bluuebunny/crossref_metadata_2025 huggingface dataset:
Acoustic emission characteristics during magnetization of ferromagnetic materials
Authors: Kusanagi H. , Kimura H. , Sasaki H.
Published: October 2005

The characteristics of acoustic emission generated during magnetization of typical ferromagnetic materials (steel, nickel, and cobalt) in alternating magnetic field of 50 Hz were investigated. Acoustic emission parameters measured were root-mean-square voltage, frequency spectrum, and waveform. The data were taken as a function of stress, magnetic field strength, and annealing time. It became clear that the magnitudes of acoustic emission during magnetization of steel and nickel were strongly dependent on stress and the features of these stress dependencies changed with magnetic field strength. It was found that the magnitude of acoustic emission during magnetization of cobalt was much smaller than those of steel and nickel. This experimental result and the theoretical calculation of elastic energy emitted by domain wall motion show that the motion of 90° domain wall is the most probable source for the acoustic emission during magnetization of ferromagnetic materials.
Tissue characterization from ultrasonic scattering
Authors: Waag R. C. , Lee P. P. K. , Gramiak R.
Published: October 2005

Scattering provides a basis for tissue characterization that is being developed to extend the diagnostic usefulness of ultrasound. Tissue architecture can be related to scattering by a model which expresses a weakly scattered wave as a volume integral involving variations in sound speed and tissue density, transducer fields, and intervening paths. Assumptions of plane wave illumination, farfield reception, distortionless paths, and negligible density variations reduce the integral description of the scattered wave to a Fourier transform of the refractive index variations. When the scattering medium is also weakly homogeneous, an analogous relation exists between average scattered intensity and the correlation function of the refractive index variations. Backscatter measurements by a number of investigators show the feasibility of differentiating tissues and determining spacings. Regular and random model scattering as functions of frequency and angle have agreed with theory. Similar in vitro measurements of scattering by liver show qualitative agreement with architecture observed optically. The results indicate the promise of scattering measurements for tissue characterization but more development is required to identify diseases about which scattering provides information.
A 5-MHz linear array ultrasound system for extracranial carotid artery evaluation
Authors: Barnes R. W. , Riley W. A.
Published: October 2005

A 5-MHz pulsed ultrasound system for studying the extracranial carotid arteries is currently being developed and evaluated within the Department of Neurology. A primary objective of this system is to provide information pertinent to atherosclerosis and other cerebrovascular diseases at early stages of development. The system is designed to provide high resolution dynamic images of the extracranial carotid system within 3 cm above and below the bifurcation and dynamic blood velocity profiles occurring within these arteries throughout the cardiac cycle. Linear transducer arrays consisting of 32 individual elements are used for receiving the imaging and Doppler information. The Doppler-shifted signals from the blood are processed using a phase detection method described recently by Grandchamp and Brandestimi which eliminates the need for individual range gates, [This work was supported in part by NHLBI Contract No-1-HV-7-2925 and the Cerebrovascular Research Center Grant (Ns-06655).]
A new biological effect exposure apparatus that measures total absorbed ultrasound energy
Authors: Christman C. L.
Published: October 2005

A reverberation method is used to determine the total acoustic energy absorbed by a mouse when exposed to ultrasound at clinical frequencies. The specimen is exposed in a cylindrical stainless steel tank supported by wires to minimize acoustic losses using a diffuse field created by frequency modulating the 1.03 MHz carrier signal with white noise having a bandwidth of 285 kHz. In the reverberation method, the rate of change of acoustic intensity, also referred to as the decay rate, is measured using a piezoelectric probe when the source is modulated. The energy absorbed by the specimen is given by the product of the total acoustic input power, the exposure time, and the fractional increase in decay rate caused by the specimen. The decay rate for samples of castor oil enclosed in spherical glass vessels was measured at two temperatures, 20° and 30°C. The absorption coefficient was calculated from the decay rate by assuming the energy density within the specimen was uniform and equal to the average energy density within the tank. The results obtained were within 15% of previously reported values for castor oil. Anesthetized ICR mice were exposed using the apparatus and typical results are presented.
Performance of an expert spectrogram reader. II
Authors: Rudnicky A. I. , Cole R. A. , Zue V.
Published: October 2005

Analysis of the videotaped performance of an expert spectrogram reader revealed the control strategy and sources of knowledge used by the expert while reading spectrograms. Spectrograms were read in two stages: segmentation and labeling. During segmentation, which was usually performed in a single pass from left to right, segment markers were placed on the spectrogram to indicate roughly phone-sized units. Segmentation was accomplished through the detection of spectral discontinuities, although the precise placement of segment markers was guided by a detailed knowledge of acoustic phonetics. Labeling was not performed left to fight, and was usually guided by an “easiest segments first” strategy. The vast majority of segments were identified through the recognition of characteristic patterns and salient features. The identification of other segments required the application of knowledge about coarticulation and phonological rules. Several experiments revealed that the expert did not use higher-order knowledge (e.g., lexical, syntactic, or semantic) during labeling—performance was slightly better on nonsense utterances than English sentences. Implications of the expert's performance will be discussed for theories of speech perception, machine recognition of speech, and the development of aids for the speech- and hearing-impaired.
Segmentation and recognition of phonemes using gross features of the speech spectrum and their dynamic properties
Authors: Miwa J. , Makino S. , Kido K.
Published: October 2005

