bluuebunny/crossref_metadata_2025 huggingface dataset:
Propagation of bottom interacting very low-frequency acoustic signals off the coast near Cape Fear, NC
Authors: Tunnell T. W. , Kramer A.
Published: October 2005

The Naval Ocean Research and Development Activity (NORDA) and the United States Geological Survey (USGS) at Woods Hole, MA, conducted the Cape Fear very-low-frequency (VLF, 20 Hz and less) exercise off the coast near Cape Fear, NC, during June 1985. The NORDA VLF array system (16-element vertical line array with a uniform spacing of 20 m) was deployed in 400 m of water in a bottom moored configuration. The data acquired by the array system were collected by the R/VG. W. PIERCE via a radio link with the telemetry buoy on the surface attached to the array. The USNS LYNCH towed a continuous wave (cw, 10 Hz) sound source at a depth of 91 m along tracks upslope and downslope from the array at ranges from 1–100 km. Slope angles along the tow tracks varied from 0.5°–2.0°. Signal-to-noise ratios of 10 dB and greater were observed on all array elements during much of the source tow. The mode structure of these signals are presented and compared with numerical predictions made by the parabolic equation and fast field program models. [Work supported by NORDA.]
Low-frequency sound attenuation in the Mediterranean Sea
Authors: Mellen R. H. , Akal T. , Hug E. H. , Browning D. G.
Published: October 2005

Sound-channel propagation measurements in the Ionian basin of the Mediterranean Sea have been analyzed to determine the attenuation coefficients in the frequency range of 50–3200 Hz. Concurrent measurements of sound-speed, temperature, salinity, and pH show a strong sound channel having a broad minimum below 100 m and highly uniform properties over the 600-km path. Explosive sources were detonated near the channel axis. The results obtained from hydrophones located near the axis are compared with predictions of the temperature/pH-dependent relaxation-absorption model (components: MgSO4, B(OH)3>, and MgCO3). Scattering loss is found to be minimal and agreement is good. Attenuation coefficients are compared with earlier values from the Ligurian Sea over a single refraction path approximately 35 km long.
Low-frequency propagation across the East Greenland Frontal Zone: Directional dependence of acoustic modes
Authors: Mellberg Leonard E. , Connors Donald N. , Johannessen Ola M. , Botseas George , Browning David
Published: October 2005

In a previous paper [L. E. Mellberg et al., J. Acoust. Soc. Am. Suppl. 1 77, S56 (1985)], an analysis was conducted to determine the relative importance of acoustic propagation modes (bottom bounce, surface duct, and convergence zone) for a shallow source (10 m) along a 100 nmi west to east track through the East Greenland Frontal Zone. In the present paper, a comparison is made to results obtained for a reciprocal east to west track through the zone to determine the directional dependence of the dominant acoustic modes. Along the west to east track, outward from the ice edge and the East Greenland Current, there is a transition from surface duct to convergence zone as the principal propagation mode. This transition, as well as the initial bottom slope, restricts the envelope of ray angles found in this convergence zone mode. For the east to west track, the convergence zone is the dominant mode along the entire path and its character changes significantly. The directional dependence of these modes result in marked changes in propagation loss with range and the energy fields as generated by the two sources. [Work supported by ONR and NUSC.]
The determination of geoacoustic models in shallow water
Authors: Frisk George V. , Lynch James F. , Doutt James A.
Published: October 2005

A technique for determining the geoacoustic models in shallow water is described. For a horizontally stratified ocean and bottom, the method consists of measuring the magnitude and phase versus range of the pressure field due to a cw point source and numerically Hankel transforming these data to obtain the depth-dependent Green's function versus horizontal wavenumber. In shallow water, the Green's function contains prominent peaks at horizontal wavenumbers corresponding to the eigenvalues for any trapped and virtual modes excited in the waveguide. From the Green's function, one can obtain the geoacoustic model via either forward modeling or perturbative inverse techniques. In the forward modeling approach, a geoacoustic model for the bottom is obtained by computing the theoretical Green's function for various values of the bottom parameters and determining the parameter set which provides the best agreement with the experimental Green's function, particularly in the positions and relative magnitudes of the modal peaks. In the perturbative inverse technique, one uses the differences between the measured modal peaks and those predicted by a background model as input data to an integral equation, which is solved for the bottom geoacoustic parameters. These techniques are demonstrated using experimental data at 140 and 220 Hz. [Work supported by ONR.]
Applications of a synthetic aperture array method for shallow-water geobottom reconnaissance
Authors: Tango G. J. , Werby M. F. , Wooten R.
Published: October 2005

