bluuebunny/crossref_metadata_2025 huggingface dataset:
High-frequency reflection and scattering by multicomponent rough surface distributions
Authors: Lucas R. J. , Twersky V.
Published: October 2005

Earlier forms for coherent reflection and incoherent scattering by mixtures of different bosses on rigid or free base planes [V. Twersky, J. Acoust. Soc. Am. 29, 209–225 (1957)] are applied to recent results for triaxial ellipsoids [R. J. Lucas and V. Twersky, J. Acoust. Soc. Am. Suppl. 1 81, S20 (1987)] to investigate continuous distributions. Key variables d (e.g., keel depth) are governed by two-parameter probability density functions f(m, u:d) with m as the mean value of d, and v as the normalized variance whose value (zero to unity) determines the curve of f. For ᴜ small, f is Gaussian (and reduces to a delta function as ᴜ approaches zero to reproduce one-component results); more generally, the curve is skewed, and as v approaches unity, f reduces to the exponential for the simplest Poisson case. Numerical integrations over finite ranges of d provide illustrative graphs to exhibit effects of truncation and parameters on the reflection coefficients and differential scattering cross sections per unit area. [Work supported by ONR.]
Application of Twersky's boss scattering theory to laboratory measurements of sound scattered by a rough surface
Authors: Chu D. , Stanton T. K.
Published: October 2005

One of the many challenges in describing the scattering of sound by rough surfaces is to address the fact that most surfaces are three dimensional. Furthermore, only their statistical properties may be known. Twersky's model [V. Twersky, J. Acoust. Soc. Am. 29, 209–225 (1957)], by using an image method and taking multiple scattering into account, established a general solution involving distributions of three-dimensional bosses. A prominent advantage of the theory over others is that it can describe bosses of arbitrary orientation. The purpose of the present paper is to compare the theory with laboratory measurements of sound scattered by a continuously rough pressure-release surface. In the model, we use prolate spheroidal bosses oriented in a range of directions and randomly packed to approximate the rough surface. The data were taken from Welton [P. J. Welton, report ARL-TR-75-30, University of Texas at Austin (1975)] and involve backscattered sound versus grazing angle for several frequencies. The comparisons of the numerical computations with Welton's data show a reasonable agreement with respect to not only the boss orientation parameters but also the surface statistical parameters (rms roughness and correlation length). An interesting phenomenon is that the number of spheroidal modes required to fit the data increases as the frequency of the incident plane wave increases.
Diffraction of harmonic acoustic radiation by the apex of a ridge
Authors: Buckingham Michael J.
Published: October 2005

In general, there are three components in the acoustic field propagating over a ridge: an arrival via a direct path, which does not interact with the ridge, a component diffracted by the apex of the ridge, and a component reflected from the boundary of the ridge. All three contributions are included in a new, exact solution of the Helmholtz equation for the three-dimensional, harmonic field from a point source over a ridge formed by two intersecting, pressure-release planes. The solution is obtained by the application of three integral transforms to the Helmholtz equation, followed by the corresponding inverse transforms. It is valid for any ridge angle θ0, including the special case of a knife edge (θ0 = 360°) and a plane (θ0 = 180°). In the latter case, which is the Lloyd's mirror problem, there is no diffraction; only the direct and reflected components contribute to the field, which is highly variable because of strong spatial interference. Diffraction by the knife edge is exemplified by the leakage of energy over the barrier, and by the presence of spatial interference in regions where there is no reflected component. Similar phenomena appear with other ridge geometries, where it is usually possible to assess whether interference effects are due predominantly to diffraction or reflection.
Investigation of 12-kHz-deep scattering layers observed with the multibeam echo-sounder Sea Beam
Authors: de Moustier Christian , Alexandrou Dimitri
Published: October 2005

During a survey over Horizon Guyot in March 1986, acoustic measurements in the deep scattering layers were carried out with the 12-kHz multibeam echo-sounder Sea Beam installed on the R.V. ATLANTIS II. Acoustic returns received by each of the Sea Beam's 16 preformed beams were sampled in quadrature at 1 kHz from the surface down to about 1400-m depth, and recorded digitally with a special purpose data acquisition system originally developed at the Marine Physical Laboratory for seafloor acoustic backscatter measurements. These complex volume reverberation data are presented as horizontal and vertical cross sections showing the 3-D structure of the scattering layers observed. Statistical analyses of the data involve homogeneity and stationarity tests on a beam to beam level as well as on a single beam along the ship's track to establish the degree of patchiness and the corresponding reverberation levels associated with the layers observed. [Work supported by ONR.]
Further investigation on concert hall acoustics
Authors: Ando Yoichi
Published: October 2005