A method for the segmentation and recognition of phonemes in spoken words uttered by unspecified speakers has been developed. Seven acoustic parameters are extracted every 10 ms from the outputs of the filter bank. The parameters are the frequencies of three spectral local peaks, the speech power, and three parameters expressing the gross pattern of the speech spectrum, which are obtained using a least-squares fit line to the spectrum. The segmentation into phonemes is carried out mainly using dynamic properties, which are the local minimum and maximum of the power and the three parameters mentioned above, in order to decrease errors due to inter-speaker variability and coarticulation. The recognition of phonemes is carried out using all the parameters. Some errors in the segmentation and recognition of phonemes can be corrected using phoneme connecting rules. In an experiment with 17 040 phoneme samples in 166 words uttered by 15 male speakers, the percentage of missed and inserted segments due to errors in segmentation were 4.3% and 10.1%, respectively, and the recognition accuracy was 82%, 60%, 56%, and 85% for vowels, semivowels, voiced consonants, and unvoiced consonants, respectively.
Extraction of acoustic features and the classification of the voiceless allocates /ts/ and /t∫/ in Japanese
Authors: Kunisaki Osamu , Higuchi Norio , Fujisaki Hiroya
Published: October 2005

The acoustic characteristics of voiceless affricates have not been clearly determined. The voiceless affricates /ts/ and /t∫/ in CV syllables are analyzed in this paper to obtain acoustic features useful for their discrimination from the voiceless fricatives /s/ and /∫/ as well as for distinguishing between /ts/ and /t∫/ themselves. The maximum rate of increase in short-term power at consonantal onset was found to be useful for separating the affricates from the fricatives, while the frequency of the spectral zero, extracted by assuming a model with one zero and two poles for the frequency spectrum up to 7 kHz, was found to be effective for separating the two affricates [H. Fujisaki and O. Kunisaki, IEEE Trans. ASSP-26, 21–27 (1978)]. Based on these results, a recognition test was carried out on 100 CV syllables consisting of /ts/, /t∫/, /s/, or /∫/ followed by one of the five Japanese vowels uttered by a male speaker, and a recognition rate of 99% was achieved. [Work supported by Ministry of Education Grant-in-Aid for Scientific Research No. 239005.]
Piecewise-planar representation of vowel formant frequencies across speakers
Authors: Broad David J.
Published: October 2005

A previous study [J. Acoust. Soc. Am. 62, 1467–1473 (1977)] showing that a single female speaker's first three vowel formant frequencies clustered about a two-part piecewise-planar surface is extended to five additional speakers. For each speaker, a similar two-plane representation is found, with the rms spread of the data about the planes being about the same for all speakers. Also, the orientations of the planes for the different speakers are similar: for the data collected so far, the back-vowel planes make an rms angle of about 10° to the average back-vowel plane, while the front vowels make about a 15° rms angle to the average front-vowel plane, Though relatively small, these variations in orientation are significant at the 99% level. As a consequence, the hypothesis of uniform scaling of vowel formant frequencies between speakers must be rejected if it is carried strictly to three dimensions. [This work supported by the AFOSR (AFSC) under Contract F44620-74-C-0034.]
A generalized K-nearest neighbor decision rule for isolated word recognition
Authors: Levinson S. E.
Published: October 2005

A decision rule which assigns an unknown signal to the class for which its average distance to the K nearest samples is minimum is derived. It is shown that this decision rule is equivalent to the classical maximum posterior probability rule based on nonparametric estimates of the local class conditional density functions. We have used the rule successfully in an isolated word recognition experiment and have found that it performs better than other nearest neighbor classification rules. Part of the improved performance is due to the selection of the appropriate value of K. We show how this choice can be tailored to the training set and why the rule is particularly appropriate for our speech recognition work.
A real-time recognition system for spoken words
Authors: Fujisaki Hiroya , Hirose Keikichi , Mizutani Mikio , Sato Yasuo
Published: October 2005

A method is presented for reducing the memory capacity and the amount of computation necessary in automatic recognition of spoken words based on the registration of word templates by individual speakers. The reduction is achieved by representing the spectral information of speech in terms of a small number of feature parameters and by sampling them at a variable rate determined by their rate of variation. A special-purpose hardware unit was designed and constructed for the extraction of the speech parameters, using microprocessors and microprogramming techniques for high-speed processing. On the other hand, a general-purpose minicomputer was utilized for word registration and recognition in order to determine detailed specifications for microprocessor implementation. The performance of the total system for real-time word recognition was tested by using three kinds of vacabulary items, including digits, arithmetic symbols, and railway station names, giving an average rate of correct recognition of 98.1%. [Work supported by Ministry of Education Grant-in-Aid for Scientific Research No. 285070.]
A phonetically based isolated word recognition system
Authors: Blomberg M. E. , Elenius K. O. E.
Published: October 2005

