bluuebunny/crossref_metadata_2025 huggingface dataset:
Evaluation of acoustic parameters for monosyllabic word identification
Authors: Davis Steven B. , Mermelstein Paul
Published: October 2005

Several recent investigations have hypothesized that syllable-sized segments may be more appropriate units than phoneme-sized segments for use in continuous speech recognition systems. The significant acoustic information in these segments may be represented by a variety of parameters, some obtained by linear predictive techniques [e.g., linear prediction coefficients (LPC), reflection coefficients (RC), and cepstral coefficients (LPCC)], others by spectral techniques [e.g., linear-frequency cepstral coefficients (LFCC) and mel-frequency cepstral coefficients (MFCC)]. This study compared the performance of these parameters in a word identification test. Two male speakers produced 57 sentences in each of two sessions, and 676 tokens of 52 CVC words in a variety of syntactic positions were manually segmented. For each speaker, a symmetric dynamic warping technique was used for time registration of half of the data to form composite templates and for comparisons of the remaining data against the templates. The local distance functions were the Euclidian metric for the RC, LPCC, LFCC, and MFCC, and the Itakura metric for the LPC. The best parameter for word identification was the MFCC (96.5% and 95.0% for each speaker), followed by the LFCC (94.7% and 87.6%), LPCC (91.7% and 86.4%), LPC (85.2% and 84.3%), and RC (83.1% and 77.3%). We suggest that the superior performance of the reel-frequency cepstral coefficients is due to the fact that they better represent the perceptually-relevant aspects of the short-term speech spectrum. [Research supported by NSF Grant MSC-76-81034 and the USAECOM Contract MDA 904-77-C-0157.]
A spoken word recognition system for unspecified male speakers
Authors: Kido K. , Miwa J. , Makino S.
Published: October 2005

A spoken word recognition system for unspecified male speakers is outlined. The system operates in the following four stages. (I) 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 spectrum. (II) Segmentation and the phoneme recognition are carried out. (III) Errors in the segmentation and phoneme recognition are corrected by means of phoneme connecting rules. (IV) The recognized sequence is determined to be the item of the dictionary having a maximum similarity to the recognized phonemic sequence. The similarity between the item of the dictionary and the recognized phonemic sequence is computed using confusion matrices made from the recognition of 17 040 phonemes. Every item of the dictionary is written in phonemic symbols derived from the word in Japanese “kana” letters by simple rules, so that the contents of the dictionary can be easily changed. The scores for word recognition were found to be 84.0% for 166 words uttered by 25 male speakers and 95.0% for 51 words selected from these 166.
Word recognition using confusion matrices for computing similarities among phonemes
Authors: Makino S. , Miwa J. , Kido K.
Published: October 2005

A method of word recognition using confusion matrices has been developed. In this method, the likelihood of additions, omissions and erroneous phoneme recognitions given by the confusion matrices are used for the computation of the similarity between the recognized phonemic sequence to every item of the word dictionary. Three confusion matrices are made for the initial and final phoneme and the medial phoneme(s) of words. The dynamic programming was adopted to increase the efficiency of the computation. The item of the dictionary having the maximum similarity to the input sequence is chosen as the output of the recognition. The score of the word recognition was 84.0% for 166 words uttered by 25 male speakers, including 15 speakers whose utterances were used for making the confusion matrices, where the averaged score of phoneme recognition was 64.7%. In experiments using a small number of words, the scores were 95.0% and 97.2% for 51 and 20 words, respectively.
Speaker-independent recognition of isolated words using clustering techniques
Authors: Rabiner L. R. , Levinson S. E. , Rosenberg A. E.
Published: October 2005

Levinson et al. have recently described a powerful set of techniques for clustering multiple replications of words spoken by different talkers into a set of composite reference templates. These techniques have been incorporated into a speaker-independent, isolated word recognition system. The vocabulary which we have tested consists of 39 words including the alphabet, the digits 0–9, and the three cueing words STOP, ERROR, and REPEAT. This vocabulary is of great utility for a wide range of applications of automatic word recognition. The features used for recognition are an eight-pole LPC set measured every 15 ms over a standard telephone line. The distance measure is the log likelihood ratio as originally proposed by Itakura. Several variations of a dynamic time warping algorithm have been incorporated into and tested in this system. Using the clustering analysis we have obtained from 2 to 12 clusters per word. The decision rule is a generalized K-nearest neighbor (KNN) rule. Recognition accuracies comparable to those of speaker-dependent isolated word recognition systems have been obtained.
Automatic recognition of spoken spelled names using speaker-independent templates
Authors: Rosenberg A. E. , Rabiner L. R. , Wilpon J. G.
Published: October 2005