Since the book [Y. Ando, Concert Hall Acoustics (Springer, New York, 1985)] was published, further work has been carried out. The results demonstrate that the running autocorrelation function of source signals may better describe preferred conditions of sound fields than does the long-time autocorrelation function. The minimum value of the effective duration of the running autocorrelation function of music signals for the interval used for the subjective preference judgment corresponds well to the preferred initial time delay gap between the direct sound and the first reflection when the amplitude A1 = 0 dB. This was discovered when Japanese music containing a so-called “MA,” a stationary sound passage without any rapid movement, was used as a source signal. Also, it was found that a flat frequency characteristic of the subsequent reverberation time is the optimal condition, and that it may be calculated by the minimum effective duration of the running autocorrelation function of the source signal. Several other results will be presented.
Early reactions to Yoichi Ando's book, Concert Hall Acoustics
Authors: Beranek Leo L.
Published: October 2005

Ando's book joins one of a handful written since 1900 on the science of concert hall acoustics. It contains seven chapters packed with physical acoustics, the behavior of the physical hearing system, the response of the nervous system, psychoacoustic judgments, the composition of a sound field in a hall, prediction of subjective preferences in halls based on the sound field, and design studies for concert halls. This paper attempts to relate Ando's teachings to the practical experience the author has had with the acoustics of a number of concert halls. Areas in which the book fits these examples well and in which further research seems indicated will be presented.
Solved and unsolved problems in concert hall acoustics
Authors: Schroeder Manfred R.
Published: October 2005

During the last 2 decades, several difficult problems in concert hall acoustics have been solved or elucidated. Among these, the recognition of the importance of early lateral sound ranks high. This piece of progress was made possible by the combined efforts of experts from widely varying disciplines (physicists, psychologists, mathematicians, and—last but not least—the observant consultant). But much remains to be done to strengthen the science of room acoustics further. With digital electronics reaching new pinnacles of sophistication, active sound field modification should be pursued much more actively—for multipurpose halls and other spaces. Diffusion of radiated sound can be controlled by the same number-theoretic principles that are finding increasing use in passive designs [M. R. Schroeder, Number Theory in Science and Communication (Springer, New York, 1986), 2nd ed.]. But the optimum degree of diffusion requires further subjective studies. Generally usable software for digital modeling of acoustic spaces should be developed to exploit more fully the capabilities of modern computers (both serial and parallel). Room acoustical measurement methods that use speech and music as test signals, to permit measurements during actual performances, should be perfected and used as a matter of course. The future of room acoustics is bright if we grasp the available opportunities.
Ray solution for finite amplitude two-dimensional waves in a hard-walled rectangular waveguide
Authors: Shu Kun-tien , Ginsberg J. H.
Published: October 2005

This paper describes a ray superposition theory for cumulative growth of nonlinear effects in a two-dimensional acoustic mode, based on decomposition of the mode into a pair of obliquely propagating, nonlinear planar waves. The mathematical foundation of the formulation is an earlier perturbation analysis of the reflection of a distorted planar wave obliquely incident on the boundary of an infinite half-space [Z. Qian, Sci. Sin. 25, 492–501 (1982)]. Based on the results of the analysis, each of the pair of rays forming the signal at a selected field point is traced back to its origin at the excitation. Each ray is described as a simple planar wave undergoing finite amplitude distortion that depends on the propagation distance along that ray between field and source points. This distance is the same for each ray at a specified field point, but differences in the excitation at the respective source points result in phase differences between the two rays. The overall signal is shown to be the same as a modal description of the propagation [H. C. Miao and J. H. Ginsberg, J. Acoust. Soc. Am. 80, 911–920 (1986)]. The ray solution explains an apparent paradox in the modal analysis, which indicated that although the signal can be resolved into a pair of planar waves, the distortion process is scaled only by the axial position along the waveguide. Conversely, the earlier solution provides validation for the superposition of rays, as well as for the linear reflection law. [Work supported by NSF and ONR.]
Sound field modeling and simulation of a concert hall
Authors: Hidaka T. , Kageyama K. , Masuda S.
Published: October 2005

A sound field simulator has been developed in order to subjectively evaluate hall acoustics in the laboratory. This process consists of three parts: (1) a computer program for deriving the impulse response of the room from the architectural drawings using the ray tracing technique; the program is written mainly in FORTRAN and the calculation is implemented using a maximum of 1500 plane surfaces and 1 000 000 rays at most; (2) a new source of orchestral music that is digitally recorded in a temporarily constructed anechoic room for this purpose, and (3) hardware to synthesize the calculated sound field, which convolutes impulse response with input signal by digital signal processing and outputs from 24 speakers set up in an anechoic room in real time. In order to apply this simulator, subjective experience by means of audition for comparison of six concert halls is made and its ability and accuracy are investigated.
Speech perception by budgerigars (Melopsittacus undulatus): Synthetic VOT stimuli
Authors: Dooling Robert J. , Okanoya Kazuo , Brown Susan D.
Published: October 2005