A general approximate inverse technique for characterizing horizontally stratified shallow-water geobottoms has recently been presented [Kuperman et al., IEEE J. Ocean. Eng. OE-10(3) (1985); Frisk et al., J. Acoust. Soc. Am. 76 (1984)]. The basic method comprises measuring the pressure field from a cw point source and Hankel-transforming the field numerically to obtain the effective depth-dependent Green's impulse response versus horizontal wavenumber. Because the full wave pressure response P contains poles at horizontal wavenumbers corresponding to eigenvalues of all discrete, continuous, and evanescent modes, G can be readily obtained from the empirical transfer function, as derived from a synthetic aperature towed array simulation. Since the inverse Hankel transform simply relates P(r) and G(k), the peaks of the true theoretical Green's function in horizontal wavenumber can be seen to be directly proportional to the discrete angles of maximum beamformer response. Frisk et al. have successfully inverted synthetic array Green's function data via least-squares forward modeling of major features via the FFP. Alternatively, an approximate geoacoustic model for the bottom can also be obtained from the modal response angles over near- and farfields. In the specific case of relatively few modes, the highest order of which is near cutoff, locating the discrete/continuous wavenumber demarcation gives the critical angle for the shallowest bottom layer. More generally, selective beamforming over near- and farfields allows modal discrimination from progressively deeper layers for successively increasing source-array offsets. Representative results for accuracy and resolution are given for the cases of single and multilayered isovelocity and gradient-structured geobottoms.
Extraction of the shear modulus profile within a sandy seabed by a passive remote bottom shear modulus profiler
Authors: Yamamoto Tokuo
Published: October 2005

We have developed the software of a bottom shear modulus profiler (BSMP). The BSMP software is a stable and high-resolution inversion procedure for extraction of the bottom shear modulus profile from measured motion of the bed surface induced by propagating gravity water waves. Measurements of the motion of a sandy bed and the wave-induced bottom pressure in the water depth of 7.5 m have been made at sea. The shear modulus profile within the sand bed has been extracted from these data using the BSMP inversion procedure. The extracted shear modulus profile agrees well with direct measurements. The paper reports on the experimentation and the inverse analysis of the data. [Work supported by ONR, Code 425UA.]
Function theoretic methods for acoustic wave propagation in the ocean
Authors: Gilbert Robert P. , Woods David H.
Published: October 2005

In this approach to modeling the sound propagation in the ocean, function theoretic methods are used in partnership with numerical methods. The problem of computing a sound field in the ocean is greatly simplified by using a two-pronged approach involving (1) function theoretic methods that analytically incorporate the major features of the ocean and its bounding surfaces, including the radiation condition and the average index of refraction, and (2) numerical methods to model the effects of the secondary but detailed features of the ocean and its boundaries. For example, to model a sound field in a uniform ocean with perfectly reflecting boundaries over a long distance is a difficult numerical problem but a trivial analytical problem. In our two-pronged approach, numerical techniques are used only to compute the changes that are observed as a result of perturbing the ocean from its analytical idealization. [This work was supported by Code 10 and Code 3332 of the Naval Underwater Systems Center.]
Analyses of large arrays: Brief theory and some techniques used in 1954–1985
Authors: Martin Gordon E.
Published: October 2005

The behavior of underwater acoustic arrays of sources and receivers is reviewed with the historical perspective of some applications during the last 30 years. The general basis for interelement interactions of acoustic, mechanical, and electrical components is described. Array analyses require the solution of large matrix equations. This is done with large computer assets (not available 20–30 years ago), or special symmetry and special algorithms. The mathematical matrix form for several types of array systems is shown including particularly those with Toeplitz symmetry. The author's early (1954) formulation of an array problem with very adverse interactions including elements with negative radiation resistances is given as a simple illustration for a small array. Modular and nonmodular electrical drives have different matrix forms. The techniques for computation of the performance of arrays with many elements including the Trench algorithm (1960's) are described. See abstracts of the author's papers given at prior meetings. Results for large arrays include damage due to adverse interactions, discovery and development of velocity control techniques, and difficulties due to electrical steering of directional response. Slides include the 1955 nearfield calibration of a planar source array with 200 elements. Some results are given for an array of more than 1000 elements, having block Toeplitz form with Toeplitz blocks.
Tang influence on handbell mode
Authors: Hansen Uwe J. , Hampton Scott , Rossing Thomas D.
Published: October 2005

Manufacturers routinely cast handbells with a stem at the crown, identified as the tang, facilitating the machining and tuning of the bell. For some models, portions of the tang are retained for handle and clapper fastening, while in others, attachments are made through a hole in the crown. The effect of the tang on the frequencies and amplitudes of musically significant modes is of interest to both handbell ringers and to scientists concerned with an understanding of the tonal structure of handbells. Modal mapping, using nearfield sound radiation and time-averaging holographic interferometry, was used to study G4, F6, and D7 handbells with and without tang. Tang removal lowered the frequency of the fundamental by approximately 1.3%. Holographically observed mode shapes for the musically significant, lowest modes (2,0; 3,0;, and 3,1) indicate very little vibrational encroachment into the crown area, suggesting that the crown and with it the tang, do not play a significant role in handbell timbre.
Digital music synthesizers as perceptually valid phenomenological models
Authors: Gibian G. L. , Mohamed M. Ben , Harnden E. N.
Published: October 2005