An automatic word recognizer is used to accept strings of spelled letters spoken in isolation. The output of the recognizer is a set of best candidate letters for each letter spoken in the string. Candidate strings forming spelled last names and initials are constructed from the candidate letters and compared and matched to name strings stored in a telephone directory. An evaluation was carried out using speaker independent reference templates obtained by the clustering procedures described by Rabiner et al., which operated on speech samples collected from a 100 speaker data base. Ten test speakers spelled a list of 50 names extracted from the directory. The median error rate for spelled letter is approximately 30% while the median error rate for correct matching directory strings is 2%. These results are compared with earlier results using speaker dependent reference templates.
Keyword and phrase spotting by use of the Harpy speech system
Authors: Lowerre Bruce T. , Reddy Raj
Published: October 2005

Two problems of a keyword spotting system that are not encountered in isolated work recognition systems are the detection of the beginning and ending of the words within the context of other words and the phenomena that occur at the word junctures. A modification of the Harpy connected speech recognition system has yielded a word and phrase spotting system that deals with these problems. This word spotting system incorporates the lexical and word juncture knowledge and uses a modified beam search technique of the Harpy system. This enables the system to not only rapidly and accurately spot single words within any context but also multiple words and/or complex phrases. Performance results of the system will be presented for various words and phrases and for different threshold values.
Speaker verification from actual telephone voice
Authors: Ichikawa A. , Nakajima A. , Nakata K.
Published: October 2005

One of the most convenient and direct ways to identify people is talker verification. This is virtually the only method feasible for telephone handset user identification without resorting to additional input devices. One of the remaining unsolved obstacles to the practical use of such a verification technique is the lack of a method to reduce the effects of variations in transmission characteristics due to changes introduced by the telephone connection path or handset utilized. A new means of average self inverse filtering to normalize the effects of linear distortion of actual telephone voice is proposed in this paper. The combined use of average self inverse filtering and selected band analysis improves verification performance beyond that obtained with PARCOR parameters directly extracted from actual telephone voice. Better than 95% verification is possible by the additional use of pitch frequency. In addition, if only one type of telephone handset is utilized almost 100% verification is attainable.
A comparison of community noise abatement strategies
Authors: Konheim Arnold G. , Caccavari Casey
Published: October 2005

As part of its technical assistance program to state and local governments, EPA has been developing a computer-optimized noise abatement model. Based upon the results of acoustical and attitudinal surveys performed by a community, the model considers the relative benefits of several countermeasures, e.g., property-line standards or vehicle noise-level standards. A community using the model will be able to achieve the most cost-effective noise abatement strategy. The results of the implementation of the model at two stages of refinement are compared for three communities.
Spectral shape of flow noise, turbulence, and wall pressure fluctuations
Authors: Clay J. P.
Published: October 2005

An elementary model of how a turbulent boundary layer (BI) generates noise is analyzed. Two similarity hypotheses are given which relate B 1 turbulence to flow noise. (I) Dynamic pressures come from the forces created by the interaction of eddies on the viscous sublayer where velocity components normal to the wall are brought to rest. (II) The forces are proportional to the second power of the eddy velocities and the distribution of separation distance between pressure points on the wall is the same as the distribution of eddy sizes. It is concluded that flow noise, wall pressure fluctuations, and BI turbulence all have the same spectral shape in frequency space: of f−5/3. Data on wall pressure fluctuations generally have a f−11/3 form due to spatial averaging by the pressure sensor. Corrections for pressure sensor averaging are given and experimental data are given to demonstrate the validity of the correction. [Work supported by NAVSEA-03133.]
The influence of consonant environment upon identification of transitionless vowels
Authors: Ohala John J. , Riordan Carol J. , Kawasaki Haruko
Published: October 2005

Recent evidence suggests that consonant environment influences listeners' identification of vowel quality in CVC syllables. The rapid formant transitions are generally held responsible for this effect. In an attempt to find out if the effect may depend on other factors besides the transitions, we had 19 subjects listen to a randomized series of C1VC2 stimuli constructed as follows: C1 was [s] or [f], C2 was [p] or [t], and V was a monophthongal [i], [u], or one of 13 vowels linearly spaced between those two vowels. The vowels were either 75 or 150 ms in duration and were completely steady-state in both F0 and formant frequencies. Ss were required to report whether a given stimulus contained an /i/ or an /u/. Identification of these transitionless vowels was significantly affected by the consonantal environment. The results are compatible with two hypothesized perceptual effects: (1) Listeners “know” that dental environments perturb [u] towards [i] and thus will accept a more front vowel as [u] in that environment than in a labial environment, other things being equal, and (2) listeners “know” that the vowel in a dental environment which should have minimal transitions is /i/. The implications of these findings for theories of speech perception and sound change will be discussed. [Supported by NSF and NIH.]
Why are nonperipheral vowels avoided?
Authors: Hombert Jean-Marie
Published: October 2005