Budgerigars (parakeets) were trained using operant conditioning procedures to detect changes in a repeating background of synthetic speech sounds. Response latencies for detection were used to construct similarity matrices, and multidimensional scaling procedures were then used to produce spatial maps of these speech stimuli reflecting perceptual organization. Birds were tested on the same alveolar, velar, and bilabial synthetic VOT stimuli previously used to test humans and chinchillas [P. K. Kuhl and J. D. Miller, J. Acoust. Soc. Am. 63, 905–917 (1978)]. Results indicate that budgerigars partition each of these continua into two perceptual categories with an abrupt perceptual change occurring at roughly the same VOT location as observed in humans and chinchillas. It is concluded that the perceptual mechanisms underlying the categorigal perception of VOT stimuli are sufficiently general so as to be common to the mammalian and avian auditory systems. [Work supported by NIH.]
A code in the nodes
Authors: Streeter Lynn A. , Kamm Candace A. , Kane-Esrig Yana
Published: October 2005

Four two-layer associative networks (one layer of hidden nodes) were trained to classify spectra extracted from naturally produced vowels of one talker, using the hack-propagation learning algorithm [Rumelhart et al., Nature 323, 533–536 (1986)]. The inputs to the networks were either LPC spectra or “perceptual” spectra, obtained by convolving the Bark-transformed spectrum with an asymmetric filter. The networks trained on LPC input classified test vowels more accurately than the networks trained on the perceptually based spectra. The networks' solutions were examined to determine (a) whether it was possible to characterize the features underlying the solutions and (b) whether the input spectral representation affected the nature of the solutions. Principal components analyses of the activation patterns of the hidden units showed that three dimensions accounted for over 80% of the variance for all networks. For networks trained on “perceptual” spectra, the first three dimensions were highly correlated with F2/F1 ratio, F3/F2 ratio, and F3, respectively. For LPC input, the first three dimensions correlated with F1, F3/F2 ratio, and F2/F1 ratio. Thus these networks classified the vowels using features related to the formant space.
Complete gradient optimization of a recurrent network applied to /b/, /d/, /g/ discrimination
Authors: Watrous Raymond L. , Ladendorf Bruce , Kuhn Gary
Published: October 2005

A complete gradient optimization technique for connectionist networks with recurrent links is presented. A truncated gradient technique is available in the literature. A network with recurrent links for discrimination of /b/, /d/, and /g/ in the context of following /i/, /a/, or /u/ is designed. The performance of this network as optimized with either of the two techniques is then compared. It will be shown that the complete gradient is exact and simpler, and it will be demonstrated that it leads to superior performance.
Speech categorization using recurrent networks
Authors: Anderson Sven , Merrill John , Port Robert
Published: October 2005

Several connectionist networks were trained to classify the English syllables ba, da, ga, pa, ta, ka collected from two male and two female speakers. Using a speech preprocessor, perceptually based spectral patterns were computed [H. Hermansky, Proc. ICASSP 87, 1159–1162 (1987)] every 5 ms. A sequential network having a limited class of recurrent connections [M. Jordan, ICS Tech. Rep. University of California at San Diego (1986)) was employed to categorize the data. Training by back propagation or second-order back propagation, a linear increase in the certainty of classification over the course of the syllable was required. Performance of the sequential networks was evaluated on both “known” and “unknown” speakers. When tested on novel tokens of a known speaker, the sequential network did very well as oppsed to very poorly on tokens from an unknown speaker. Sequential networks trained with back propagation are capable of integrating cues distributed over time and using them to categorize data. However, their learned behavior may not generalize to data from other speakers. [Work supported by NSF.]
Recognition of continuously spoken letters by listeners and spectrogram readers
Authors: Daly Nancy A. , Zue Victor W.
Published: October 2005

Because of acoustic similarities between some letters of the alphabet, automatic recognition of continuously spoken letters is a difficult task. The goal of this study is to determine and compare how well listeners and spectrogram readers can recognize continuously spoken letter strings from multiple speakers. The interest in spectrogram reading results is motivated by the belief that this procedure may help to identify acoustic attributes and decision strategies that are useful for system implementation. Listening and spectrogram reading tests involving eight listeners and six spectrogram readers, respectively, were conducted using a corpus of 1000 wordlike strings designed to minimize the use of lexical knowledge. Results show that listeners' performance was better than readers' (98.4% vs 91.0%). In both experiments, string lengths were determined very accurately (98.1% and 96.2%), presumably due to the large number of glottal stops inserted at letter boundaries to facilitate segmentation. Most of the errors were due to substitution of one letter for another (68% and 92%), and they generally fall into two categories. Asymmetric errors can often be attributed to subjects' disregard for contextual influence, whereas symmetric errors are largely due to acoustic similarities between certain letter pairs. Subsequent acoustic study of four of the most confusable letter pairs has resulted in the identification of a number of distinguishing acoustic attributes. Using these attributes, overall recognition performance better than that of the readers was achieved. [Work supported by NSF and DARPA under contract N00014-82-K-0727, monitored through the Office of Naval Research.]
Age-related changes in auditory brain-stem responses of Mongolian gerbil
Authors: Mills John H. , Schmiedt Richard A. , Adams Joe C. , Schulte Bradley A. , Kulish Lawrence R.
Published: October 2005