The principal object in using a phonetic approach is the reduction of the influence on recognition rate caused by the intra- and inter-speaker speech variations. The system is implemented on a 16-K minicomputer and uses a filter bank delivering spectral sections, 0–5 kHz, every 10 ms. Estimates of the first three formants are calculated and energies in different spectral bands are used to segment the speech signal into broad classes. The following measures are calculated depending on the segmental class and the speech parameter: mean values, steady-state values, durations, transition rates, and some distances between formants. In a learning phase the statistics of the measures of the used vocabulary are automatically calculated by a program given the quasiphonetic spelling of the input words. The statistics are based on phoneme pairs, i.e., diphones. In the recognition phase the program uses the statistics and the quasiphonetic spelling to recognize the input words. Six male speakers were used for calculating the statistics of a 41-word vocabulary. Their mean recognition rate was 98%, using a new recording. The rate decreased to 96.3% using four male talkers, unknown to the system. [Work supported by STU, Sweden.]
On parsing direction and tree search in the LITHAN speech understanding system
Authors: Nakagawa Sei-ichi , Sakai Toshiyuki
Published: October 2005

We compared two parsing directions in a search space, one which starts the parsing from left to right, and another which first identifies the predicate and then starts the parsing from left to right. The latter method was found to be superior. The predicate can be easily found in Japanese, since it is located at the end of a sentence. For this reason, LITHAN adopts the latter method. LITHAN searches a few best paths with backtracking in parallel using the following algorithm (α-β-γ search). Expand the best a nodes at one time. If the number of new nodes which have been expanded by a selected node exceeds a preset threshold (β), only the best β nodes are kept and others are pruned. If the total number of newly generated word sequences and nonexpanded word sequences exceeds a preset threshold (γ) only the best γ word sequences are kept, and others are removed. Repeat these consecutive processes until a complete sentence is generated. We evaluated this method by simulations and experiments, and obtained good performances both in the search efficiency and in the recognition rate.
Performance evaluation of speaker verification in entry control
Authors: Hydrick Barbara M. , Doddington George R.
Published: October 2005

Texas Instruments has controlled access to its corporate computer facility for four years by voice authentication. Imposter acceptance rate is less than 2%. User rejection rate averages 0.5%, with 0.2% rejection between 9:00 a.m. and 3:00 p.m. and 0.9% between 9:00 p.m. and 3:00 a.m. User rejection rate is 0.3% for single entrants and 0.8% for accompanied entrants. An analysis of this system's voice authentication performance will be presented.
Feature selection using adaptive learning networks for text-independent speaker verification
Authors: Cheung R. S.
Published: October 2005

A nonlinear transformation which makes use of the polynomial discriminant function (PDF) is applied to the selection of features for text-independent speaker verification. For each speaker within the set, a PDF is established such that it maps the original attribute set onto the one-dimensional one resulting in the best discrimination of the speaker from the others in the new feature space. The complicated PDF is implemented using the technique known as the adaptive learning network (ALN). In this case, the multinomial discriminant function is estimated using interconnecting fundamental building blocks each of which computes a two-element quadratic discriminant function. Through a comprehensive training procedure, the coefficients needed to describe each building block and the interconnecting configurations of them are determined. Application of the chosen attributes to a text-independent speaker verification experiment yields relatively low false speaker rejection and verification error rates.
Effects of long-term spectral variability on speaker recognition
Authors: Furui S.
Published: October 2005

One of the most difficult problems in speaker recognition is that the feature parameters frequently vary after a long time interval. We examined this effect on two kinds of speaker recognition; one uses the time pattern of both the fundamental frequency and log-area-ratio parameters and the other uses several kinds of statistical features derived from them. Results of speaker recognition experiments revealed that the long-term variation effects have a great influence on both recognition methods, but are more evident in recognition using statistical parameters. In order to reduce the error rate after a long interval, it is desirable to collect learning samples of each speaker over a long period and measure the weighted distance based on the long-term variability of the feature parameters. When the learning samples are collected over a short period, it is effective to apply spectral equalization using the spectrum averaged over all the voiced portions of the input speech. By this method, an accuracy of 95% can be obtained in speaker verification even after five years using statistical parameters of a spoken word.
Speaker identification by analysis of sound islands
Authors: Wood C. A.
Published: October 2005