Commercially available digital music synthesizers can produce sounds which perceptually resemble many traditional acoustic instruments. As with all phenomenological modeling, the quality of the output (of a synthesizer, in this case) does not require the generation process to incorporate the physics of the modeled systems (which are acoustic musical instruments in this case). Spectrum-versus-time and spectrum-versus-pitch plots for several commercially manufactured digital synthesizers employing different synthesis techniques (including audio-rate frequency modulation) will be compared with their counterparts from traditional acoustic instruments. It is also of interest to compare the spectra after imposing transformations which model psychoacoustic processing, such as loudness curves and integration within critical bands.
A new method for the measurement of acoustic impedance and its application to musical instrument research
Authors: Bruneau Anne-Marie , Plitnik George R.
Published: October 2005

A method to accurately measure acoustic impedance over an extended frequency range without the limitations imposed by capillary tubes is described. The apparatus consists of two electrostatic microphones, one used as a source of excitation and one as receptor. By means of the equations of reciprocity, the frequency dependence of an unknown acoustic impedance may be accurately deduced. Experimental results and theoretical calculations will be compared for several well-known acoustic transmission line systems, and the results of utilizing this system to measure the imput impedance of a musical horn will be presented. The advantages of this system over capillary tube measurements for musical instrument research, will be discussed, and future applications will be mentioned.
Eigenfrequencies of elastic cylinders and shells; surface waves generated by normally and non-normally incident sound
Authors: Subrahmanyam J. V. , Überall H. , Nagl A.
Published: October 2005

We obtain the characteristic equation of an elastic cylinder, or cylindrical shell, immersed in a fluid and enclosing another fluid for vibrations containing both azimuthal and axial components. Numerical solutions are found for aluminum and steel cylinders, in the form of distinct “layers” of roots in the complex frequency plane, which may be interpreted as the resonances of azimuthal or helical surface waves around the shell. Rayleigh waves, Stoneley waves, and Lamb (whispering gallery) waves are identified and the interaction of Rayleigh and Stoneley branches at low frequency is clarified. Surface wave speeds and attenuations are obtained as functions of frequency. [Work supported by the Office of Naval Research.]
Experimental study of interface waves on a curved surface
Authors: Luppé F. , Quentin G. , Stoyanov A. , Überall H.
Published: October 2005

We present an experimental study of surface waves generated on a curve surface at grazing incidence. We have observed three types of circumferential waves travelling at a velocity close to that of water on the external surface of an elastic cylinder imbedded in water. Attention is focused on the attenuation and reemission of these waves in the surrounding fluid. We then study the influence of a discontinuity of the curvature on the propagation of these waves. [Work supported by the Direction des Recherches, Etudes et Techniques, France, and by the U.S. Office of Naval Research.]
Hardware implementations of real-time digital speech processing algorithms
Authors: Singer Elliot
Published: October 2005

Continuing advances in hardware technologies are permitting the realization of increasingly sophisticated speech processing algorithms in real-time equipments. The availability of commercial digital signal processing integrated circuit components has been especially responsible for a reduction in the size and cost of these devices. This presentation will describe the unique requirements of speech compression and speech recognition algorithms with respect to arithmetic calculations, memory, and I/O. Representative equipment designs developed at Lincoln Laboratory for realizing real-time speech processing algorithms will be described. These include: the Lincoln digital signal processor (LDSP), a programmable, general-purpose ECL machine suited for real-time evaluation of speech processing algorithms; the advanced linear predictive coding microprocessor (ALPCM), a flexible bit-slice processor designed for use in operational environments; and the compact linear predictive coder, a small, narrow-band vocoder terminal based on DSP microprocessors. The application of advanced VLSI technology to meet the processing demands of large vocabulary speech recognition will be discussed, with specific focus on an approach being pursued at Lincoln Laboratory which uses wafer scale integration and restructurable VLSI technology to exploit the high level of concurrency in the recognition algorithm. [Work sponsored by the Department of the Air Force.]
A dictionary-intensive letter-to-sound program
Authors: Coker Cecil H.
Published: October 2005

We treat the letter-to-sound problem as that of deriving all regularly inflected forms of English words from a dictionary of base forms. A computer program must perform all lexical operations a human might have to perform in using an abridged desk dictionary. Of course this includes addition of “s.” “es,” “d,” “ed,” “er,” and “ing,” with their accompanying sound changes, deletions of mate “e,” and changes from “y”to “i.” The program must also handle other forms. Stress-neutral endings like “less” and “ness” are quite simple. Four other types of endings, exemplified by “ation,” “ity,” “ian,” and “ency,” are more complex—reordering stress within the base word, and causing vowel shifts and consonant sound change. The problems with these endings are much like those for straight letter-to-sound rules, but with pronunciations of base form and ending as available information. With prefixes and compounds, the relative stress of constituent parts will depend on the intended part of speech for the target word, and on the properties of the parts. Final vowels of Greek prefixes may or may not reduce to schwa, depending on properties of the base forms they modify. With a programmed dictionary driver, we are currently finding ∼99.9% of words in the Brown corpus (excluding proper names and misspellings)—∼75% from exact dictionary entries, and the rest by derivation. We default to Mark Liberman's Name-Say program for words not found or derived.
Stress assignment in letter-to-sound rules for speech synthesis
Authors: Church Kenneth W.
Published: October 2005