At the last meeting it was shown that the perception of nonperipheral vowels was poor compared to the perception of peripheral vowels [J. M. Hombert, J. Acoust. Soc. Am. 63, 1 (1978)]. The present study is aimed at investigating the extent to which vowels with the same height value but with very different front/back values can be confused. Speakers of three Cameroonian languages (Kom, Limbum, Moghamo) were asked to classify 53 synthetic vowel stimuli (F1 was varied between 250 and 750 Hz and F2 between 650 and 2350 Hz) according to their own vowel system. The results show that stimuli with drastically different F2 values can be perceived as the same vowel. These data are consistent with our previous findings: The perceptual salience of stimuli with F2 between 1000 and 1500 Hz is less than for stimuli with F2 below or above these values. These data explain at least partially why peripheral vowels are preferred (more frequent) in the languages of the world. [Work supported by NSF.]
Effect of vowel quality on perception of nasals in noise
Authors: Zee Eric
Published: October 2005

This study investigates the effect of noise marking on the identification of nasal consonants [m, n, ŋ] in VC syllables, in which V is [i, e, a, o, u]. These VC syllables, spoken by male phoneticians, were masked with white noise (60–4000 Hz), and played to subjects who were professors and graduate students of phonetics. There were four listening conditions with different signal-to-noise ratios. Preliminary results suggest three principal findings: (1) Nasals are correctly identified after [a] even in the noisiest condition, (2) [m] tends to be identified as [n] after the front vowels [i, e] and, less strongly, (3) [n] tends to be identified as [m] after the back rounded vowels [o, u]. These misidentifications of [m] and [n] occur more frequently in the noisier conditions. The results will be discussed in relation to patterns of sound change in syllable final nasals. [Research supported by NSF.]
Discrimination perception of nasal and nonnasal vowels: A reaction times test
Authors: Nartey Jonas N. A. , Goldstein Louis
Published: October 2005

A language-specific reaction time experiment was designed to test the hypothesis that it is because nasal vowels are more difficult to distinguish one from another that languages tend to have fewer contrasts among nasal vowels than among nonnasal vowels. Fifteen native speakers of Gā (a Western Kwa language of southeastern Ghana) were asked to listen to randomized pairs of the language's seven oral and five nasal vowels, and to respond as to whether the pairs contained “same” or “different” vowels. The subjects' reaction times were computed and compared both within and between sets. It was hypothesized that since the only acoustic difference between pairs such as [a-ε] and [ā-ε̄] is that of nasality, a difference in reaction time could be safely attributed to the effect of the presence or absence of nasality. Preliminary results show that the reaction times for nasal vowels were greater than for nonnasal vowels, thus suggesting that vowel discrimination is impaired by nasalization. [Work supported by NIH.]
The perceived nasality of vowels with gradual attenuation of adjacent nasal consonants
Authors: Kawasaki Haruko
Published: October 2005

Phonological studies reveal an apparent paradox. On the one hand, distinctive vowel nasalization often develops diachronically from loss of a contiguous, especially a following, nasal consonant. On the other hand, synchronic distinctive nasalization rarely occurs on vowels before nasal consonants. This paradox could be resolved if it could be shown that non-distinctive assimilatory vowel nasalization becomes perceptually more evident as the nasal consonant (the segment conditioning the nasalization) weakens and becomes less evident. To test this hypothesis, natural-speech tokens of /mIm/, /mUm/, and /mam/ were digitally processed so that the m's were either unaltered, reduced in amplitude to −12, −24, or −36 dB (with respect to their original amplitude), or eliminated completely. Eleven subjects made scalar judgements of the perceived degree of nasality of the vowel in each stimulus, and seven subjects determined the relative nasality of paired stimuli. The results indicate that although the mere presence of a contiguous nasal consonant endows a vowel with some apparent nasality, the gradual reduction of the nasal consonant, as hypothesized, enhances their perceived nasality. [Supported by NSF.]
An adaptive procedure for estimating phoneme-boundaries: Application to selective adaptation for speech
Authors: Raz Israel , Wightman Frederic L.
Published: October 2005

Previous studies have estimated the boundary between two phoneme categories from identification data. This procedure is time consuming, especially in studies of selective adaptation where it is necessary to test identification twice, before and during adaptation. We developed a simple up-down adaptive procedure to estimate the phoneme boundary. The stimulus to be presented on a given trial is determined from the Ss' past responses. Two stimulus continua were synthesized; A voicing continuum (13 stimuli changing from /ba/ to /pa/) and a place continuum (13 stimuli changing from /ba/ to /da/ and /ga/. Ss were given an identification test with each set of stimuli. A logistic curve was fit to the obtained identification function of each subject. Next, phoneme boundaries were directly estimated with the adaptive procedure. The 50% points on the curves from the identification procedure were highly correlated with the 50% points estimated by the adaptive procedure. Phoneme boundaries were also measured before and during selective adaptation (using all endpoint stimuli as adaptors in separate sessions). Boundary shifts were obtained in the expected directions and magnitudes. The adaptive procedure is fast and exhibits high test-retest reliability. [Work supported by NIH.]
Musical intervals in fundamental frequency contours in English intonation
Authors: Levine Arvin
Published: October 2005

