitters and newer electret-type transmitters that have come to replace the carbon-type in contemporary telephone set designs. Focus is on the validity of the transmit objective loudness rating (TOLR) metric as a predictor of transmitter efficiency in response to real speech input. TOLR is the IEEE recommended metric (IEEE Standard 269-1983) for objectively characterizing the efficiency with which telephone transmitters convert acoustic energy to electric energy. These studies indicate that the relationship between TOLR and transmitter efficiency for real speech is not the same for carbon and electret transmitters. Relative to what is observed with carbon transmitters, TOLR was shown to underpredict the real speech efficiency of electret transmitters by approximately 3 dB. Two mechanisms are involved in this effect. First, carbon transmitters have a lower peak-clipping, or compression, threshold than electret transmitters. This difference is not accounted for in the measurement of TOLR where the input signal consists of a swept sine wave whose constant amplitude (sound-pressure level of 88 dB) is below the compression threshold of carbon transmitters. Second, the efficiency of carbon transmitters, but not electret transmitters, declines during the first minutes after the telephone handset is taken off-hook. The formal TOLR measurement protocol overestimates carbon transmitter efficiency because the transmitter is characterized immediately after being taken off-hook, not over a broader period of time which would be more predictive of its efficiency in real use. These results have implications for the choice of TOLR design objectives for electret transmitter telephone sets. The greater relative efficiency of electret transmitters for real speech input also needs to be considered when grade-of-service models, which now assume the use of carbon transmitter telephone sets, are used to estimate the transmission quality of network connections.
The use of prosody in syntactic disambiguation
Authors: Price P. J. , Ostendorf M. , Shattuck-Hufnagel S. , Fong C.
Published: October 2005

Prosodic structure and syntactic structure are not identical; neither are they unrelated. Knowing when and how the two correspond could yield better quality speech synthesis, could aid in the disambiguation of competing syntactic hypotheses in speech understanding, and could lead to a more comprehensive view of human speech processing. In a set of experiments involving 35 pairs of phonetically similar sentences representing seven types of structural contrasts, the perceptual evidence shows that some, but not all, of the pairs can be disambiguated on the basis of prosodic differences. The phonological evidence relates the disambiguation primarily to boundary phenomena, although prominences sometimes play a role. Finally, phonetic analyses describing the attributes of these phonological markers indicate the importance of both absolute and relative measures.
Lipreading sentences with vibrotactile vocoders: Performance of normal-hearing and hearing-impaired subjects
Authors: Bernstein Lynne E. , Demorest Marilyn E. , Coulter David C. , O’Connell Michael P.
Published: October 2005

Three vibrotactile vocoders were compared in a training study involving several different speech perception tasks. Vocoders were: (1) the Central Institute for the Deaf version of the Queen’s University vocoder, with 1/3-oct filter spacing and logarithmic output scaling (CIDLog) [Engebretson and O’Connell, IEEE Trans. Biomed. Eng. BME-33, 712–716 (1986)]; (2) the same vocoder with linear output scaling (CIDLin); and (3) the Gallaudet University vocoder designed with greater resolution in the second formant region, relative to the CID vocoders, and linear output scaling (GULin). Four normal-hearing subjects were assigned to either of two control groups, visual-only control and vocoder control, for which they received the CIDLog vocoder. Five normal-hearing and four hearing-impaired subjects were assigned to the linear vocoders. Results showed that the three vocoders provided equivalent information in word-initial and word-final tactile-only consonant identification. However, GULin was the only vocoder significantly effective in enhancing lipreading of isolated prerecorded sentences. Individual subject analyses showed significantly enhanced lipreading by the three normal-hearing and two hearing-impaired subjects who received the GULin vocoder. Over the entire training period of the experiment, the mean difference between aided and unaided lipreading of sentences by the GULin aided hearing-impaired subjects was approximately 6% words correct. Possible explanations for failure to confirm previous success with the CIDLog vocoder [Weisenberger et al., J. Acoust. Soc. Am. 86, 1764–1775 (1989)] are discussed.
The effect of linguistic entropy on speech perception in noise in young and elderly listeners
Authors: van Rooij John C. G. M. , Plomp Reinier
Published: October 2005