This paper will describe a program that attempts to determine word stress from spelling. Four sources of constraints will be discussed: (1) syllable weight, (2) part of speech, (3) morphology, and (4) etymology. Stress assignment is a well-established weak point for many speech synthesizers because stress dependencies cannot be determined locally. It is impossible to determine the stress of a word by looking through a five- or six-character window, as many speech synthesizers do. Well-known examples such as degráde/dègradátion and télegraph/telégraphy demonstrate that stress dependencies can span over two and three syllables. In addition, examples such as calculi and tortoni demonstrate that stress is sensitive to etymology, a nonlocal property of the entire word. Calculi, like most other latinate forms, obeys the main stress rule of English, which assigns antipenultimate stress to nouns ending in two light syllables. In contrast, tortoni, like most other Italian and Japanese loan words, receives penultimate stress. Stressing Italian and Japanese surnames with the main stress rule of English as some commercial synthesizers do, leads to unacceptable results, as will be demonstrated. Our program produces much more acceptable results by estimating the etymology (by counting three letter tri-grams) and then adjusting the stress rules appropriately.
Text-to-speech synthesis of Japanese based on the Cepstrum-CV method
Authors: Ishikawa Yasushi , Nakajima Kunio
Published: October 2005

A text-to-speech synthesis system for Japanese is described. In synthesis of Japanese, generation of fundamental frequency pattern is one of the most important problems. We proposed the prosody rules that generate intonation pattern of any Japanese text using syntactic information. The prosody rules consist of modification rules, syntax rules, and a generation rule. Connecting several words, the system makes an accent phrase and gives it an accent using the dictionary and the rules (modification rules). The parameters used in a subsequent generation rule are modified on the basis of the syntactic structure of the input text (syntax rules). Then, a fundamental frequency pattern is generated from these parameters (generation rule). In this system, the synthesizer parameter is Cepstrum. Parameter rules assign segment duration and concatenate CV syllables Cepstra. Thus, this system translates any text of Japanese written with the syntactic symbols into good quality and natural speech.
Vowel errors in noise and in reverberation by hearing-impaired listeners
Authors: Nábělek Anna K. , Dagenais Paul A.
Published: October 2005

The effects of noise and reverberation on identification of monothongs and diphthongs were evaluated using ten subjects with moderate sensorineural hearing losses. Stimuli were 15 English vowels, spoken between /b/ and /t/ and recorded in a carrier sentence. The test was recorded without reverberation under quiet conditions then degraded either by recording in a room with reverberation time 1.2 s or by adding babble of 12 voices at speech-to-noise ratio of 0 dB. Both degradations caused statistically significant reductions of mean identification scores, but the difference between degraded means was not significant. The pattern of errors was different in noise and reverberation. Errors for monothongs in reverberation seemed to be related to an overestimation of vowel duration and to a tendency to perceive the pitch of the formant frequencies as being higher than without reverberation. Errors for monothongs in noise were not related to duration or pitch overestimation. Errors involving diphthongs were more frequent in reverberation than in noise. In reverberation, there was a tendency to judge a diphthong as the beginning monothong.
Relevance of time-varying properties of the first formant frequency in vowel representation
Authors: Di Benedetto Maria-Gabriella
Published: October 2005

Acoustic analysis of the vocalic portions of CVC nonsense syllables spoken by three speakers (two males and one female) in the sentence frame “The—again” shows that ambiguities between vowels, for each speaker, occur if the vowels are represented by the values of F1 and F2 sampled at the time where F1 reaches its maximum. These ambiguities occur primarily in the F1 dimension. The examination of the F1 trajectories of the vowels for which confusion occurs, shows variations in the initial slope of this trajectory among different vowels. In particular, if two different vowels such as /ɪ/ and /ε/ have the same maximum F1, then F1 for the lower vowel reaches its maximum value earlier. Perceptual experiments have been carried out to examine the perceptual importance of F1 slope, using synthetic CV syllables. Preliminary results are in agreement with the hypothesis that stimuli with a steeper initial slope of F1 are perceived as lower vowels. Results are similar for subjects of different languages, leading to a suggestion that this phenomenon can be explained on an auditory basis.
Vowel perception in noise
Authors: Besing J. , Hurtig R. R. , Collins M. I.
Published: October 2005