Previously reported research [A. Levine, J. Acoust. Soc. Am. 60, S1 (1976)] showed declarative fundamental frequency contours to be analyzable as underlying melodies, consisting of a few discrete tones, interacting with a sentence's syntax and semantics. The syntax and semantics select the underlying melody and determine how it is elaborated to fit the words of the sentence. While doing this research we emphasized the close relationship between music and speech. One interesting prediction from such a relationship is that the pitch intervals between structural elements should correspond to musical intervals. Other researchers [I. Lehiste and G. E. Peterson, J. Acoust. Soc. Am. 33, 4 (1961)] have not been convinced of the reality of musical intervals in speech. We recognize the validity of this opposing work, but think that given suitable criteria for selecting data (independent of the music hypothesis) we can find new evidence for the presence of intervals. This paper attempts to present such evidence in favor of this position, both from acoustic measurements and from perceptual experiments. [Work supported by NSF Grant SED77-09698.]
Stress perception in words varied in fundamental frequency
Authors: O'Shaughnessy Douglas
Published: October 2005

The relationship of fundamental frequency (F0) to perceptual stress was examined. Using linear predictive coded speech, the natural F0 contour on a monosyllabic word was replaced by simple, piece-wise-linear F0 patterns, without altering other aspects of the speech. By analyzing stress judgments using word pairs differing only in F0, this study verified for English and extended some findings reported only in other languages. Acoustic features such as the amount, timing and duration of an F0 movement, as well as relative F0 levels, were systematically investigated as to their effects on stress perception. Using an F0 range of 108–160 Hz, just-noticeable stress differences required F0 changes in the range of 6–17 Hz, depending on the F0 pattern. Similarly, using a syllable with 260 ms of voicing, timing changes of 25–120 ms were necessary to yield stress differences. These results can help form guidelines for speech analysis systems, which need only measure F0 to an accuracy corresponding to perceptible changes. Thus the information rate for F0 in narrow-band-width transmission could be reduced.
Normalization across languages
Authors: Ferrari Disner Sandra
Published: October 2005

One of the few characteristics shared by most of the proposed algorithms for vowel normalization is their utilization of the mean formant frequency values of the vowel system as a correction factor for individual vowels. This reflects the accepted view that the quality of a particular vowel depends less on the absolute formant values than on the relations between these values and the general range of the speaker's formants. While such approaches very satisfactorily reduce the dispersion within a single language, they are unsatisfactory for cross-linguistic comparisons, since the means can differ drastically from language to language when the vowel systems are different. For example, the overall mean of the F2 values of Danish, which has two sets of front vowels (rounded and unrounded) but only one set of back vowels, is significantly higher than that of English; moreover, the overall mean F1 of Danish is over 100 Hz lower than that of English. Using overall means to normalize in these cases confounds phonetic differences with systemic differences. A method of comparing vowels across languages which avoids this problem will be discussed. [Work supported by NIH.]
On the distinctiveness of distinctive features
Authors: Fromkin Victoria A.
Published: October 2005

The notion that speech sounds are complexes of intersecting sets of phonetic features is as old as the earliest attempt to describe speech. Such features are used to describe the languages of the world and also figure importantly in attempts to construct models of speech production and perception. There is, however, still no set of features on which all linguists, phoneticians, speech communication engineers, psychologists, etc., agree, nor is there agreement on whether these features should be defined in physiological-articulatory or acoustic terms or both. This paper will discuss some possible reasons for the continued “confusion,” will examine some phonetic properties which to date have not been proposed as distinctive, and will suggest a mapping procedure by which linguistic “cover features” can be related to physical events. [Work supported by NSF.]
Acoustic variability due to layered finestructure in the Arctic
Authors: Schulkin M. , Garrison G. R. , Wen T.
Published: October 2005

Acoustic transmission measurements, in the frequency range 10–60 kHz, have been carried out in the presence of layered, thermal fine structure under arctic ice. The term “layered fine structure” refers to relatively stable, slowly changing thermal layers in the ocean that have a vertical scale of about a meter to several meters. Their horizontal extent may be 10–100 times larger. In these measurements, acoustic transmission is strongly affected by the presence, near the transmitter, of a layer or layers with a sound speed maximum. In the literature these warm layers are sometimes called “antiwaveguide” or “barrier” layers. Three types of space-time variability have been found in the received intensity versus depth records: (1) A small random variability observed at relatively steep transmission angles, (2) a frequency-independent variability which can be related to convergence/divergence of rays in the refraction field of the layer, and (3) a frequency-dependent variability which seems to arise from the interference between direct rays and those sharply refracted by the layer. A spectral analysis of the interference region confirms that the spatial wavelength in depth is inversely proportional to the acoustic frequency. [Work supported by Arctic Submarine Laboratory of the Naval Ocean Systems Center, San Diego, CA.]
On the scattering of sound by internal waves in the ocean
Authors: Desaubies Yves J. F.
Published: October 2005