The rationale for a method to quantify the information content of linguistic stimuli, i.e., the linguistic entropy, is developed. The method is an adapted version of the letter-guessing procedure originally devised by Shannon [Bell Syst. Tech. J. 30, 50–64 (1951)]. It is applied to sentences included in a widely used test to measure speech-reception thresholds and originally selected to be approximately equally redundant. Results of a first experiment reveal that this method enables one to detect subtle differences between sentences and sentence lists with respect to linguistic entropy. Results of a second experiment show that (1) in young listeners and with the sentences employed, manipulating linguistic entropy can result in an effect on SRT of approximately 4 dB in terms of signal-to-noise ratio; (2) the range of this effect is approximately the same in elderly listeners.
Reiterant speech as a test of non-native speakers’ mastery of the timing of French
Authors: Levitt Andrea G.
Published: October 2005

The reiterant speech of ten native speakers of French was analyzed to develop baseline measures for syllable and consonant/vowel timing for a series of two-, three-, four-, and five-syllable French words spoken in isolation. Ten native speakers of English, who learned French as a second language, produced reiterant versions of both the French words and a comparable set of English words. The native speakers of English were divided into two groups on the basis of their second language experience. The first group consisted of four university-level teachers, who were relatively experienced learners of French, and the second group of six less experienced learners of French. The French reiterant imitations of the two groups of native speakers of English were compared to the native French speakers’ productions. The timing patterns of the experienced group of non-native speakers did not differ significantly from those of the native French speakers, whereas there was a significant difference between these two groups and the group of six less experienced second-language learners. Deviations from the French baseline measures produced by the less experienced group are discussed in terms of the influence of the timing patterns of English and the literature on a sensitive period for second language acquisition.
Lateralization of low-frequency tones: Relative potency of gating and ongoing interaural delays
Authors: Buell Thomas N. , Trahiotis Constantine , Bernstein Leslie R.
Published: October 2005

Several types of interaural delay can affect the lateral position of binaural signals. Delays can occur within the gating (onset and/or offset) or ongoing portions of the signal, or both. Extent of laterality produced by each of these delays was measured for low-frequency tones with an acoustic pointing task. Relative potency was assessed by presenting the delays singly or in combinations (where the types of delay were consistent or in opposition). Rise/decay time, duration, and frequency of the tonal targets were also varied. The major finding was that ongoing delays were much more potent than gating delays in determining extent of laterality. Gating delays were most effective when the interaural phase of the ongoing portion of the tones was more or less ambiguous with respect to which ear was leading. Many of our findings are qualitatively well described by considering properties of patterns of activity produced within a cross-correlation network by such interaurally delayed signals.
Low-dispersion finite difference methods for acoustic waves in a pipe
Authors: Davis Sanford
Published: October 2005

A new algorithm for computing one-dimensional acoustic waves in a pipe is demonstrated by solving the acoustic equations as an initial-boundary-value problem. Conventional dissipation-free second-order finite difference methods suffer severe phase distortion for grids with less that about ten mesh points per wavelength. Using the signal generated by a piston in a duct as an example, transient acoustic computations are presented using a new compact three-point algorithm which allows about 60% fewer mesh points per wave length. Both pulse and harmonic excitation are considered. Coupling of the acoustic signal with the pipe resonant modes are shown to generate a complex transient wave with rich harmonic content.
A numerical solution for the general radiation problem based on the combined methods of superposition and singular-value decomposition
Authors: Fahnline John B. , Koopmann Gary H.
Published: October 2005

In the method of wave superposition, the acoustic field, due to a complex radiator, is expressed in terms of a Fredholm integral equation of the first kind called the ‘‘superposition integral equation.’’ In general, Fredholm integral equations of the first kind are ill-posed and therefore difficult to solve numerically. In this paper, it will be shown that a simple collocation procedure, when combined with the singular-value decomposition, can yield accurate results for the numerical solution of the superposition integral. As an example of the application of the method, the acoustic radiation from a circular cylinder will be analyzed using this numerical procedure and compared to the exact solution. It is shown that, for this problem, the accuracy of the numerical solution can be judged by evaluating how well the superposition solution approximates the specified boundary condition on the surface of the radiator. An example is also given of a problem which has no exact solution. In this situation, it is suggested, without proof, that the accuracy of the numerical solution can be judged in a similar manner by evaluating the error in the superposition solution’s satisfaction of the boundary condition.
Localization of azimuthal sound direction by the great horned owl
Authors: Beitel Ralph E.
Published: October 2005