A study [Pfeifer, 1978 IEEE Conf. ASSP 283–286 (1978)] using methods for text-independent speaker identification based on vowel characteristics, showed impressive results that improved when using detected vowel-like sounds than when going further and classifying them according to vowel category. It was explained that the improvement may be due to the pooled references being representative of each speaker's vowel space and vowel categorization errors being eliminated. The present experiment attempted to expand upon this finding by automatically spotting vowel-like (including nasals and liquids) sounds, in running speech, through digital signal processing methods. These “sound islands” were then analyzed by a systematic study to determine the best set of parameters to use for speaker identification under these conditions. Some of the parameters tested were LPC, area functions, LP Cepstrum, Reflection Coefficients, Cepstral Coefficients, either mel, log, or linear based, and temporal characteristics. In addition, various classification schemes in both open and closed sets of talkers, were tested and ranked. The results, rankings, and interrelationships of the above algorithms will be described.
The reliability of closure features as cues to medial stop voicing in English
Authors: Lisker L.
Published: October 2005

The evidence that closure duration is a cue to the voicing of medial stops in English trochees is as convincing as any we have for other acoustic features considered to be factors governing the linguistic interpretation of speech signals. Measurements of natural speech show /b/ closures to be regularly shorter than /p/ closures in words such as rabid rapid, and there are experimental data to indicate that this difference has perceptual/phonetic significance. Another closure feature, glottal pulsing, also plays a role in the /b/-/p/ distinction in medial position. New data gathered to test the reliability of these two features as cues to the intelligibility of naturally produced tokens of rabid rapid indicate (1) stop closure duration does not suffice to separate /b/ from /p/ across speakers, (2) the phonetic effect of manipulating silent “closure” differs greatly for different tokens of the source word produced by a single speaker, and (3) the effect of replacing buzz with silence in natural tokens of rabid is similarly unpredictable. Thus it appears that neither the duration nor the acoustic nature of the closure is sufficient to predict listeners' phonetic interpretations of the acoustic segment corresponding to the interval of articulatory occlusion. [Supported by NICHHD to Haskins Laboratories.]
A further parallel between selective adaptation and response contrast
Authors: Diehl Randy L. , Lang Michelle , Parker Ellen M.
Published: October 2005

It is generally believed that selective adaptation effects in speech perception are due to a reduction in sensitivity of auditory feature detectors. We have recently presented evidence that these effects may derive instead from response contrast. In a further test of the contrast hypothesis, we conducted both an adaptation and a contrast experiment with matching sets of stimuli. In the adaptation experiment, subjects identified two series of velar stimuli varying in VOT, [ga-kha] and [gi-khi], before and after adaptation with each of the following test stimuli: [ga], [kha], [gi], and [khi]. In the contrast experiment, subjects identified either of two ambiguous test items (drawn from near the phonetic boundaries of the [ga-kha] and the [gi-khi] series) following a single presentation of [ga], [kha], [gi], or [khi]. For both the adaptation and the contrast experiments, (1) the [Ca] test items were more greatly affected (in a contrast direction) by [Ca] than by the [Ci] adaptor/context stimuli, and (2) the [Ci] test items were not differentially affected by the [Ca] and [Ci] adaptor/context stimuli. These parallel results suggest that vowel contingent adaptation results may well be due to response contrast rather than sensory fatigue.
Structure and function in teleost auditory systems
Authors: Fay R. R. , Popper A. N.
Published: October 2005

Morphological, behavior, and physiological data indicate that there is considerable interspecific variation in auditory function among the more than 20 000 teleost species. This variation includes mechanisms for sound reception, pathways of sound conduction to the ear, excitation patterns within the ear, and possibly, the mechanisms for information processing by the auditory nerve and central nervous system. Although it is still difficult to understand hearing function in terms of our present conceptions of the underlying physiology and structure, it is becoming clear from limited behavioral and physiological data that the auditory systems in at least several teleost species are well adapted for temporal analysis of acoustic signals. This suggestion is in accordance with recent observations that the temporal rather than the spectral patterns of teleost sounds are significant for communication. [Work supported by grants from NIH and NSF to ANP and NSF to RRF.]
Underwater localization—a major problem in fish acoustics
Authors: Schuijf A.
Published: October 2005

Several fish with swimbladders have been shown to detect the angular location of a source of low-frequency pure tones, e.g., 75 Hz, if it is situated in the “horizontal” plane of the fish. Moreover cod possess truly three-dimensional directional hearing. Two variables are pertinent: particle acceleration a(t) and acoustic pressure p(t) at the position of the fish. Directional information in a(t) might be obtained through differently oriented acceleration detectors such as the otolith organs. Electrophysiological evidence supports this. Time of arrival differences appear irrelevant. Bony fish with swimbladders (cod, ide) show a 180° reversal in directional response if a sinusoidal sound field with a(t), p(t) is replaced by a(t), −p(t). This suggests a phase analysis to overcome the 180° directional ambiguity when inferring propagation direction from a(t). Pressure is transformed by the swimbladder into accelerations along a line of action characteristic for the recording site in the labyrinths. Sharks, which lack swimbladders, show acoustic attraction from afar. It is unknown how they are able to localize the source if the detection is also based on a phase comparison system.
The relative importance of initial/final pitch ratio and of pitch peak location for the perception of Serbo-Croation tone
Authors: Purcell Edward T.
Published: October 2005