The parabolic approximation to the wave equation is applied to the calculation of the spectral density of amplitude and phase fluctuations of a plane sound wave propagating through a random field of internal waves in the ocean. The dependence of the predicted spectra on wave number and frequency shows that the lowest mode internal waves dominate the phase variance, while the amplitude fluctuations are controlled by intermediate scales. The results are compared with previous theories and available data. There is a discrepancy between observed amplitude spectra and predictions based on the internal wave models. It is concluded that internal waves are not solely responsible for the scattering of sound in the ocean and that other effects must contribute.
Low-frequency acoustic fluctuations with a moored pulse compression system
Authors: Spindel R. C. , Porter R. P. , Worcester P. F.
Published: October 2005

A self-contained microprocessor-controlled, acoustic signal processor has been designed to receive long-time series of finite-bandwidth acoustic signals in order to study the effect of mesoscale oceanic variations on acoustic signals. The data capsule is intended to be fixed to deep ocean moorings, where it receives transmissions from moored acoustic sources. The signal is a complement-phase maximal-length shift-register sequence with a bandwidth between 10 and 20 Hz yielding a travel-time resolution of approximately 50–100 ms, sufficient to resolve groups of multipath arrivals having a similar number of upper and lower turning points. The capsule is equipped with an acoustic position keeping system to monitor, and thus account for, travel-time variations due to mooring motion. We present the results of several deployments of the system with source and receivers at various depths and ranges from 2 to 1000 km.
Nature of the caustics in range-averaged ocean sound channels
Authors: Weston David E.
Published: October 2005

An energy-flux calculation for the intensity averaged over a wide range interval shows as a function of depth some nominal infinities, whose nature is exposed here from six viewpoints. First, they may be regarded as a smearing in range of the horizontal ray-theory caustics. Second, note that a collection of rays with the same vertex depth gives a crude model for a mode, the smearing in depth of the resulting caustic produces one of our infinities. Third, a ray-tracing demonstration is provided, with suitable angle distribution and random initial ranges. Fourth, the general energy-flux method shows a logarithmic dependence on vertical distance from the infinity, in agreement with the second approach. Fifth, note that these caustics are not closely related to another mode effect, which can produce a small peak at the source depth even for isovelocity water. Sixth, it is stressed that these infinities are not the same as the ducting peaks, which can occur if both source and receiver are axial and the duct cross section is sharper than parabolic. [Work supported by NORDA and ONR.]
Problem of the caustics in range-averaged ocean sound channels
Authors: Focke Karl C. , Weston David E.
Published: October 2005

The caustics present after range averaging (see companion paper by D. E. Weston) can be understood but pose the problem of how best to deal with them. Infinities as such do not occur in practice, and they prevent the range-averaging method from giving a well-behaved summary of the propagation. The problem is illustrated by depth plots of both energy-flux calculations and range-averaged ray calculations. One solution is to assume some source directivity: In particular, a variation with the square of the grazing angle not only corresponds to the physically meaningful dipole radiation from a near-surface source with its image, but is just sufficient to remove all trace of the infinities. An obvious alternative is to take explicit account of the wave nature of the sound, as represented here by range-averaging of some parabolic-equation field calculations. Another wave approach patches together a solution, replacing the infinity by a peak shape calculated from an Airy function convolution. [Work supported by NORDA and ONR.]
Interrelation of modes and the branch line integral for the “Pekeris” cut
Authors: Bucker H. P.
Published: October 2005

Sound propagation in a uniform duct can be represented as a sum of normal modes and a branch line integral s. Recent work by A. O. Williams and by C. T. Tindle has clarified the contribution of the branch line integral for the problem of propagation in a liquid layer over a liquid half space when the “EJP” branch cut is used. If the “Pekeris” branch cut is used, the sound field is represented by a sum of real (low angle) and complex (high angle) modes and a branch line integral (lateral wave). A complication is that as the frequency is slowly changed a complex mode can move from an improper Riemann sheet to a proper sheet. This must be accompanied by a discontinuous change in the branch line integral because the total sound field must have a smooth variation with frequency. The complicated migration of the normal mode poles with frequency will be displayed and a simple algorithm for the calculation of the Pekeris branch line integral will be discussed.
A method of determining normal mode eigenvalues in an arbitrary number of ocean layers
Authors: Ozaki S. , Ohta K. , Otsubo H.
Published: October 2005