Localization of azimuthal sound direction was studied in the great horned owl. Head orienting responses (ORs) initiated following brief stimuli and ORs completed during long duration stimuli had similar mean localization errors. The accuracy of the OR apparently was not influenced by response-produced auditory feedback. For sources of sound located in the anterior sound hemifield, errors were relatively small (mean=7.0°) and the magnitude of ORs was correlated significantly with azimuth. Localization of sources in the posterior sound hemifield was poor (mean error=41.1°). However, ORs occasionally were followed by secondary, corrective responses during trials with long duration stimuli. Corrective responses significantly reduced errors for sources located in the posterior sound hemifield.
Acoustics of a cavity-backed membrane: The Indian musical drum
Authors: Bhat R. B.
Published: October 2005

The acoustics of a cavity-backed composite membrane is analyzed under the assumptions that the cavity air pressure on the membrane is uniform, and hence sustains only axisymmetric membrane modes under impact excitation of the membrane. Axisymmetric modes of the membrane possess a harmonic relationship among themselves under certain parameter combinations. This mathematical model is chosen to explain the presence of harmonic overtones that are present in the Indian musical drum Mridanga, when struck with a characteristic stroke. Experimentally measured sound spectra from Mridanga when it is impact excited with different stroke patterns are also provided.
Ultrasonic beam method for localized weak debonding in a layered aluminum plate
Authors: Felsen Leopold B. , Zeroug Smaine
Published: October 2005

Weakened bonds in layered materials cannot easily be detected by vertically probing ultrasonic beams because the flaw is insensitive to the predominantly compressional (P) wave input. By tilting the incident P beam, which is taken to be Gaussian in this model study, it is possible to generate tangential shear components parallel to the bond line and thereby induce scattering from the flawed region. This scenario is explored here for a two-layer aluminum plate with a weak localized debond. A two-dimensional obliquely incident Gaussian P beam, modeled rigorously by the complex source point technique, is tracked through multiple reflections with P–SV coupling. Its interaction with the flaw is treated in the Born approximation that renders the induced source distribution along the flaw equal to the unperturbed field multiplied by the weak debonding profile, taken here as quasi-Gaussian. The resulting scattered field is computed by spectral wave-number synthesis, and the displacements of the plate surface generated by the incident and scattered fields furnish the desired data. The accuracy of the computation is confirmed by comparison with an independent solution developed previously3,4 by modal summation. Examination of the data reveals a weak, but separately identifiable contribution due to the scattering if the incident beam interacts directly with the flaw without having undergone reflection. Because all relevant phenomena are dominated by quasi-Gaussian wave processes, a simple Gaussian beam tracking and scattering model is proposed and found to be capable of explaining the relevant features in the data. This beam modeling of the problem is then applied to determine from the data the location, extent, and strength of the Gaussian debonding profile. Both the forward and inverse algorithms are structured around physically transparent wave processes and can easily be implemented.
Calculation of the Green’s function for the wedge-shaped layer
Authors: Ansbro A. P. , Arnold J. M.
Published: October 2005

The Green’s function for a two-dimensional planar wedge environment is derived in the spectral domain and using the Poisson summation formula the solution becomes rapidly convergent. This Green’s function field is calculated for two separate source positions to demonstrate its global modelike properties. A boundary-value approach is formulated to model mode launching into the wedge structure using the spectral Green’s function allowing direct comparison with the parabolic equation method and the beam propagation method.
Iceberg active sonar acoustics at 36 kHz
Authors: de Heering Ph.
Published: October 2005

The paper presents results from active sonar trials of the detection of icebergs (both in free water and ice locked) at 36 kHz. It is observed that the average target strength of the bergs, can be approximately expressed as a function of their estimated mass by a simple expression having physical plausibility. Further, observations of the echo duration, maximum detection range, sonar self-noise and under ice reverberation are presented and discussed.
Directional modal scattering by a ridge
Authors: Fawcett John A.
Published: October 2005

The three-dimensional scattering of modes from a ridge is considered. A simple plane-wave interpretation is demonstrated for the conversion of an incident mode into transmitted/reflected modes by interaction with a compact ridge. There are, however, particular angles of incidence for which the transmission/reflection coefficients of modal conversion exhibit singular-type behavior. This results in a banded structure in the horizontal two-dimensional plots of scattered modal power. Both ‘‘small’’ and ‘‘large’’ ridges are considered and a boundary integral equation method is used to compute the modal scattering from these ridges numerically.
Scattering matrix and boundary integral equation methods for acoustic propagation over a slope
Authors: Dawson Trevor W.
Published: October 2005