Vowel confusion data presented by Pickett [J. Pickett, J. Acoust. Soc. Am. 29, 613–620 (1957)] were reanalyzed using a multidimensional scaling procedure. The data were analyzed using this method to determine the relative contribution of each dimension under different noise and context conditions. The original data were collected using vowels produced in phonetically balanced words and bVb syllables under different noise spectra conditions. Data were fit with a three-dimensional solution; two dimensions were comparable to F1 and F2. The subject weights provided estimates of the relative salience of each dimension for each noise condition. Examination of those weights indicated that the relative salience of each dimension varied with vowel context. Dimensions one and three were more heavily weighted for bVb contexts. Dimensions two and three were weighted more heavily for phonetically balanced contexts. Differences in weighting of each dimension yielded perceptual representations that were distortions of the traditional representation of vowel space in terms of F1 and F2.
Relationship between LP-residual spectral distances and phonetic judgments
Authors: Kamm C. , Kahn D.
Published: October 2005

The relationship between LP-residual spectral distances [F. Itakura, IEEE Trans. Acoust. Speech Signal Process. ASSP-23, 67–72(1975)] and phonetic judgments was examined in two studies. One analysis compared published phonetic similarity results [D. Klatt, Proc. ICASSP, 1278–1281 (1982)] with Itakura distances between steady-state synthetic reference vowels (/æ/ and /a/) and a set of acoustic variants of each vowel (approximating Klatt's stimulus sets). Despite the documented utility of the Itakura measure for speech recognition, correlations between the phonetic similarity measures and Itakura distances were very low (0.20 and 0.02 for /æ/ and /a/, respectively). Significant differences in rank across the two measures were observed for acoustic variants incorporating spectral tilt, low-pass filtering and changes in F3. A second experiment measured Itakura distances and phonemic identification judgments for 21 acoustic variants of each often English reference vowels. For six reference vowels, at least one variant was identified as phonemically distinct from its reference: these phonemically distinct variants yielded smaller Itakura distances than 48% of those variants judged not phonemically distinct from the reference. These results quantify and corroborate the well-known lack of optimality of the Itakura distance measure for speech recognition and may contribute to the design of more appropriate distance metrics.
Acoustic and perceptual correlates of prevocalic pharyngeal and uvular consonants in Arabic
Authors: Alwan Abeer
Published: October 2005

The first part of this study investigated the acoustic correlates of two pharyngeal consonants /ʕ,ħ/ and three uvular consonants /ʁ,ɦ,ǫ/in prevocalic position with the three vowels /a/, /i/, and /u/, in Arabic. Analysis of these consonants showed distinctive formant trajectories for each class of sounds and the existence of several allophonic realizations for the voiced consonants in both classes. The second part of the study examined the perceptual correlates of the two voiced consonants /ʕ/ and /ʁ/ prevocalically with the vowel /a/. Formant trajectories and bandwidths were manipulated independently in synthetic nonsense /Ca/ syllables. These synthetic stimuli were then presented to subjects in identification tests. Results show that a high F1 and the proximity of F1 and F2 are essential for the perception of /ʕ/, whereas a widened F1 bandwidth is essential for the perception of a natural /ʁ/. The acoustic and perceptual results are discussed in terms of the mechanisms of production of sounds with a narrow constriction between the glottis and the velum, and the corresponding articulatory-acoustic transformations involved. [Work supported in part by an NSF grant.]
The use of microcomputers in an undergraduate acoustics laboratory
Authors: Korman Murray S.
Published: October 2005

Microcomputers are used in the laboratory as a tool to collect, store, analyze, and graph data from acoustics experiments. Emphasis is placed on teaching the physics involved from individual experiments and not on particular features of the computer. These microstations allow students to record data from the experimental apparatus using the computer. Simple menued software instructions are available to organize the collection of data for printout or for X-Y pairs to be plotted and labeled. Students can also scale their data and store or plot these variations. This laboratory facility speeds up the tedious and routine data taking chores of the experiment and frees up time for discussion of the physics. In many cases time allows the student a chance to try a few things on his own. Simple acoustics experiments will be shown involving transient vibration and Fourier analysis using our mini-computer station. This station has a 6502 microprocessor, a two-channel analog to digital plug-in board, a disk drive, monitor, and printer. Use is made of an inexpensive FFT program. Currently, there are four stations available in the acoustics course taught for physics and engineering majors. [Work supported by USNA.]
Sound speed in air: Past and present
Authors: Wong George S. K.
Published: October 2005

The calculation of a new value for the sound speed (C0 = 331 ⋅ 29 ms−1) in standard dry air at 0°C and at a barometric pressure of 101.325 kPa is presented. The maximum uncertainty in this sound speed C0 is estimated to be approximately 200 ppm. The theory of the above calculation is based on the equation of state, and with the knowledge of γ/M which is derived from published theoretical and experimental thermodynamic data on the constituents of the standard atmosphere [G. S. K. Wong and T. F. W. Embleton, J. Acoust. Soc. Am. 76, 555–559 (1984)], and M is the molar mass. Investigations which led to the general acceptance of the previous sound speed are examined, and there is strong evidence that enables one to conclude that the maximum possible uncertainties in previous sound-speed assessments are sufficient to encompass the above new sound speed. The effects of carbon dioxide on sound speed are discussed.
Ray, mode, and hybrid options for time-dependent source-excited vibrational motion in an elastic plate
Authors: Lu L T. , Felsen L. B.
Published: October 2005