Ivic and Lehiste presented a number of acoustic factors which may be of importance for the perception of Serbo-Croatian tone. Purcell indicated that three of these factors regularly appeared across three statement and three question environments. Lehiste and Ivic showed, with perceptual data, that the ratio between the initial and final fundamental frequency in the accented syllable could be used by natives as a cue for tone perception. Purcell showed, with perceptual data, that the relative location of the fundamental frequency peak within the accented syllable could be used as a cue for tone perception. The present study will present the results of a multiple regression analysis of perceptual data from an experiment in which these two factors are combined and contrasted. The analysis to be presented will indicate the relative importance of initial/final fundamental frequency ratio and pitch peak location for the perception of Serbo-Croatian tone.
The effect of pitch change on rate normalization
Authors: Souther Art , Diehl Randy L.
Published: October 2005

In accordance with earlier results of Summerfield, the VOT value for the perceptual boundary between GA and KA was found to vary as a function of the tempo of a precursor phrase. The boundary shifts toward shorter VOTs with increased rate of utterance. Analogous rate dependent results were obtained when the perceived voicing characteristic was cued by closure duration rather than VOT. A sizable downward shift in pitch between the precursor phrase and the test item leads to a significant reduction in rate normalization.
Acoustic correlates of perceived stress in an isolated synthetic disyllable
Authors: Isenberg D. , Gay T.
Published: October 2005

In production, syllables that are stressed differ from those that are unstressed on at least four well-known acoustic parameters—duration, pitch, amplitude, and spectral reduction of the vowel. In order to determine how these parameters interact in perception of stress a disyllabic nonsense word—/bapbap/—was synthesized on the Haskins Laboratories OVEIII speech synthesizer. Four series were constructed in which one of the two syllables was systematically shortened, reduced, or lowered in steady state pitch or steady state amplitude. Linguistically sophisticated listeners were able to hear stable and reliable differences in stress only when duration was manipulated. Since duration seemed to be the critical parameter, naive subjects listened to the duration series under two conditions. In the duration judgment condition, listeners were asked to judge whether the first or second syllable sounded longer, while in the stress judgement condition they were asked whether the disyllable sounded more like PERmit or perMIT. Subsequent experiments tested the influence of small changes in one of the two syllables along each of the other three acoustic parameters upon the stress and duration boundaries found when only duration was manipulated. The results relate to how acoustic events convey articulatory effort. [Work supported by NINCDS.]
An auditory—visual interaction phenomenon
Authors: Scott Brian L.
Published: October 2005

Ten subjects listened to the word “dress” presented repetitiously at a rate of 3.5 times/s. As described by Warren and Gregory [Am. J. Psychol. 71, 612–613 (1958)], the repetitious presentation of a word results in perceiving other words. In the case of “dress,” the most commonly perceived alternatives are “stress” and “Esther”; alternatives resulting from resegmenting the word “dress.” In the present study, it was found that a visual presentation of “dress,” “stress,” or “Esther” results in hearing the presented form better than 90% of the time. Random responses to the control words “tress,” “breast,” and “skirt” indicate that the effect is not caused by suggestibility. In addition, a drawing of a dress was found to evoke the auditory perception of “dress” whereas a control drawing elicited random auditory responses. The results indicate a cross-modal integration phenomenon.
Acoustic phase fluctuations caused by Garrett-Munk internal waves
Authors: Hughes B. A.
Published: October 2005

Perturbed acoustic propagation between a point source and an array receiver is considered in which the perturbations are due to a Garrett-Munk internal wave field. Resultant acoustic fluctuations along the array are determined. Straight ray propagation is assumed, and three array orientations are considered: broadside horizontal, endfire horizontal and broadside vertical. It is shown that for an acoustic frequency of 150 Hz, an acoustic range of 50 km, and a horizontal broadside receiver separation distance (in the array) of 1 km, the rms phase difference between the two receivers at 1000-m depth in the North Pacific is typically 5°, whereas in the North Atlantic at the same depth it is typically 60° and in the Arctic it is typically 2°. It is shown that these geographical variations are due mainly to variations in the potential sound velocity gradient. An examination of several profiles which have been calculated for these areas indicate further that the potential sound velocity gradient and hence the phase fluctuation increases strongly as the source receiver depth decreases.
Random wave propagation theory for intermediate and saturated ranges
Authors: Meecham W. C. , Yura H. T. , Tavis M. T.
Published: October 2005