Boundary integral equation methods are combined with a scattering matrix approach to provide an accurate model for acoustic propagation over an arbitrarily shaped boundary deformation joining two uniform waveguide sections of dissimilar depth. The method is illustrated by up- and down-slope propagation over an idealized shallow water shelf model.
Remote sensing of ocean flows by spatial filtering of acoustic scintillations: Observations
Authors: Farmer D. M. , Crawford G. B.
Published: October 2005

The concept of spatial aperture filtering is tested with a linear array of four sources and four hydrophones on opposite sides of a tidal channel. The system is used to develop time series of the spatially filtered signal at different locations in the channel, from which flow speed may be derived. Results are in reasonable accord with independent current measurements, and reproduce the observed time-dependent horizontal shear of the flow. A concept of synthetic spatial aperture filtering is introduced whereby additional transmitter-receiver paths are synthesized through appropriate time delays so as to derive new information on the flow field.
A fast algorithm for the reconstruction of a layered acoustic medium from its transmission response
Authors: Yagle Andrew E.
Published: October 2005

We derive a layer-recursive algorithm for reconstructing an equal-travel-time layered acoustic medium from its transmission response to an impulsive plane wave incident from below. The medium is bounded below by an infinite half-space from which it is probed, and it is bounded above by a free surface at which its transmission response is measured. This problem arises in the nondestructive evaluation of layered composite materials, in which the transmission response is measured. The algorithm is similar in form to the Levinson algorithm, which reconstructs a layered medium from its reflection response; it is closely related to the lattice algorithm for computing the QR factorization of a matrix. We relate the new algorithm to the Levinson, fast Cholesky, and lattice algorithms, all of which are well-known fast algorithms for signal processing.
Postlingual deaf speech and the role of audition in speech production: Comments on Waldstein’s paper [R. S. Waldstein, J. Acoust. Soc. Am. 8
8, 2099–2114 (1990)]
Authors: Sapir Shimon , Canter Gerald J.
Published: October 2005

Using acoustic analysis techniques, Waldstein [J. Acoust. Soc. Am. 88, 2099–2114 (1990)] reported abnormal speech findings in postlingual deaf speakers. She interpreted her findings to suggest that auditory feedback is important in motor speech control. However, it is argued here that Waldstein’s interpretation may be unwarranted without addressing the possibility of neurologic deficits (e.g., dysarthria) as confounding (or even primary) causes of the abnormal speech in her subjects.
Comparison of numerical calculations of two Biot coefficients with analytical solutions
Authors: Yavari B. , Bedford A.
Published: October 2005

The Biot theory for the acoustics of porous media contains drag and virtual mass coefficients that depend on the physical properties of the fluid and solid constituents, the frequency, and the microstructure of the porous medium. Biot derived an equation for the drag coefficient as a function of frequency by assuming cylindrical pores. In this paper, the finite element method is used to obtain values of the drag and virtual mass coefficients for face-centered cubic granular materials with three different porosities and compare our numerical results to Biot’s analytical solutions. By making appropriate choices of three parameters in Biot’s analytical solution for cylindrical pores—the pore size parameter, the Kozeny parameter, and the tortuosity—the analytical solution matches these numerical results very well. This suggests that with appropriate choices of these parameters the analytical approach can predict the dependence of the drag and virtual mass coefficients on frequency for an arbitrary pore geometry. These results support a relation suggested by Hovem and Ingram for approximating the pore-size parameter for spherical grains, and they also agree with a relation suggested by Berryman for the dependence of the tortuosity on the porosity.
Precision method for phase match of microphones
Authors: Wong George S. K.
Published: October 2005

This paper describes a precision method for phase match of microphones. The coupler method consists of an acoustical driver that excites two test microphones simultaneously in a small cavity. The phase response contribution of the associated preamplifiers and amplifiers is eliminated with an interchange procedure, which enables the relative phase of a pair of microphones to be assessed accurately. The uncertainty of the method is dependent upon the uncertainty of the phase meter or phase measuring system. Over a frequency range from 63 Hz to 2 kHz and with a commercial phase meter that has an uncertainty of ±0.02 deg, the uncertainty of this coupler method for phase comparison is approximately ±0.05 deg. The coupler is useful over a frequency range from 10 Hz to 6.3 kHz.
The effect of digitally recorded and/or mixed records on power spectra and turntable velocity
Authors: Stepanian Christopher J. , Mead Philip
Published: October 2005

