bluuebunny/crossref_metadata_2025 huggingface dataset:
Strict passbands of waves on flat, periodically supported, fluid-loaded, elastic layers
Authors: Smith P. W.
Published: October 2005

A “strict” passband is defined as a frequency interval in which the Floquet wave on a periodic surface propagates without any attenuation. Their existence is characteristic of conservative systems in vacuo. Examples of strict passbands on a system in contact with an infinite fluid have been published, but, in each case, the elastic layer is a classical thin plate. In such a band, energy flows along the wetted layer, but none is radiated as sound despite the supports. It is shown that strict passbands exist for any layer, however complex in the thickness direction, if the system satisfies certain broad conditions: (1) The system is conservative; (2) the supporting ribs do not cut the elastic layer; and (3) the layer is statically stiff at wavelengths in the neighborhood of twice the rib spacing.
Resonant frequencies and mode shapes of open fluid-loaded prolate spheroidal shells
Authors: Yahner Jack , Burroughs Courtney B.
Published: October 2005

Governing differential equations for the vibrations of open fluid-loaded prolate spheroidal shells are presented. Transverse shear and rotary inertia are included in the shell equations. The open end of the shell, defined by an axisymmetric coordinate line, is assumed to be clamped. Approximate solutions for the axisymmetric resonant frequencies and mode shapes of in vacuo and fluid-loaded shells are derived using Galerkin's vibrational method and are compared.
Vibrations of damaged trapezoidal plates
Authors: Holger David K. , Jiao Qunying
Published: October 2005

The modal frequencies and shapes of damaged edge-clamped trapezoidal plates have been investigated for the first five transverse modes. Damage considered consisted of narrow, rectangular, centered slots. An impulse hammer was used to excite transverse plate vibration, and plate response was measured with a nearfield microphone. The frequency response of the plate was investigated for all combinations of three slot lengths and three slot orientations. The first five modal frequencies and shapes were estimated from ensemble-averaged, frequency-response functions. Observed changes in modal frequencies for the damaged plates were in good agreement with predictions of a finite element model. The results suggest that slot presence can be detected from changes in plate modal frequencies, that slot length and orientation have significant effect on the modal frequencies, that a nearfield microphone is an adequate response transducer, and that an appropriate finite element model and the frequency response measurement are of comparable sensitivity to damage. It is suggested that combining experimental modal analysis and the finite element method has significant potential for macroscopic, nondestructive evaluation.
In-plane and bending wave analysis of a step discontinuity in a thick plate structure
Authors: McCollum M. , Cuschieri J. M.
Published: October 2005

Having developed the response equations for in-plane and bending waves in a thick L-shaped plate structure of infinite extent (Noise-Con 88), when both plates are of the same thickness and material, the analysis is now extended to plates of different thicknesses. In this study, instead of the L-plate configuration, a flat plate with a step discontinuity is investigated. The boundary between the two plates is excited by a Mindlin bending wave at some arbitrary angle of incidence. In this paper, the complete thick plate solution for the transmitted and reflected waves is obtained using Mindlin's equations to describe the plate flexure, and including the generation of in-plane shear and in-plane longitudinal waves at the junction. The solution is obtained for different angles of incidence and different step configurations. The results that will be presented are for the total transmitted power and the power transmitted by the individual wave components, averaged over all angles of incidence, as a function of the plate step discontinuity.
The acoustic field of an oscillating bubble near a free surface
Authors: Pumphrey Hugh C. , Crum L. A.
Published: October 2005

Bubbles may be entrained into the ocean by a variety of natural processes such as rainfall and breaking waves. It is often postulated that the free oscillations of these bubbles at their resonance frequencies are a significant source of ambient noise in the ocean. Experimental results will be presented which show that a bubble entrained near the surface has the resonance frequency and damping constant predicted by theory, with deviations that may be explained by such factors as the nonsphericity of the bubble and the proximity of the free surface. The bubble is a simple source within a small fraction of a wavelength of a free surface and should therefore have a dipole radiation pattern. Measurements of the field were made; these show that the radiation pattern is as predicted and allow calculation of the amplitude of oscillation of the bubble. It is known from previous work that raindrops in a certain size range will entrain bubbles. Using the known source strength of a bubble, the spectrum level produced by a typical rain shower is calculated and compared with measurements made of real rain. [Work supported by ONR.]
Offshore measurements of the underwater acoustic noise spectrum during rainfall
Authors: Kraeutner Paul
Published: October 2005