As part of a large study of age-related hearing loss, the Mongolian gerbil as an animal model is evaluated. Part of this effort involves monitoring electrical potentials recorded from the auditory nerve and brain stem. The gerbils are born and raised in an acoustically treated quarters where the median sound level is 35 dBA. Test signals are 1.8-ms tone pips ranging from 1–16 kHz in octave steps. By age 22–24 months, auditory sensitivity is decreased by about 10 dB at most frequencies although there are large differences between animals. By age 36 months, mean threshold changes are largest at 8 and 16 kHz, about 35 dB, and decrease to 10–20 dB at lower frequencies. The mean data for the 36-month group form an audiometric configuration that is qualitatively similar to that observed for 60- to 65-year-old human males and 70-year-old human females. Individual differences are large with some animals having nearly normal hearing and others having hearing losses of 50–70 dB at all test frequencies. [Work supported by NINCDS PO NS-25039.]
Intraoperative variability of auditory brain-stem responses during cranial nerve microvascular decompression
Authors: Hannley Maureen , Huang Jer-Min , Raudzens Peter A.
Published: October 2005

The auditory brain-stem response (ABR) recordings of 40 patients who underwent microvascular decompression of cranial nerves V and VII were analyzed. The latencies of waves I, V, and the I-V interpeak interval were examined as a function of subject age, subject gender, operative site, and operative stage. The results indicated that: (1) subject age was not a significant factor in intraoperative ABR variability; (2) significant male-female latency differences were present only in the earlier operative stages; (3) greater latency changes were produced during decompression of the facial nerve than during trigeminal nerve decompression; and (4) greater response variability was present during the later operative stages. The patients' postoperative auditory status was studied to determine the predictive efficiency of three currently used criteria for significant intraoperative latency change.
Evolution of unstable disturbances at a mean-flow stagnation point
Authors: Thompson Charles , Manley Martin
Published: October 2005

The interaction between infrasonic, acoustic disturbance and the mean flow near the stagnation point of a bluff body will be examined. The criterion for stability and intermodulation of the disturbances will be discussed. The spatial and temporal evolution will also be addressed. It will be shown that infrasonic and unstable vortical disturbances, working in conjunction, can serve to modulate the amplitude of the acoustic wave. For a harmonically time-varying acoustic wave, this modulation is manifested in an increase in the spectral bandwidth of the wave. [Work supported by Analog Devices Professorship.]
Photoacoustic method for measuring high-order multiphoton absorption in transparent materials
Authors: Jones Scott C.
Published: October 2005

A photoacoustic technique for measuring multiphoton absorption from sharply focused pulsed laser beams is developed for arbitrary order of nonlinear absorption. The resulting sound source is neither spherical nor cylindrical, and a multipole expansion is presented through quadrupole terms for thermal and electrostrictive source terms. The analysis shows that the acoustic amplitude is proportional to absorbed energy, and proportional to the mth power of incident laser energy under conditions of m-photon absorption. The calibrated measurement of the four-photon absorption cross section of NaCl at wavelength 532 nm is presented. [Work supported by NSF.]
Propagation of sound in vibrationally excited N2/He, N2/H2, N2/CH4, and N2/H2O mixtures
Authors: Shields F. Douglas , Lafleur L. Dwynn
Published: October 2005

Previous measurements of resonant reverberation of sound in N2/H2 mixtures in a closed tube have shown an amplification of the sound following the rapid excitation of the gas by an electric discharge [J. Acoust. Soc. Am. 81, 87 (1987)]. These measurements have been extended in this paper to include N2/He, N2/CH4, and N2/H2O mixtures. Some of the excessively large amplification previously observed in N2/H2 mixtures has now been attributed to nonlinearity in the microphone response. However, even after correcting this error the amplification is much larger and lasts for a longer time than theoretically predicted. The variation in the translational temperature following the discharge is monitored by measuring the sound velocity. Relaxation times determined from the temperature versus time curves differ only slightly in magnitude and/or temperature dependence from previously reported values. [Work supported by ONR.]
Instability and particle agglomeration in acoustically driven flows
Authors: Thompson Charles , Mehta Vineet
Published: October 2005

The mechanism responsible for instability and turbulence generated by high amplitude acoustic waves has been of interest of in acoustic agglomeration of particles. When a low-frequency oscillation is used in conjunction with a higher frequency acoustic wave, localized unstable boundary layer disturbances can be used to drive the outer flow field. By virtue of the Reynolds stress, the harmonic spectrum of the unstable vortical disturbance serves to modulate the acoustic wave amplitude. This process enhances the ability of the acoustic wave to agglomerate particles. A theoretical model of the instability mechanism and the interaction process will be presented. [Work supported by Analog Devices Professorship.]
Pulse scattering from an object submerged in the ocean
Authors: Collins Michael D. , Werby Michael F.
Published: October 2005