This study concerns a normal mode technique for the computation of sound propagation loss in a horizontally stratified ocean. Normal mode eigenvalues can be classified into the mode families depending on a sound-speed profile. As a function of complex wave number, the determinant in the characteristic equation can be analytic in the domain which corresponds to a mode family. Therefore, all the eigenvalues in one family go along with a smooth curve, along which the real (or imaginary) part of the determinant equals zero. Furthermore, each of them is connected to the real axis in the complex plane with a curve, along which the imaginary (or real) part of the determinant equals zero. To find the eigenvalues numerically, it is suggested one follow these curves as a new procedure. It gives a favorable way of determining eigenvalues and checking of their sequential disposition.
Mode and ray theory comparison for completely absorbing bottom
Authors: Davis J. A. , Chin-Bing S. A.
Published: October 2005

One of the simplest configurations in ocean acoustics consists of a pressure-release boundary at the sea surface, a positive linear or pseudolinear gradient representation for the water column sound speed, and an ocean bottom represented by a semi-infinite homogeneous medium with sound speed and density equal to that at the bottom of the water column. In ray theory this bottom is completely absorbing, whereas in mode theory some energy is returned to the water column. Quantitative differences between the sound field determined by ray and mode theory calculations can be significant at low frequencies. These differences can be understood qualitatively by comparison of the ray and mode reflection coefficients, Rpw and Re, respectively. [See J. A. Davis and S. A. Chin-Bing, “Mode-theory reflection coefficient,” this meeting.] The results for this and other idealized cases suggest that bottom loss analysis for more realistic cases should take into account diffraction effects due to an inhomogeneous water column. [Work supported by CNR/NORDA.]
Asymptotic normal mode propagation in water
Authors: Fukushima Masatada
Published: October 2005

An asymptotic depth function for the sound field in water having arbitrary velocity profiles and surface and bottom reflections is obtained by the extended WKB approximation and connection formulas at turning points. This problem is identical to the quantum mechanical one except for the boundary conditions. Results of applications of the method to problems of interest in underwater acoustics are presented. The surface duct leakage is described as a function of the sound velocity profile and the boundary reflection coefficients.
Transmission loss through double-layered gratings of simply supported hallow bars in a fluid layer
Authors: Radlinski R. P. , Rembetski F. W.
Published: October 2005

The transmission loss of a normally incident plane wave through dynamically compliant gratings has been extended to multiple layers. The gratings, which consist of various sized simply supported bars that have flexural resonances in the frequency range of interest, are immersed in a fluid layer between two fluid half spaces. By treating the periodic structure as a wave guide, the solution is formulated in terms of series of propagating and evanescent plane waves. For certain cases, generalization to non-normal incidence is discussed. The transmission loss bands resulting from staggering fundamental resonances of a single element and/or various elements within the gratings are computed. The effects of the properties of the fluid layer, grating interactions, and material constants of the compliant elements are considered. [Work supported by Naval Sea Systems Command.]
Random incidence energy transmission of bending waves at junctions
Authors: Irie Yoshihiko , Nakamura Tadao
Published: October 2005

This paper describes analytical expressions and numerical results for the random incidence transmission efficiencies of bending wave at typical types of junctions. Analyzed junctions are formed by semi-infinite flat plates, perpendicularly intersecting with each other, for which arbitrary plate thickness and material property are permitted. To obtain a basic formula of transmission efficiency, the analysis for oblique incidence energy transmission at the cross-type junction is carried out, and the total transmission efficiency is expressed by the sum of partial transmission efficiencies among the incident and each transmitting plates. The above basic formula can utilize for other types of junctions, L and T types, by omitting some arguments in the formula, corresponding to each junction. As numerical examples, the transmission and the reflection losses for oblique incidence at each incident angle are calculated for L-, T-, and X-type junctions as a function of the plate thickness ratio, and the average transmission loss for each type junction is obtained from the results of oblique incidence. These calculated average transmission losses for L, T, and X types completely agree with the Cremer's results for the case when plate thickness ratio is unity, and increases around 2–4 dB more than that in the normal incidence over the same range of thickness ratio.
Hardware for the control of structure-borne sound in buildings
Authors: Mason Norman J.
Published: October 2005

The control of structure-borne sound in buildings is reviewed from the viewpoint of a manufacturer of vibration isolation hardware. In addition to describing the available hardware, and the proper selection to satisfy both acoustic and functional requirements, common mistakes in hardware selection and installation will be discussed.
Impact-noise reduction by floor coverings
Authors: Nakamura S. , Koyasu M.
Published: October 2005

Measurements were made under various impact conditions as follows: Hammers of mass = 200 g, radius = 50 cm; mass = 500 g, radius = 50 cm; and mass = 500 g, radius = 2.5 cm; and fall height of ∼32 cm. First the usual frequency characteristics of ΔL were obtained by a 13-octave-band frequency analysis for a single impact sound with a multichannel analyzer operating in fast and Max-hold modes. Since the shape of the ΔL curves are, as is well-known, very simple and resemble each other, it is enough to take f0, the frequency where the curve rises up from the horizontal axis. In addition, our results show a noticeable close relationship between f0 and such level difference as L(dBC) − L(dBA), measured with a test covering. Finally, it was found reasonable to illustrate the experimental results on a diagram of MHR−1/2 vs f0H−1/2; that is, the properties of a sample are shown as a single curve indicating clearly the individual characteristics. For instance, the slope of the curve for carpet has a sudden change at a certain impact condition, and so-called needle-punch carpet is quite different from a usual elastic plate.
Assessment of impact-sound floor insulation
Authors: Yasuoka M. , Fujii H.
Published: October 2005