This study had a twofold purpose: to determine if music recorded or processed digitally retains a sampling frequency artifact on vinyl records, and to find whether the playing of such records has an effect on the turntable speed. A sharp peak in the record’s acoustic power spectrum was noted on some recordings at 44.1 kHz, an agreed upon standard sampling frequency for digital recordings. No unexplainable differences in the rotational speed between the test ‘‘digital’’ records and the control ‘‘analog’’ records were observed during the study.
Dual-probe laser interferometer
Authors: Huang Jin , Achenbach Jan D.
Published: October 2005

A single-probe heterodyne laser interferometer has been modified to a dual-probe modality by utilizing the reference beam as a second probe. It is shown that the two probing beams display the same sensitivity, independent of the difference in reflectivities at the two points of detection on the specimen. The dual-probe interferometer allows measurements of the same surface wave signal at two points. The instrument is particularly suited for the measurement of the speed and attenuation of surface waves. The interferometer has been employed to determine phase velocities and attenuation coefficients for surface wave propagation over an aluminum plate with increasing degrees of surface roughness.
Laser detection of sound
Authors: Vignola Joseph F. , Berthelot Yves H. , Jarzynski Jacek
Published: October 2005

A differential laser Doppler velocimeter (LDV) has been assembled and tested to provide noninvasive absolute measurements of acoustic particle displacements of standing waves generated in a water-filled tube. The principle of the technique [see K. J. Taylor, J. Acoust. Soc. Am. 59, 691–694 (1976)] is to measure the Doppler shift of laser light scattered from colloidal microparticles oscillating under the action of an acoustic field. The system tested is capable of detecting particle displacements of the order of a few nanometers with a bandwidth of several kilohertz. The performances and limitations of the system are discussed. In particular, the effect of Brownian motion is shown to produce only negligible broadening of the spectral density of the signal of interest. The sensitivity of the present LDV system is estimated to be very close to the shot noise limit of the photomultiplier tube used to detect the Doppler shift of the scattered light. Experimental results are obtained under controlled laboratory situations with calibrated microparticles of radii 10, 1, 0.1, and 0.01 μm in suspension in very pure (reverse osmosis) water.
Shear-wave detection of structural effects in aqueous solutions of bovine serum albumin and hemoglobin
Authors: Barnes C. , Lewis T. J.
Published: October 2005

Using cylindrical quartz crystal torsional resonators operating at 39 and 75 kHz to generate shear waves in aqueous solutions of the proteins bovine serum albumin and hemoglobin and the polypeptide poly l-glutamic acid, it has been possible to determine the complex dynamic shear viscosities of the solutions. The effects of concentration, pH, and denaturation using various agents have been studied. It is possible to relate the viscosity and configurational elasticity of the solutions, to the intramolecular and intermolecular forces associated with the proteins. Various effects that had been found earlier in compressional wave absorption studies of the proteins at frequencies between 60 and 400 kHz and attributed to conformational changes of bovine serum albumin and the quaternary doublet interactions of hemoglobin have been confirmed and emphasized by the use of shear waves.
Listening bandwidths and frequency uncertainty in pure-tone signal detection
Authors: Schlauch Robert S. , Hafter Ervin R.
Published: October 2005

The effect of frequency uncertainty on the detection of tonal signals in noise was studied using a modified probe-signal method.Widths of the listening bands used during detection were measured directly, allowing for an analysis that separates the effects of having to monitor multiple independent bands from those due to limited frequency resolution. Uncertainty was varied by beginning each trial with a cue consisting of one, two, or four randomly chosen, simultaneously presented tones. An expected signal, whose frequency matched one of the components in a cue, was presented on a majority of trails. However, on remaining trails, the signal was a probe, which meant that its frequency differed from one of the components in the cue by a constant ratio. Performance as measured in percent correct declined for probes at increasingly distant ratios from the expected values. The results were converted to dB using individual psychometric functions for expected signals and listening bands were fitted using the rounded exponential filter of Patterson et al. [J. Acoust. Soc. Am. 72, 1788–1803 (1982)]. The obtained bandwidths are comparable to those reported using notched-noise maskers, but there is a small but consistent increase in bandwidth with increased numbers of components in the cues. The primary result is that the effects due to uncertainty are well described by a 1-of-M orthogonal band model, which takes into consideration limitations of the detector, including the widths of the listening bands.
Human brain-stem auditory evoked responses obtained by cross correlation to trains of clicks, noise bursts, and tone bursts
Authors: Burkard Robert
Published: October 2005