The low grazing angle asymptotic limit, which is used to derive the parabolic equation, is used to derive an efficient computational method for propagation and scattering of acoustic pulses in the ocean. In order to avoid the time-consuming frequency decomposition, all calculations involving the acoustic field are done in the time domain. The time domain equivalent of the parabolic equation [B. E. McDonald and W. A. Kuperman, J. Acoust. Soc. Am. 81, 1406–1417 (1987)] is used for propagation of the incident field up to the scatterer and for propagation of the scattered field away from the scatterer. Waterman's extended boundary condition, which is a frequency domain method, is used to construct a time domain integral operator that transforms incident waveforms into scattered waveforms and depends only on the geometry and composition of the scatterer. Like the parabolic equation method, this method offers an attractive combination of accuracy and efficiency. [Work supported by ONR and NORDA.]
Weak foci in three-dimensional linear shock waves and the stability of real shock waves
Authors: Marston P. L.
Published: October 2005

A first approximation for identifying foci in weak shocks is to consider the focusing process for a linear discontinuity. This is sometimes known as the “acoustic shock” approximation. A case studied previously is the longitudinal cusp of “arête” located where three rays merge at the focus of a converging cylindrical wave; for finite amplitudes, the wave can become unstable with respect to the formation of a pair of shock-shocks in the focal region [B. Sturtevant and V. Kulkarny, J. Fluid Mech. 73, 651–671 (1976)]. In the present research, catastrophe theory is used to identify wave-front shapes W(x,y) that propagate to produce novel acoustic foci with caustics spread out roughly transverse to the direction of propagation [P. L. Marston, J. Acoust. Soc. Am. 81, 226–232 (1987) and Proceedings of the APS 1987 Topical Conference on Shock Waves in Condensed Matter (in press)]. The initial displacement of the wave front from the xy plane is given by W(x,y). Linear waves with W = − (a1x2 + a2y2x + a3y2), where the shape parameter a2≠0, propagate to produce transverse cusps described by a cubic cusp curve. A stronger focus, the hyperbolic umbilic, is produced if W = − [(αx3 + 3γy2x)/6 + a1(x2 + y2)] with a1⩽0. A focal section is produced at a distance z = − (2a1)−1 from the xy plane. In this section, caustic lines form a V with an apex angle of 2 arctan (γ/α)1/2 Issues related to shock stability are noted. [Work supported by ONR.]
Calculation of the intensity and absorption of a finite-amplitude sound wave
Authors: Blackstock David T.
Published: October 2005

An operational definition of absorption α = − ∇-I/2I, where I is the sound intensity, was popular in the 1950s as a means of quantifying the increased attenuation of finite-amplitude ultrasonic beams. Note here that α is the total absorption of the wave, not just the absorption of the fundamental component. Although the relation fell into disuse in the 1960s and 1970s, Carstensen et al. [Acustica 51, 116–123 (1982)] recently revived it as a natural and useful tool to characterize attenuation in biological media. For a wave sinusoidal at the source, Carstensen et al. calculated α and I from the Fourier series expression for the pressure of a finite-amplitude wave. Shown in this paper is an alternative calculation based on the time-domain version of the weak shock solution. Exact, closed-form expressions for I and α are given for all distances from the source. The extension to pulses is indicated. [Work supported by NIH and ONR.]
The effect of an object in a waveguide on the field produced by a point source
Authors: Norton Guy V. , Werby M. F.
Published: October 2005

When the acoustic wave produced by a point source in a shallow water waveguide ensonifies an object, the object acts as a secondary source and in turn generates a guided wave. The effect that the object has on the initial incident field is estimated. In particular, comparisons are made with transmission loss versus range for the point source by itself and with transmission loss versus range for the object as ensonified by the point source. To obtain the scattered field from the object, the scattered field is first generated in the vicinity of the object using a transition matrix that relates the incident field to the scattered field in a waveguide. The transition matrix is obtained from the extended boundary condition (EBC) method. This solution is coupled with the waveguide solution [G. V. Norton and M. F. Werby, “Some Numerical Approaches to Describe Acoustical Scattering from Objects in a Waveguide,” in Proc. Sixth Int. Conf. Math. Modeling, St. Louis, MO, 4-7 Aug. 1987]. This method satisfies all appropriate boundary conditions and yields a continuous solution throughout the waveguide. The object used is a rigid spheroid of aspect ratio 5:1. The frequencies are 100 and 450 Hz. The source, receiver, and object depths are different combinations of 50, 75, and 200 m, and two waveguide water depths at 150 and 400 m are considered. The nature of the effect on the initial incident field due to the object and as influenced by the relative placement of the source and receiver relative to it is demonstrated. [Work supported by NORDA.]
Fuzzwall: Blocking sound transmission above suspended ceilings
Authors: Quirt J. D. , Halliwell R. E.
Published: October 2005