The fundamental method of calculating sound insulation for various kinds of slab materials is described theoretically, based on the assumptions that there is a well-diffused bending wave field in a floor slab and that the impact-force power spectrum of the tapping machine depends on floor covering and slab materials, while that of the tire is assumed not to depend on these materials. After proving on the basis of many kinds of experimental results that these assumptions do not matter in practical application of the calculating method, the total assessment system that is provided for the purpose of the JIS evaluation indexes is shown. These indexes can be obtained from the two types of impact-sound level, caused by a tire or tapping machine, and the inverse A curve. Using the assessment system, many kinds of practical floor constructions, such as bare concrete slab, floating floor, and timber joist floor, are classified on a matrix of the two L values.
Resiliency of gymnasium floors
Authors: Ono Hidenori , Yoshioka Makoto
Published: October 2005

The resiliency of gymnasium floors, as well as slipperiness and durability, should be regarded as one of the most important measures of its performance from the viewpoint of athletes' psychology and physiology. A fundamental investigation of the resiliency of gymnasium floors was carried out based on the relations between dynamic deflections of gymnasium floors measured with a newly developed apparatus for measuring resiliency and athletes' sense of resiliency scaled by means of the sensory test. From the result of the correspondence between the athletes' preference scale of resiliency and physical characteristics of gymnasium floors, the resiliency of gymnasium floors is defined as follows: Resiliency of gymnasium floors consists of softness, repulsion, and vibration damping. Softness, repulsion, and vibration damping of floors are found to correspond to the physical characteristics of gymnasium floors measured with newly developed apparatus. Then, the estimating equation, which consists of a preference scale of resiliency and the physical characteristics of gymnasium floors, is presented, and the most suitable value of resiliency for gymnasium floor proposed.
National roadway traffic noise exposure model
Authors: Manns Basil , Rudder Fred
Published: October 2005

This model is one element of a continuing effort to quantify the exposure of the U. S. population to noise generated by motor vehicle traffic. This model simulates the noise generated by motor vehicles operating on the national roadway network. To calculate noise impact estimates both deterministic and statistical techniques are used, incorporating roadway configurations, population density distributions, and vehicle classes. A large data base is utilized to characterize the national roadway network, the traffic conditions on all roadways, and the distribution of the population surrounding the roadway network. Population growth, changes in roadway characteristics, and changes in traffic conditions that vary from year to year can be accounted for. Path and receiver control measures can also be incorporated into the model for decision making purposes. This paper discusses the model, illustrating its flexibility as a decision making tool in assessing national community noise exposure and changes due to various control measures as a function of time.
The effects of joint distribution of velocity and PWL of vehicles on the road traffic noise
Authors: Mishina Y. , Kuno K. , Ikegaya K.
Published: October 2005

To evaluate the road traffic noise, a simplified traffic model is usually used, that is, a model with vehicles of the same acoustic power and the equal headway distribution. But in an actual road, fast and slow vehicles and noisy and quiet ones are mixed randomly. The resulting increase in Leq due to variance of PWL and velocity of vehicles has been investigated for the following three cases: (1) when variation coefficient δv of velocity of vehicles is rather small (i.e., δv ≪ 1), (2) when the velocity distribution of the vehicles is the normal distribution, or (3) in the case of gamma distribution of the velocities. As the variance of PWL is 2.55–3.27 (according to the results by Y. Watanabe et al.), the estimated increase of Leq is about 1 dB. And as variation coefficient of velocity of vehicles is 0.13–0.18 (according to the published data on Route 1, Route 16, and Meishin highway), the increase of Leq is about 0.1–1.0 dB. Summing up these effects, the increment of Leq is supposed to be 1–2 dB.
A statistical study for estimating the instantaneous wave form of road traffic noise by three new estimation methods of generalized Kalman's filtering theory
Authors: Hatakeyama K. , Ohta M. , Hiromitsu S.
Published: October 2005