Brain-stem auditory evoked responses (BAERs) were obtained in eight normal-hearing young adults. Stimuli included clicks, noise bursts, and tone bursts. Tone bursts included carrier frequencies of 1, 2, 4, and 8 kHz. All stimuli were presented at 60 dB nHL. BAERs were obtained by presenting stimuli in pseudorandom trains, called maximum length sequences (MLSs). BAERs were recovered by cross correlating the responses with a recovery sequence. MLS–BAERs were obtained with minimum pulse intervals (MPIs) of 6, 4, and 2 ms. Conventional BAERs were also obtained for stimuli presented at a rate of 30 Hz. BAERs were obtained for all stimuli, for both the conventional averaging technique and for the cross-correlation technique. BAERs were observed for MPIs as short as 2 ms for all stimuli. Wave V was the only peak consistently identifiable for these stimuli. For all stimuli, wave V latency increased and wave V amplitude decreased with decreasing MPI. This is the first demonstration of the use of maximum length sequences combined with cross correlation to obtain BAERs to noise burst and tone burst stimuli.
Detection and localization of narrow-band transient signals using the Wigner distribution
Authors: Rao Preeti , Taylor Fred J.
Published: October 2005

The Wigner distribution (WD) is applied to the detection and localization in time, of narrow-band transient signals of unknown waveform in an additive background comprised of quasiharmonic and random components. A traditional method of processing such signals is the spectrogram or short-time power spectrum. Based on an important relationship between the spectrogram and the WD, the application of the WD to the problem is investigated. The smoothed-pseudo-WD is shown to provide superior time and frequency resolutions of the received signal. Based on monitoring the received power in localized regions of the WD spectrum, a detector that achieves good time localization of the transient waveform and is relatively insensitive to changes in signal duration and frequency is obtained. The signal-to-noise ratio of the proposed detection statistic is derived for a transient signal in additive Gaussian noise and its dependence on various signal and detector parameters discussed.
Sound transmission through concrete blocks with attached drywall
Authors: Warnock A. C. C.
Published: October 2005

The furring system for attaching drywall to concrete block walls must be chosen with care if sound transmission is not to be increased at low frequencies. Once drywall is attached to a normal-weight block wall, sound transmission loss is increased above a frequency fx given by fx=K/(Md)1/2 . (M is the surface mass of the drywall, kg/m2, and d is the depth of the cavity behind it, m. K is 108 for an empty cavity and 60 for a cavity filled with fibrous sound-absorbing material.) Sound transmission loss data measured for a number of 190-mm normal-weight concrete block wall systems are presented to support the above result. Walls were measured with drywall systems mounted on different furring types on one side and on both sides. The cavities behind the drywall were either empty or filled with glass fiber. A simple, limited prediction scheme that can be used to recreate the data is described. The measurements in this paper present a consistent picture of the influence of the depth of the air space between the drywall and the block, and the presence of sound-absorbing material in the cavity.
Effect of time compression and expansion on the discrimination of tonal patterns
Authors: Sorkin Robert D. , Montgomery DeMaris A.
Published: October 2005

This experiment tested how well human listeners can discriminate between temporal patterns that are compressed or expanded in time. The listener’s task was to determine whether two arrhythmic, tonal sequences had the same or different temporal patterns. According to the pattern correlation model [R. D. Sorkin, J. Acoust. Soc. Am. 87, 1695–1701 (1990)], listeners perform this task by computing the correlation between the pattern of time intervals marked by the tones in each sequence. Listener performance dropped when one of the sequences was compressed or expanded in time. In order for the model to describe the observed performance, it was necessary to postulate an internal noise component that was proportional to the magnitude of the difference between the sequence transformations.
Piezoelectric strain rate gages
Authors: Lee C. -K. , O’Sullivan T. C.
Published: October 2005