A hybrid theory for source-excited propagation in multiwave multilayer media is developed whereby ray fields and normal mode fields (with a smoothing remainder) are combined in self-consistent proportions so as to take advantage of the favorable features of each of these descriptions [I. T. Lu, L. B. Felsen, and A. H. Kamel, Wave Motion 6, 435–457 (1984)]. To avoid proliferation of multiple reflected ray fields, caused by wave coupling at boundaries, a new spectral object called “eigenray” has been introduced, which has dispersive characteristics similar to those of a normal mode but undergoes reverberations like a single ray field in a single wave medium. For time-harmonic excitation by a line forcing function, these new formulations have already been tested numerically on the simple but nontrivial example of P-SV propagation in a single elastic plate [I. T. Lu and L. B. Felsen, I. Acoust. Soc. Am. (to be published)]. The validity of the hybrid algorithm has been confirmed, and parameter regimes have been found wherein the hybrid approach offers a competitive alternative to other options. The test calculation is extended to excitation by a high-frequency Gaussian pulse transient source. The results reveal the complicated multiple arrival structure, well resolved at early times but tending toward modal behavior at later times. These features, which pertain especially to long observation intervals and wave constituents with high-frequency spectral content, are explained well, and are computed efficiently, by the “optimal” hybrid format. [Work supported by NSF and ONR.]
Normal mode propagation in a rough-walled waveguide
Authors: D'Spain Gerald L. , Reitzel Ken J. , Medwin Herman
Published: October 2005

The effects of a randomly rough boundary on normal mode propagation in a rigid-walled waveguide have been studied theoretically and experimentally. The theoretical solution considers steep-sloped roughness elements and uses the low-frequency technique of Tolstoy [J. Acoust. Soc. Am. 75, 1–22 (1984)]. It is predicted that, at frequencies below the first mode cutoff, the mode generated in the rough-walled waveguide has a greater amplitude at the rough boundary than at the smooth boundary and propagates with a constant phase plane wavefront at a slower speed than in the smooth-walled waveguide. The laboratory experiment, using gravel and frequencies up to 10 kHz, shows excellent agreement with the theoretical prediction. The eigenfunction perturbations and phase velocity changes caused by steep-sloped roughness elements are discussed for higher frequencies and higher order modes. [Research supported by the Office of Naval Research.]
Strain sensing using interface waves in clad rod acoustic waveguides
Authors: Claus Richard O. , Bennett Kimberly D.
Published: October 2005

The measurement of strain in materials using clad rod acoustic waveguides has been considered recently by several authors [R. T. Harrold and Z. N. Sanjana, Proc. Soc. Plastics Eng. Conf., Washington, DC (May 1985)]. In such waveguides, axisymmetric torsional, axisymmetric radial-longitudinal, and core-guided shear modes typically propagate if the velocity of plane shear waves in the clad exceeds that velocity the core. If instead the materials are reversed, these modes are not supported but an interface wave may propagate on the core-clad boundary. In principle, such waves may propagate without attenuation if the core and clad materials properties are suitably related. The theoretical sensitivity of the attenuation and mode conversion of such nonattenuating interface waves to strain-induced variations in the geometry and elastic constants of the waveguide materials are considered in this paper. Experimental measurements using strained glass-on-glass clad rods which approximately satisfy the assumed boundary conditions are reported. Potential applications in the internal monitoring of materials are suggested. [Work supported by NASA and Simmonds Precision.]
Vibration response of ring-supported, fluid-loaded, circular cylindrical shells to turbulent boundary layer pressure fluctuations
Authors: Bostian David A. , Burroughs Courtney B.
Published: October 2005

Using the analytic model presented at the 109th Meeting of the Acoustical Society of America at Austin, TX in April 1985, predictions of the vibration response of ring-supported, fluid-loaded, circular cylindrical shells to turbulent boundary layer pressure fluctuations are presented. At different frequencies, wavenumber spectra of the shell response are presented and wavenumber bands where the shell response is maximum are identified. Examples of the dependence of the shell response on the observation location, and ring size and spacing are also given. Sums over wavenumbers are taken to obtain the response of the shell at a point as a function of frequency.
Determining the spatial decay rate of free waves in layered plates
Authors: Jameson Paul W. , Chandiramani Khushi
Published: October 2005

We will show that the spatial decay rate of free bending waves in multilayered fluid-loaded plates can be estimated very easily using only a knowledge of the input admittance of the plate as a function of the frequency ω and wavenumber k of the normal stress applied to the plate. If the admittance is written in the form of a magnitude and a phase θ, the spatial decay rate ασ for free waves is given approximately by the expression ασ(ω)≈ − 2⧸∂(ω,k)∂k|θ = 0. The evaluation of the partial derivative of θ is performed with respect to real values of k. The justification for the result is based on arguments by K. Chandiramani and G. B. Witham which identify a proportionality between the Lagrangian energy density and the derivative with respect to frequency of the imaginary part of the surface impedance of the plate.
Random superharmonic resonance
Authors: Rajan Srivatsa , Davies Huw G.
Published: October 2005