The intensity covariance and other higher moments, as well as the statistics, have been calculated for sound propagating in a single scale ocean to ranges of up to 250 km at a nominal frequency of 1 kHz; thus the calculations range from the weak scattering regime well into the saturated regime. The method begins with the Tatarskii-Rytov calculated results for the second and third moments (the fourth and higher moments are divergent at large ranges using those methods). The Wiener-Hermite stochastic expansion is used to generate the fourth moment (intensity covariance) from these lower moments. We find that the sound field never deviates by more than 10% from Gaussian; the transverse correlation length for the intensity covariance is of order the medium fluctuation scale for shorter ranges (as predicted by Tatarskii for weak scattering); this correlation length reduces with range in the strong scattering regime. There is a second, larger scale—as predicted by some other theories—but its contribution to the intensity correlation is quite small. The normalized variance (mean square) of the intensity fluctuation approaches unity for larger ranges, as would be so if the sound field became Gaussian there—as this theory predicts. Other results will be discussed. [The work is supported by the Office of Naval Research.]
Numerical models of Λ-Φ parameters
Authors: DeFerrari H. A. , Leung R.
Published: October 2005

A theory of propagation through a fluctuating ocean [Flatte et al. (1977)] predicts statistics of acoustic variability in regions of Λ-Φ space, where Λ and Φ are the diffraction and strength parameters. We have developed general numerical methods for the computation of Λ and Φ for arbitrary depth dependence of sound speed C(z) and buoyancy frequency N(z). Numerically computed wave functions, dispersion relationships, and spectra are valid near the surface and for the higher frequencies of the internal wave spectrum, although they are considerably more complicated to compute than the Garrett-Munk (GM) model with canonical profile. The models are used to examine the effects of source-receiver geometry, sound speed, and Brunt-Väisälä profile. Results indicate that Λ parameter is very sensitive to ray geometry. Source-receiver depth can result in order of magnitude changes even at the same range. Rays having turning points near the surface can incur significantly less Φ variation than predicted with GM72. Examples are presented to illustrate these points. [Work supported by ONR.]
Evaluation of acoustic fields with complex rays
Authors: Pedersen M. A. , Watson W. H.
Published: October 2005

Rays with complex parameters are used to evaluate the field on the shadow zone side of a caustic. The analytic continuation of standard (real) ray calculation techniques is utilized. The Newton-Raphson method in the complex plane is used to iterate to the desired real range. Two rays pass through each field point. The ray parameters and travel times, associated with these rays, are complex conjugates while the amplitudes satisfy A1 = −iA2*. The field can then be evaluated by uniform asymptotics. Calculations have been made for a bilinear sound speed model of the deep ocean. Results agree well with those obtained from power series expansions about the caustic point [Pedersen et al., Proc. 9th Int. Congr. Acoust. (Madrid 1977)]. For a particular receiver depth the entire shadow zone has been bridged with a continuous solution connecting the caustic structures which bound the shadow zone. For deep penetration into the shadow zone, uniform asymptotics can be simplified to yield a valid geometrical ray-theory result.
Low-frequency normal modes using improved phase integral methods
Authors: Ryan F. J.
Published: October 2005

Normal mode (NM) theory provides a solution to sound propagation in an inhomogeneous ocean. Use of NM theory is, however, restricted by the ease with which depth eigenfunctions Φn and eigenvalues λn satisfying d2Φn/dz2 + [w2/c2(z) − λn] Φn = 0, plus boundary conditions, may be computed. Employing generalized Langer transformations, uniform asymptotic expansions to Φn, are obtained in terms of parabolic cylinder functions for realistic sound velocity profiles (SVP). The expansions are valid at turning points, require no connection formulae (unlike WKB), and yield complex eigenvalues. The method is asymptotic to A = (2πfh/c0)2 with f denoting frequency, h the SVP scale depth, and c0 the axial sound speed. For deep ocean A ∼ (2f)2, indicating useability to frequencies as low as 3–5 Hz. Results of calculations for Munk's canonical SVP: c(η) = c0 (1 + ε[e−η + η − 1]), η = (z − z0)/h will be described. Extensions to range-dependent environments and inclusion of arbitrary surface and bottom boundary conditions will be discussed. [Supported in part by Naval Ocean Systems Center.]
Mode-theory reflection coefficient
Authors: Davis J. A. , Chin-Bing S. A.
Published: October 2005

Given the physical parameters of the ocean bottom, a plane wave reflection coefficient Rpw can be defined, e.g., the Rayleigh reflection coefficient. Rpw, or the equivalent bottom loss, is normally used in ray-theory calculations. A reflection coefficient, per se, is not an explicit parameter in a normal mode formulation. [See H. P. Bucker, J. Acoust. Soc. Am. 48, 1187–1194 (1971), for an exception to this.] For an inhomogeneous water column an effective plane wave reflection coefficient Re can be defined which plays the role of reflection coefficient in mode theory. Re and Rpw are equivalent only for an isovelocity water column. The phase of Re plays a direct role in determining the eigenvalues and contributes indirectly to the mode cycle or skip distance rC. The relationship between rC and the cycle distance arising from beam displacement rBD [E. L. Murphy, J. Acoust. Soc. Am. 61, 511 (1977)] is shown to be an approximate one. It is further shown that rBD is a fundamental constituent of mode theory and enters explicitly as a parameter in the mode amplitude. Several simple examples are selected to illustrate the various points. [Work supported by CNR/NORDA.]
Footfall-induced vibrations of floors supporting sensitive equipment
Authors: Ungar Eric E.
Published: October 2005