Piezoelectric strain rate gages have been designed using linear piezoelectric theory and relatively simple circuitry that can be used to measure an average strain rate at a point of a structure. By combining the effective surface electrode, appropriate skew angle and the correct polarization profile, a uniaxial strain rate gage that measures only the strain rate along a specified direction and a pure shear strain rate gage that measures the in-plane shear strain rate are developed. Experimental as well as theoretical results are presented. Various types of generalized piezoelectric strain rate gages are also introduced and discussed.
Acoustic resonance frequency shifts in a spherical cavity with an eccentric inner small sphere
Authors: Roumeliotis John A. , Kanellopoulos John D. , Fikioris John G.
Published: October 2005

Analytical expressions are derived for the acoustic resonance frequency shifts in a hard-walled spherical cavity, caused by the introduction of an eccentric small inner sphere. A straight-forward and simple approach is employed to obtain a first-order perturbation of the cavity wave function and the corresponding resonant frequency shift. The results are useful in a variety of problems connected with acoustic levitation and with excitation or probing of resonant cavities. Graphical results for some of the lower-order resonant frequencies are given for various values of the parameters.
Use of translational addition theorems for spherical wave functions in nonaxisymmetric acoustic field problems
Authors: Lease Steven A. , Thompson William
Published: October 2005

The translational addition theorems for spherical wave functions are used to develop solutions for nonaxisymmetric acoustic field problems involving nonconcentric spherical boundaries. The forms of the addition theorems are discussed along with their use for translating modal expansions of spherical wave functions centered on one origin to another. In a typical problem involving a number of arbitrarily positioned spherical sources within a spherical-shaped region, an infinite set of equations for the modal expansion coefficients is developed. This set must be truncated and then solved numerically. Assuming lossless media, comparison of the near-field and far-field-radiated acoustic power provides a test of the accuracy of the numerical results. Other quantities of interest such as the radiation impedance of each source, including the mutual effects of all the other sources, and directivity patterns can be readily computed once a sufficient number of modal coefficients have been computed. In a companion paper [S. A. Lease and W. Thompson, Jr., J. Acoust. Soc. Am. 90, xxx–xxx (1991)] numerical results are presented for the specific case of two identical monopole sources within a fluid sphere that is embedded in another infinite fluid medium.
Nonaxisymmetric acoustic radiation from a pair of sources embedded in a fluid sphere
Authors: Lease Steven A. , Thompson William
Published: October 2005

The general analysis of nonaxisymmetric acoustic radiation in a spherical coordinate system, contained in a companion paper [S. A. Lease and W. Thompson, Jr., J. Acoust. Soc. Am. 90, xxx–xxx (1991)], is specialized to the case of two, identical, in-phase monopole sources similarly positioned within a fluid sphere. This special case allows the truncated system of equations to be reduced to a set involving (N+1)2 unknowns. Special considerations in the development of a fortran program to solve the system of equations are discussed. Numerical results for the radiated acoustic power, directivity patterns, and source radiation impedance for this specific case can be calculated by the program. Some numerical results are presented for a few specific choices of the many physical parameters that characterize the problem. The performance of the program for the various choices of physical parameters is discussed.
Normal hearing threshold levels in the low-frequency range determined by an insert earphone
Authors: Arlinger Stig D.
Published: October 2005

Hearing threshold levels have been determined in the low-frequency range (20–500 Hz) on a group of 30 young normal-hearing subjects using monaural stimulus presentation through an insert earphone (Etymotic Research ER-3A). A retest was performed on half of the group to provide data on test–retest reliability. The mean hearing threshold levels obtained agree closely with the Minimum Audible Field data of ISO 226, however, with some deviation at the very lowest frequencies below 40 Hz. The test–retest difference results yielded mean values that averaged 1.15 dB with an average standard deviation across test frequencies of 3.9 dB. The results show that low-frequency hearing thresholds for pure tones of frequencies from 40 Hz and up can be determined with acceptable validity and reliability by the use of this type of insert earphone.
Presbycusis and noise-induced permanent threshold shift
Authors: Macrae J. H.
Published: October 2005

Bies and Hansen [J. Acoust. Soc. Am. 88, 2743–2754 (1990)] have proposed an alternative formulation of the relationship between noise exposure and noise-induced hearing impairment to that presented in International Standard ISO 1999, in which they assume that presbycusis and noise-induced permanent threshold shift (NIPTS) are additive on an antilogarithm basis. Data concerning deterioration in hearing threshold levels at 4000 Hz due to aging in war veterans with NIPTS do not support the Bies and Hansen assumption but provide support for the formula for combining presbycusis and NIPTS incorporated in International Standard ISO 1999.