Interoffice partitions commonly stop at the suspended ceiling. The plenum space above the ceiling is used as an air-return for the ventilation system. Sound transmission through this space often limits interoffice speech privacy. This paper reports the performance of a barrier system intended to reduce sound transmission without preventing airflow. Fibrous absorptive material has been used as a barrier system. Dependence of the noise reduction on physical parameters (flow resistance, barrier thickness, density, etc.) and the effect of providing for airflow have been examined. Laboratory results and preliminary field test data will be presented.
The measurement of acoustic reflection coefficients by using cepstral techniques
Authors: Bolton J. Stuart
Published: October 2005

The author has recently developed a new free field technique for the in situ measurement of acoustic reflection coefficients by the use of transient acoustic signals [J. S. Bolton and E. Gold, J. Sound Vib. 110(2), 179–202 (1986)]. The measurement is conducted by using an omni-directional loudspeaker to generate a test signal; a microphone placed between the loudspeaker and the reflecting surface is used to detect both the incident (i.e., the direct) and reflected signals. Cepstral processing was used to eliminate some of the difficulties associated with existing reflection measurement methods. Specifically, in the new technique the direct and reflected signals are measured simultaneously and are allowed to overlap in time, thus relaxing geometrical constraints. Further, if the incident signal is sufficiently broadband, a knowledge of its spectrum is not required nor is it necessary to measure it individually. The technique is based on the direct extraction of the impulse response of the reflecting surface from the power cepstrum of a signal containing both the direct and reflected signals. Full-scale acoustical trials of this technique will be described and measurements of the reflection coefficients of samples of polyurethane foam will be shown.
A boundary integral equation method for transient problems in architectural acoustics
Authors: Holger D. K.
Published: October 2005

A numerical method for solving transient interior acoustic field problems by combining boundary integral equation and inverse Fourier transform methods is described. An indirect boundary integral equation formulation of the Helmholtz equation using hybrid layer potentials is combined with a numerical inverse Fourier transform to obtain time domain solutions for acoustic field variables. Frequency domain representation of transient sources, frequency domain windowing of the integral equation solutions, and frequency domain resolution enhancement are discussed. Numerical results are presented for the acoustic field inside a rectangular prism with various boundary conditions and for various transient source excitations.
Scattering of sound by finite cylinders—A preliminary study applicable to microphones
Authors: Seybert A. F. , Lin M. L. , Wu T. W.
Published: October 2005

When the wavelength of sound is of the same order as the diameter of a microphone, the diffraction of sound in the nearfield of the microphone will alter the indicated sound-pressure amplitude. This fact has long been observed experimentally, and the effect of diffraction is considered in the design of microphones. However, when a pair of microphones is used, as with sound intensity measurements, diffraction alters the amplitude and phase difference indicated by the microphones, and these effects become important at frequencies within the normal operating frequency range of a single microphone of the same diameter. It has been difficult to predict diffraction effects from analytical models, due to the complex geometry involved. A numerical method has been developed, based on the boundary element method, to study the scattering of sound by obstacles of arbitrary shape. As a preliminary model of a microphone, the finite cylinder was used. In this paper numerical results are presented for scattering from a single finite cylinder and from a a pair of cylinders. [Work supported in part by the NSF.]
Differences in measured microphone responses at different distances from artificial mouth
Authors: Kruger Frederick M. , Mayer Mitchell , Bareham John , Kruger Barbara
Published: October 2005

When calibrated and operated at different distances, available artificial voice/mouth frequency response measuring systems yield significantly different results. These differences are greater for different distances than for different voice/mouths. Virtually all reported voice measurements are made at the CCITT recommended distance of 25 mm from the lips, or 25 mm in front of the “lip ring” of the artificial voice/mouths. Yet, all high-noise environment military microphones are tested 1/4 in. (6 mm) from the voice/mouth opening. This is, however, several millimeters behind the lip plane of the voices. At this distance, noise canceling microphones show minimal cancellation effects. Historically, the use of 1/4-in. distance began with a laboratory artificial voice/driver sound system, which predated the first commercially available, relatively large scale production artificial voice (B&K 4216). This paper shows the feet(s) on measured microphone response when military noise canceling microphones (M-138, H-250, M-162) and a calibrated laboratory microphone are tested, on axis, at different distances from several calibrated voices (6 mm from opening, 6 and 25 mm in front of lip ring). Real world implications are discussed.
Normative data and physical measurements for a nonsense CVC intelligibility test
Authors: Rankovic Christine M. , Van Tasell Dianne J.
Published: October 2005

A nonsense CVC stimulus set has been developed for investigations into the application of the articulation index (AI) to speech recognition testing with hearing-impaired persons. Each test list consists of 57 items drawn at random from a corpus of 228 items comprising syllables spoken by four talkers. Crossover functions support the similarity of this stimulus set to that of French and Steinberg [J. Acoust. Soc. Am. 19, 90 (1947)] in terms of frequency importance function. Because the number of vowels was restricted to three, the long-term spectrum of the stimulus set is less smooth than that usually associated with connected speech. The AI versus percent-correct phoneme score functions have been obtained from normally hearing subjects. Comparisons with published data for similar stimulus sets will be made. Talker differences, reliability of subject performance, and preliminary data from hearing-impaired subjects will be reported. [Work supported by NICHHD T32 HD-07151 and NINCDS NS 12125.]
Relations between the intelligibility of sentences and words for subjects with normal hearing and with presbycusis
Authors: Smoorenburg Guido F. , Bosman Arjan
Published: October 2005