Case histories are presented, illustrating that building floors most often vibrate more as the result of foot traffic on them than as the result of nearby rail or road traffic. Data are summarized from which some vibration criteria may be derived for optical and microassembly equipment. A method for predicting floor vibrations due to footfall is discussed and shown to give reasonable results. Building design and modification approaches for the reduction of floor response to foot traffic are described.
The persistent low-frequency rumble in buildings
Authors: Pelton Howard K. , Carnes Ted N.
Published: October 2005

Lightweight buildings are prime candidates for noise and vibration problems. In many instances residual vibration, airborne sound coupled to lightweight drywall partitions and ductwork tied to partitions, result in high noise levels in adjacent occupied spaces. With the cost of office space today, placing noncritical spaces adjacent to noisy mechanical rooms is not profitable. Some case histories will be given to illustrate the effect of structure borne vibration propagation in buildings that resulted in audible noise in occupied spaces. Solutions for reduction within the realm of practical architectural acoustics will also be provided.
Field measurements of structure-borne sound propagation in buildings
Authors: Ishii K. , Tachibana H.
Published: October 2005

Two types of measurements were carried out: one the impact of a hammer on the floors of the buildings themselves, and the other the impact on the floor of the railway tunnel structure just under and near the buildings. The vibration distribution in the floors and walls of the buildings was measured by accelerometers and the output was squared and integrated in time domain to obtain a result equivalent to steady-state distribution. From these measurements, the following results were obtained. (1) The reduction of vibration mainly depends on the distance from the source, and propagation direction is not always a big problem. (2) It is not suitable to show the reduction of vibration by dB per floor or room. (3) The vibration of lightweight wall is much larger than that of concrete wall at the same room.
Measurements of impact noise using a modified tapping machine
Authors: Warnock A. C. C.
Published: October 2005

Preliminary experience with the modified tapping machine proposed by T. J. Schultz [J. Acoust. Soc. Am. 60, 645 (1976)] revealed some problems with lightweight floors covered with soft carpeting. Measurements of impact velocity and internal impedance of human walkers with a variety of footwear are described. A modified tapping machine based on the measurements of foot impedance was produced. The peak levels produced by this modified machine are compared with those produced by live walkers and standard machines.
Fish and amphibian auditory neuroanatomy
Authors: Glenn R.
Published: October 2005

Morphological variation in the primary projections of the inner ear and acousticolateralis variation in agnathans, chondrichthians, bony fishes, and amphibians is reviewed and used to examine several hypotheses concerning auditory evolution in anamniotes. The acousticolateralis hypothesis regarding the origin of the inner ear is rejected on the basis that all vertebrates, including the earliest fossil agnathans, possess both lateralis and labyrinth systems. Embryonic evidence regarding development of the labyrinth, lateralis, and other special sense organs suggests they are serial homologues sharing a common neural crest-placodal origin, and that all special senses are equally old phylogenetically. The labyrinth hypothesis of the origin of audition can not be rejected or accepted, as there is insufficient evidence regarding the phylogenetic distribution of audition in anamniotes. The hypothesis that audition arose by the invasion and replacement of lateralis fibers by acoustic afferents in the acousticolateralis medullar centers is rejected, based on new experimental evidence that these systems in fishes maintain primarily separate central pathways. Finally, the acousticolateralis area of Latimeria is discussed and compared with that of amphibians.
Multiple reflection of traffic noise in the city area
Authors: Yamashita M. , Koyasu M.
Published: October 2005

In Tokyo, it is very difficult to get the space for road construction. Twelve-million people are concentrated in a limited area. Many houses and buildings are arranged closely on both sides of the street, and city freeways are built frequently above the main street by overhead elevated constuction. In such cases the noise radiated from vehicles is reflected between houses, buildings, the bottom of the elevated road, and the lower street surface, and consequently the sound level observed along the street is higher than that in the case of open area. The effect of the buildup of sound level caused by multiple reflection in city area was studied by means of 1/40 scale model experiments and field measurements. Scale model experiments were carried out concerning various conditions of reflection by using an incoherent line source, and spacial distribution of SPL surrounding the road and the street was observed. Results of the scale model experiments showed similar tendency to that of field measurements. Relations between the buildup of SPL and various factors of sound reflection, for instance, the width of street, the height of buildings, houses, and elevated road and so on, were discussed. The results of this study will be applicable in the road traffic noise prediction at the point located in the roadside.
Probabilistic approach in determining noise exposure from multiple roadways
Authors: Manns Basil , Goldstein Jeffrey , Ma Y. Y.
Published: October 2005