The third-order superharmonic mean-square response of a Duffing oscillator to narrow-band random excitation is analyzed. The analysis uses harmonic balance to separate the primary and superharmonic responses and statistical linearization to obtain a differential equation for the autocorrelation of the superharmonic response. The main emphasis of the analysis is to demonstrate the effect of excitation bandwidth on the response and stability of the oscillator. The analysis shows that a tripling of both excitation frequency and bandwidth occur and that occurrence of multivalued superharmonic response and jump phenomena depends upon the bandwidth of excitation. Stability of the superharmonic response is considered by generating a mean-square phase plane which shows that multiple values represented by two stable sinks and an unstable saddle point occur when the excitation bandwidth is small. The phase plane reduces to a single sink when the excitation bandwidth is increased. Stability is also examined by perturbing the superharmonic response equation. The locus of vertical tangents and stability regions are obtained. The theoretical results are compared with results obtained from digital simulation. [Work supported by NSERC, Canada.]
Variance in magnitude and phase of structural transfer functions
Authors: Lyon Richard H. , Cribson Robert
Published: October 2005

The statistics of transfer functions has long been of interest in room acoustics because of the necessity of frequency averaging to generate smooth response curves. An important parameter in these statistics is modal overlap, the ratio of modal bandwidth to modal spacing. When this parameter is large, as it often is in room response, the magnitude statistics become fairly simple. The phase statistics in room response have not been greatly studied. Time delays in room acoustics are more often related to pulse arrival statistics using an image model for wall reflections. Structures are different in the behavior of their transfer functions primarily because they tend to have low modal overlap. This means that both the magnitude and phase of their transfer functions need to be reevaluated. This has been done and the results are presented in this paper. Of particular interest are the phase statistics, which appear to be derivable from a random walk model.
The use of finite-element techniques in statistical energy analysis
Authors: Brown Richard N.
Published: October 2005

When using the statistical energy analysis (SEA) technique to compute the structureborne and acoustic power flow through complex structures, a parameter of paramount importance is the coupling loss factor which characterizes the power flow across substructure boundaries. For many simple boundary types (e.g., plates connected to plates) the coupling loss factors are available in the literature, but in applying the SEA method to structures with more complex connections the finite element method can be used to advantage in computing these factors. The structure studied here consists of two beams, capable of supporting flexural, compressional, and torsional motion, connected by a small, builtup box which has several resonances in the frequency range of interest. The admittance matrix of the box structure was computed using NASTRAN. From this admittance matrix the coupling loss factors are computed. The resonant structure of the “coupler” (the box) is apparent in the results, which compared well with results from experiment.
Resonant frequencies and mode shapes of open prolate spheroidal shells
Authors: Yahner Jack , Burroughs Courtney B.
Published: October 2005

Governing differential equations are derived for an open prolate spheroidal shell with inner and outer surfaces defined by prolate spheroidal coordinate surfaces. Transverse shear and rotary inertia are included in the shell equations. The open end of the shell, defined by an axisymmetric coordinate line, is assumed clamped. Approximate solutions for the axisymmetric resonant frequencies and mode shapes of the in-vacuo shell are derived using Galerkin's variational method.
Objective quality measures applied to enhanced speech
Authors: Hansen John H. , Clements Mark A.
Published: October 2005

Previously reported experiments have indicated that computer enhancement of single channel speech in wideband noise generally does not result in increased intelligibility. Our research goal has been to investigate the effect of such enhancement on speech quality. Our experimental framework employed natural speech degraded by additive white, non-white, and aircraft cockpit noise over a wide range of signal-to-noise ratios (− 15 to 20 dB). An iterative enhancement technique based on a method reported by Lim and Oppenheim, was generalized to allow nonwhite additive noise. Spectral estimates of the noise were made using maximum-likelihood method, maximum-entropy method, Burg's method, periodogram, Bartlett's method, and Pisarenko harmonic decomposition. Quality was estimated using a variety of objective measures which have been shown to correlate well with subjective quality scores resulting from the diagnostic acceptibility measure (DAM). Over a large number of noise conditions the best objective qualities resulted when Bartlett, Burg, and maximum-entropy spectra were used. In general, significant improvements in terms of objective measures were observed when enhancement took place. [Work supported by Lockheed Georgia Company.]
Evaluation of a two-microphone speech-enhancement system
Authors: McConnell M. V. , Zurek P. M. , Peterson P. M. , Rabinowitz W. M.
Published: October 2005