The problem of estimating the state of a stochastic environmental system from noisy observation is of central importance in the actual engineering field. Furthermore, from the technical point of view, there is very often a necessity of estimating especially any of statistical values like the higher-order moments, in the actual case where the observation process is a non-Gaussian process. In order to establish a unified method treating generally the above estimation problem under these actual needs, the so-called Bayesian point of view is first employed here. The chief purpose of this paper is how to extend the well-known results of Kalman and Bucy in the field of linear filtering and prediction problem by finding Bayes' theorem in a new form of unified series expansion, which is suitable for finding a recursive algorithm matched to our successive observation. Three different methods of extending Kalman's filter are newly derived in the unified forms of different type, for each of which recursive estimation formulas can be obtained in the form of a very compact and general solution for the so-called linear filtering (or prediction) problem. In addition, it is noticeable that the above new results include some generalization on the type of estimation processes, viz., those that could be generated by passing arbitrarily distributed input noise through an arbitrary nonlinear dynamical system (possibly time variant). Finally, the validity and the effectiveness of our theoretical results are experimentally clarified through several applications to actual urban noises in Hiroshima City.
The structure of the amphibian auditory periphery: A unique experiment in terrestrial hearing
Authors: Lombard R. Eric
Published: October 2005

The amphibian auditory periphery is traditionally viewed from one of two perspectives: (1) that it is degenerate and by implication derived from the amniote condition or (2) that it represents, at least in most frogs, a simple intermediate condition between the ears of rhipidistian fishes and reptiles. These two perspectives, or various hybrids, guide modern workers in the design of their investigations and the interpretation of the data resulting from these studies. The development and diversity of the amphibian auditory periphery is reviewed and compared to that of amniotes. The otic region in key fossil tetrapods is reviewed. Our current concepts of otic homology are reviewed and some alternative homologies are suggested. This review indicates that the most parsimonius perspective on the amphibian auditory periphery is that it represents a unique experiment in aerial hearing unrelated to that found in modern amniotes. The implications of this perspective for modern morphological and experimental work are discussed. [Work supported by NSF.]
Optimum value of sampling period in digital sound level measurement
Authors: Hattori S. , Minagawa M.
Published: October 2005

Empirically, 1 s is widely used as the sampling period in the digital measurement of sound level. In order to clarify the theoretical background of this optimum sampling period, we consider the correlation between the data of each sample and discuss the effectiveness of the data from statistical point of view. In the case of general road sound, the optimum of sampling period is the value of Δt for which the autocorrelation coefficient ρ(Δt) of the wave form is equal to 12, where the envelope wave form is rms value obtained by the integration circuit corresponding to the FAST dynamic characteristics, by weighting the sound with A characteristics. This value is a function of the distance between the lane and the measuring point on the road and the vehicle speed. We also measured various kinds of road sounds and obtained the autocorrelation of the time series for each kind. For present status of the road condition in Japan, the optimum of sampling period is in the order 0.5–1.0 s.
An estimation theory of level distribution over a long time interval on the basis of the level distribution over a short time interval under a digital measurement of random noise and its experiment
Authors: Hiromitsu S. , Ohta M. , Nishimura M.
Published: October 2005

A strictly stationary random process seldom exists in practical random phenomena such as urban noises within a long time interval. Furthermore, noise data within a long time interval has recently been automatically measured in the form of digital level and discrete time. Accordingly, in order to predict and/or estimate the state variables in non-stationary urban noise, it will be necessary to find a skillful method of data analysis by use of digital computer. Concerning the actual road traffic noise with many different types of level distribution forms, at first a new trial of a statistical treatment to estimate the noise level distribution within a long time interval is considered on the basis of any information of the noise level statics (with a high degree of experimental reliability) within a short time interval. The above estimation technique is suitable to the form of digital level and any finite number of data. Finally, the validity of the above estimation theory is experimentally confirmed by applying it to the data of actually measured road traffic noise. The experimental result is in good agreement with the theory.
Predicting noise from stop-and-go traffic
Authors: Peppin Richard J. , Manns Basil H.
Published: October 2005

Most previous work in traffic noise prediction has been based on uniform spacing (and flow) of motor vehicles. This paper presents the results of a literature search on the topic and the progress to date of an analytical prediction method that can be used to determine the noise produced by stop-and-go traffic. Discussion includes the effects of nonuniform spacing, varying speeds, queuing, and varying volume. The acoustical effects of congested vehicle flow in a nonreverberant space is considered. Suggestions for future work are discussed.
Tire noise mechanisms
Authors: Bender E. K.
Published: October 2005

A comprehensive evaluation of previously published and newly identified tire noise generating mechanisms is presented. Of the numerous excitation, response, and radiation mechanisms investigated, two are found to be dominant. These are tire vibration due to roadway and tread pattern irregularities and air pumping associated with tire deformation at the tire/roadway interface. Experimental and analytical evaluations of these mechanisms are conducted to determine their radiation efficiencies. The vibration mechanism is characterized by acoustically slow bending waves with radiation primarily from the structural nearfield surrounding the tire/road interface. Air pumping radiation from crossbar tires is characterized by a simple monopole at low frequencies and by an acoustic resonance in the vicinity of 1000 Hz. Because of the effect of roadway baffling, the efficiency of this source is found to be greater than previously reported.