It is well known from empirical measurements and surveys that people in any one location are exposed to noise generated from multiple roadways. In calculating noise exposures from urban traffic, usually deterministic or empirical approaches are used. The deterministic approach usually takes into account only the exposure from one primary roadway. Empirical approaches, although capable of considering both primary and secondary roadway exposures, are not always feasible for many applications. This paper describes a probabilistic approach for estimating the combined primary and secondary exposures generated from multiple roadways contained within areas under evaluation. Incorporating an approach that considers both primary and secondary traffic noise exposures, using deterministic and probabilistic methods, and spatial densities of roadways in various urban areas, a more representative estimation of actual urban noise exposure may be accomplished. Examples using the probabilistic approach will be presented.
Modeling the effects of operational truck noise regulations
Authors: Rackl R. , Manns B.
Published: October 2005

Based on experience in enforcing operational noise standards for medium- and heavy-duty trucks it is, in certain instances, of interest to prescribe a standard more stringent than the maximum noise level vehicles of that type are capable of producing. By taking advantage of the fact that trucks usually need not operate at full power and/or maximum engine speeds on a level street within inhabited areas a “level-street noise standard” would require restrictions on the truck operator in order for the vehicle to meet the requirements of such a standard. A twofold measurement program was undertaken with a view of constructing a truck noise model from which the effects of various possible standards on the truck population could be determined. A small number of trucks was subjected to strictly controlled tests in order to collect data on the relationship between noise levels and such operational parameters as engine speed and engine torque. A large number of trucks was subjected to simpler controlled tests in order to determine the statistical distribution of truck noise levels as a function of truck design parameters and vehicle speed. Both sets of data were combined with data in the literature to formulate the level-street truck noise model.
Asymptotic decay of periodic spherical waves in dissipative media
Authors: Webster Don A. , Blackstock David T.
Published: October 2005

In studies of plane periodic waves of finite amplitude in dissipative media, it has been found that if nα1 < αn, the nth harmonic pressure component pn, decays asymptotically as exp(−nα1x), not as exp(−αnx), where αn is the small signal absorption coefficient for the nth harmonic component. Since generally nα1 < αn, this means that the nonlinearly generated nth harmonic decays more slowly than would be expected from linear theory. Moreover, save for the fundamental, no distance is ever reached at which the linear theory decay is established. In this paper the spherical wave problem is addressed. The results are based on an asymptotic solution of Burgers' equation. It is found that for both weak and strong waves the harmonics decay as exp(−nα1r)/rn. The apparent absorption is the same as in the plane wave case. The apparent geometrical spreading, however, is r−n, not r−1. The asymptotic decay is still, however, slower than would be expected on the basis of linear theory. [Work supported by NASA, AFOSR, and ONR.]
Effect of ultrasound on liquid flow through porous materials
Authors: Fairbanks Harold V.
Published: October 2005

The results of studies where ultrasound was introduced into three different processes involving liquid flow are reported. The systems include (1) crude oil flow through porous sandstone, (2) filtration of slurries containing fine particles, and (3) drying of moist powdered material. It was found that the introduction of ultrasound into these processes produced easier movement for the liquid phase and in case of the slurries, aided in the coagulation of the particles. The introduction of ultrasound appears to change the normal viscous liquid flow movement to a plug like type of liquid flow. The ultrasonic frequency used was 20 KHz with intensities up to 160 dB.
Experiments with lenses for parametric acoustic source beam broadening
Authors: Moffett Mark B. , Konrad William L.
Published: October 2005

For some applications of parametric acoustic sources, broader beams than can be obtained from plane-piston primary projectors are desired. Beam broadening via focusing and divergent lenses is described. The lenses, molded from silicone rubber, had an aperture of 23 cm, but the difference frequency source levels and beam patterns were equivalent to those from plane-piston sources of diameter 3.5 cm for the focusing lens and 2.3 cm for the divergent lens. The primary frequency was 260 kHz, and the difference frequencies ranged from 2.6 to 52 kHz. [Work sponsored by Naval Sea Systems Command.]
Experimental processes in the finite-amplitude excitation of weakly-perturbed rectangular cavities
Authors: Sanders James V. , Kuntsal Ender
Published: October 2005

In an earlier paper [“Finite-amplitude standing waves within real cavities,” Alan B. Coppen and James V. Sanders, J. Acoust. Soc. Am. 58, 1133–1140 (1975)], it was reported that when there was a degeneracy involving any normal mode stimulated by finite-amplitude distortion of a driven normal mode and a normal mode that could not be so stimulated, energy was observed to couple into the “forbidden” normal mode. To test A. B. Coppens theoretical demonstration that weak perturbations of an otherwise ideal rectangular cavity can stimulate “forbidden” degenerate normal modes, experimental studies were conducted on the finite-amplitude excitation of an air-filled, nearly-rectangular cavity. The cavity was perturbed by having two walls slightly nonparallel, and the dimensions of the cavity were adjustable so that the eigenfrequencies of the normal modes in question could be varied about the degenerate condition. The amplitude and phase of the “forbidden” normal mode was measured and compared to the predictions of the theory.