Two variants of a processing scheme first proposed by Kaiser and David [J. Acoust. Soc. Am. 32, 918 (1960)] were evaluated for enhanced intelligibility. In both systems the running cross correlation between two microphone signals is used to derive a time-varying gating signal that multiplies the sum of the microphone signals to form a monaural stimulus. With a target source straight ahead and an interference source off axis, this processing results in an output that is increasingly attenuated with decreasing target-to-interference ratio. In the first system this processing was applied to the wideband inputs and in the second it was applied independently in four octave bands. With the latter system three degrees of processing (determined by the function relating output attenuation to cross-correlation) were investigated. Intelligibility tests were conducted using sentences as the straight-ahead target and continuous discourse as the off-axis (90° azimuth) interference. Contrary to expectations based on the report of Kaiser and David, in no case was speech intelligibility improved as a result of processing. Apparently, the benefits of removing remote interference (temporal and spectral) do not outweigh degradations of the target signal. [Work supported by NIH.]
Optimizing control rooms for stereo imagery
Authors: Jones Douglas R. , Martens William L. , Kendall Gary S.
Published: October 2005

Control rooms designers typically measure and specify rooms according to their physical structure and acoustic properties. They are unable, however, to measure or predict how well the room will support the subjective qualities of stereo imagery produced over loudspeakers. As the quality and salience of stereo imagery improve through the use of more sophisticated recording and processing techniques, control room requirements become more stringent. Beyond speaker placement, there are three primary factors that influence the perception of stereo images: time-energy-frequency characteristics of the speakers, spatio-temporal distribution of early reflections, and the inclusion of acoustic diffraction. These are easily measured through the use of time delay spectrometry (TDS), but at present an adequate model for predicting subjective response from these physical measurements is lacking. Ensuring the perception of optimal stereo imagery requires the application of standardized subjective evaluation techniques. Currently under development at Northwestern Computer Music (NCM) is an evaluation technique using the Listening Environment Diagnostic Recording (LEDR™), which enables the immediate assessment of changes in stereo imagery that result from progressive changes in control room acoustical treatment. Field tests indicate that LEDR™ is valuable in the design and modification of control rooms for optimizing stereo imagery.
Recording control room design incorporating a reflection-free zone and reflection phase grating acoustical diffusors
Authors: D'Antonio Peter , Konnert John H. , Bilello Charles
Published: October 2005

The acoustical diffusing properties of flat, mono, and bicylindrical surfaces, alternating absorptive and reflective panels, and a wide variety of reflection phase grating diffusors (RPG) will be discussed. Experimental TEF measurements including energy-time, energy-frequency, time-energy-frequency, and polar angle-energy-frequency curves plus theoretical Kirchhoff modeling calculations will be presented. A new control room design incorporating a reflection-free zone (RFZ), in the front half of the room, and a diffuse zone, in the rear half of the room created with RPG diffusors, will be described and documented. The RFZ minimizes the speaker boundary interference over a wide volume surrounding the mix position. The RPGs provide a diffuse sound field which enhances the perception of spatial textures and helps maintain the stereo perspective across the entire width of the mixing console and in the rear of the room. The RFZ/RPG design minimizes frequency coloration, image shifting, and provides accurate stereo imaging.
Stream segregation effects in the perception of syllables
Authors: Garrison Laurie F. , Sawusch James R.
Published: October 2005

Perceptual grouping processes and their role in the perception of speech syllables were investigated using repeated presentations of syllables. This method, often referred to as “streaming,” has previously been used to study perceptual grouping processes for various nonspeech, acoustic stimuli. This experiment utilized a wide variety of synthetic consonant-vowel and consonant-consonant-vowel syllables. Some of these stimuli contained formant transitions while other stimuli consisted of adjacent noise (burst) and vocalic segments with no formant transitions. The effects of stimulus structure upon stream segregation and the implications of these results for selective adaptation studies will be discussed. [Work supported by NINCDS.]
Contributions of the lexicon and sentential context to speech perception
Authors: Connine Cynthia M.
Published: October 2005

In experiment one, we investigated the influence of lexical status on speech perception first demonstrated by Ganong [J. Exp. Psychol.: Human Percept. Perform. 6, 110–125 (1980)]. Two voice onset time (VOT) series were presented to listeners for identification: DICE-TICE, the voiced endpoint forms a word, and DYPE-TYPE, the voiceless endpoint forms a word. Similar to Ganong, we found a longer VOT boundary value for the voiced compared to the voiceless lexical bias series. Reaction times for labeling responses showed an advantage for word compared to non-word responses for ambiguous stimuli (at the category boundary) but no word advantage for clear stimuli (at the continua endpoints). Two additional experiments investigated sentence context effects on phoneme identification in a word-word voicing continua (e.g., DENT-TENT). Each stimulus appeared as the final word in sentences pragmatically biased toward the voiced and voiceless endpoint. Voiced biased contexts resulted in longer VOT boundary values than voiceless biased contexts. Reaction times showed an advantage for pragmatically appropriate responses at the continua endpoints and the boundary. The lexical and sentence context effects will be discussed within modular and interactive theoretical frameworks. [Work supported by NIMH.]