In order to see whether hearing-impaired listeners make more extensive use of nonacoustic cues than listeners with normal hearing, different speech materials were presented to three groups of subjects. The speech material consisted of meaningful and meaningless words of the consonant-vowel-consonant type (CVC words) and short, simple sentences of eight to nine syllables [R. Plomp and A. M. Mimpen, Audiology 18, 43-52 (1979)]. All material was uttered by a male and a female speaker. Three groups of subjects participated in the experiments: young normal-hearing subjects, elderly subjects with good hearing regarding their age, and subjects with presbycusis. Test-retest reliability for both word and sentence material was found to be close to the theoretical value. Averaged across all groups of subjects, the correlation between the SRT for word and the SRT for sentence material was 0.97, which resulted in a prediction error of 5.3 dB. Prediction of the SRT's from the tone audiogram with the Fletcher index (average hearing loss at 500, 1000, and 2000 Hz) resulted in a prediction error of 5.5 and 6.8 dB for the SRT for word and sentence material, respectively.
Acoustic characteristics of speech produced by an older geriatric population
Authors: Liss Julie M. , Weismer Gary , Rosenbek John C.
Published: October 2005

The effect of advancing age on speech production characteristics is not clearly understood. Yet it is crucial for those who study and serve the normal and neurologically impaired geriatric populations. The purpose of this study was to obtain a normative database for the speech production characteristics of an older geriatric group. Fourteen veterans, 87 to 93 years old, served as subjects. A total of 40 sentences were produce at a conversational rate by each gentleman. Wideband (300-Hz) spectrograms were created from high-quality tape recordings. Specified acoustic measures were made (consonant, vowel, and voice onset time durations, and vowel formant trajectories) via a hit-pad digitizer and microcomputer sonogram analyzer program. Descriptive analysis was performed to obtain normal parameters of speech production. Current theories of aging and neurological diseases affecting speech will be discussed.
Acoustic properties of isolated stop consonant release transients
Authors: Repp Bruno H. , Lin Hwei-Bing
Published: October 2005

This study focuses on the initial component of the stop consonant release burst, the release transient. In theory, the transient, because of its impulselike source, should contain much information about the vocal tract configuration at release, but it is usually weak in intensity and difficult to isolate from the accompanying frication in natural speech. For this investigation, a human talker produced isolated release transients of /b,d,g/ in nine vocalic contexts by “mouthing” these syllables. He also produced the corresponding CV syllables with regular phonation for comparison. Spectral analyses showed the isolated transients to have a clearly defined formant structure, which was not seen in natural release bursts whose spectrum was dominated by the frication noise. The formant frequencies varied systematically with both consonant place of articulation and vocalic context, with the latter, “coarticulatory” influence actually exceeding the former. The transient thus provides an acoustic snapshot of the vocal tract immediately after the release. [Work supported by NICHD.]
Some phonetic characteristics of sentences produced in noise
Authors: Bond Z. S. , Moore Thomas J. , McCreight Kate
Published: October 2005

In previous work, relatively systematic differences were found in the phonetic structure of spondee words produced in isolation and in the presence of noise. The purpose of this study was to determine whether noise affects continuous speech in a similar manner. Four speakers were recorded reading 20 sentences, two times in quiet and two times while hearing 95 dB SPL pink noise. The sentences were digitized, segmented, and transcribed using SPIRE. The resulting database consisted of approximately 850 segments per subject per speaking condition. The distributions of spectral and temporal properties of classes of segments were determined using SEARCH. All subjects produced sentences in the presence of noise with increases in fundamental frequency and total energy, as has been found for isolated words. Segment durations and spectral characteristics were affected by noise for some subjects more than for others. [Work supported in part by AFOSR.]
A neuropsychological model of voice perception
Authors: Van Lancker Diana , Kreiman Jody
Published: October 2005

It will be argued that two neuropsychological processes mediate voice perception: pattern recognition and featural analysis. Properties of voices that determine the relative dominance of these two kinds of processing include: (1) the familiarity of the voices, and (2) the heterogeneity of the voice set—how “similar” the voices are to each other. Familiar voices that differ substantially are processed largely as complex patterns, while a homogeneous set of unfamiliar voices recruits some featural analysis. This model is derived from the following observations: (1) Acoustically altering familiar voices affects their recognizability unevenly, suggesting that no fixed set of parameters underlies the recognition process; (2) performance of dichotic listening and brain-damaged subjects indicates that the right hemisphere is specialized for recognizing familiar voices (the right hemisphere is believed to be superior in pattern recognition); (3) in paired comparison tasks, normal subjects compare discrete auditory features, as well as the overall pattern; (4) brain-damaged subjects who show impaired performance in unfamiliar voice discrimination have damage to either the right or left temporal lobe (specialized for processing of auditory detail). Testable hypotheses generated by this model will also be discussed.
Speech rate and gender as mediating factors in person perception
Authors: Dohm Faith-Anne , Feldstein Stanley , Crown Cynthia L.
Published: October 2005

A previous study by the authors found that judgment of speech rate is independently influenced by the gender of the judges and of the speakers. The research of others [e.g., B. Brown et al., J. Acoust. Soc. Am. 54, 29–35 (1973)] has shown that speech-rate perception mediates other aspects of interpersonal perception. The study reported here, then, investigated the possibilities that: (a) Individuals who speak rapidly are viewed in more positive ways than those who speak slowly; (b) males are viewed in more positive ways than females; and/or (c) speech and gender jointly influence person perception. Seventeen male and 28 female college students listened to and rated, each of ten adjective scales, three male and three female speech samples. An hierarchical multiple regression analysis indicated that, whereas the male speakers received less positive ratings when they spoke rapidly, the female speakers received less positive ratings both when they spoke rapidly and slowly. The results suggest that person perception is influenced jointly by the speech rate and gender of a speaker.
Spectral shaping and the intelligibility of speech in noise
Authors: Pedlow R.
Published: October 2005

Various studies have shown that attenuation of the first formant results in enhanced speech intelligibility in noise. One explanation (Nye et al., 1974) suggests that it is caused by upward spread of masking in normal speech perception. The other explanation proposed is that attenuating F1 increases the amount of intelligibility information relative to the total speech energy in the signal. The stimulus materials were phonetically balanced (PB) words produced by one male speaker. Stimuli were digitally filtered with a high-pass filter that had a passband from 1100–5000 Hz and a stopband from 0–850 Hz (the attenuation levels of the stopband were 10, 20, 40, and 80 dB relative to the unfiltered signal. Items were equated for rms energy after filtering. Processed stimuli were presented to subjects in a perceptual identification task at 0 dB S/N. Articulation index values were then computed for the test items. Fitting transfer functions to the data showed that the increased intelligibility obtained by high-pass filtering was not well predicted by the articulation index values. This supports the hypothesis that attenuating the F1 results in an increase in intelligibility due to a release from masking effect.
Perceptual evidence for stress foot structure
Authors: Edwards Jan , Beckman Mary
Published: October 2005

A year ago, Beckman and Edwards [J. Acoust. Soc. Am. Suppl. 1 81, S67 (1987)] reported on two durational effects. Phrase-final lengthening is a phonologic mark of intonational phrase boundaries: It is a substantial effect whose domain is highly consistent across speakers. Seeming word-final lengthening is a much smaller and more variable effect—how to describe the domain of this smaller effect, or even whether it should be described phonologically, was unknown. The perceptual salience of these two effects has been examined, reasoning that if they are phonologic marks of prosodic constituent boundaries, then listeners should be able to perceive the edges so marked. Listeners can consistently use intonational phrase-final lengthening, but not the smaller durational effect, to discriminate between phrases that are otherwise identical. It is suggested that the smaller effect is not a phonologic mark of a prosodic constituent within the intonational phrase. Rather, it is a result of articulatory timing relationships within the stress foot, with the longer [pa] in ‘pop opposed’ as compared to ‘poppa posed’ due to their different stress feet structures. [Work supported by NSF Grants. No. IRI-8617873 and IRI-8617852.]
Do segmental attributes distinguish spontaneous and prepared speech
Authors: Remez Robert E. , Sherman Anne , Klapwald Thea S. , Rubin Philip E.
Published: October 2005

A set of perceptual studies was conducted to assess the salience of segmental phonetic differences between spontaneous and prepared speech. Prior studies employing our perceptual paradigm revealed that naive listeners can differentiate lexically, syntactically, and thematically identical sentence pairs that have been extracted from brief monologs (> 40 s), identifying the spontaneously uttered version at high levels of performance. No single acoustic attribute appears to predict the perceptual salience of spontaneity in production, though the variability of f0 as the sentence ends and the average duration of syllables are both correlated with the detection of spontaneity. The present investigation exploits the perceptual differentiation test to determine the salience of segmental differences between spontaneous and prepared utterances. Against the benchmark of differentiability of intact sentence pairs, the perceptual distinctiveness of spontaneous and read utterances was evaluated by deleting the segmental information alone by low-pass filtering. In additional converging tests, versions of the sentence pairs without formant variation were synthesized, setting the spectrum to the vowel [ɑ], while preserving natural differences in duration and f0; and, synthetic sentence pairs lacking variation in f0 were tested in filtered and [ɑ] versions. The results will be discussed in reference to the perceptual constraints on the comprehension of spontaneous speech. [Research supported by grants from NINCDS and NICHHD.]