The underwater acoustic noise spectrum during rainfall has been measured in a coastal ocean environment for a wide range of sea states and rainfall intensities. The relationship between spectral noise level and rain rate was examined and a power law function fitted to data at 5, 8, 15, 20, and 30 kHz. Finally, a family of curves representing the acoustic noise spectrum for sea states 12, 1, 2, 3, 4, and 5 and rain conditions of 1.0, 3.0, 5.0, and 7.0 mm/h was derived. Results indicate a domination of the acoustic noise spectra by rain-generating mechanisms at frequencies in the band 7–25 kHz and sea states less than 4. Furthermore, the smearing of the rain-generated spectral peak near 15 kHz strongly supports the postulation of a strong dependence of rain-generated acoustic noise on sea surface conditions. [Work supported by DSS Unsolicited Proposals Program, IOS, and DND.]
Radiated noise increase of a submerged turbulent jet flow due to bubble entrainment
Authors: Korman M. S. , Pumphrey H. C. , Crum L. A.
Published: October 2005

Experiments are performed in an open tank (L = 1 m, W = 0.75 m, H = 0.6 m) to study the effects of entrained air bubbles on the noise radiation produced by a turbulent shear flow created by a submerged water jet (nozzle diameter D = 4 mm located 27 cm from the bottom). Using a small hydrophone, measurements of the average frequency spectrum level between 1.9 and 20 kHz are compared for case (1) no bubble entrainment and case (2) bubble entrainment, under the same experimental conditions of the turbulence. A 2.54-em column of bubbles (with radii between 14 mm and 112 mm) is generated below the nozzle exit that becomes entrained in the jet core between 4 and 8D from the exit. Nozzle exit velocities U are chosen below the cavitation threshold of the jet [σ = 2(Pamb − Pvap)/ρU2 and σ > 0.4]. Results indicate an increase of over 10 dB in the radiated noise level measured between 6 and 10 kHz and over 20 dB between 10 and 20 kHz for studies where 0.42 < σ < 1.47. It is hypothesized that bubble breakup may be one cause of this increase. [Work supported in part by the ONR.]
Device for reducing noise in underwater acoustics
Authors: Wagner J. M.
Published: October 2005

In the case of an underwater acoustics antenna, the transducer is usually mounted as follows: One face of the sensor is leaned against a structure withstanding the pressure, and the other one is in contact with water. Any structure vibration is transmitted to the sensor and generates noise that alters the receiving performances. It will be shown that, in the case of an asymmetrical mounted transducer, the sensor sensitivities depend on the excitation direction. This feature is applied in inserting a pressure sensor between the transducer support and the structure. The response of this system is analyzed and shows that the S/N ratio can be significantly increased by subtracting noise. Trials in an acoustic water tank are described and results on real antenna are given.
Absolute and relative pitch identification by absolute pitch possessors
Authors: Miyazaki Ken'ichi
Published: October 2005

The identification of absolute and relative pitch was investigated with absolute pitch (AP) possessors and nonpossessors as subjects. In an AP experiment, isolated tones of different pitches (including microtonal pitches) were presented as stimuli in random order. Subjects were requested to select and press a key corresponding to each stimulus on a musical keyboard. While the responses of non-AP subjects were widely dispersed, the AP subjects could categorize the presented tones quite consistently. Specifically, they identified white-key notes more correctly and rapidly than black-key notes. In a relative pitch experiment, isolated melodic ascending intervals from the minor second to the octave were randomly presented. The lower tone of an interval was either middle C, F♯, or a 50-cent higher D♯. Then non-AP subjects performed more or less successfully and almost equally regardless of transposition. On the other hand, the AP subjects made more errors and gave longer response times when the lower tone of an interval was F♯ or the higher D♯ than when it was C. These results suggest that at least some AP possessors have difficulty in dealing with musical intervals and tend to rely upon their AP sense in the face of a relative pitch task.
Modeling the acquisition of cross-cultural differences in tonality with neural nets
Authors: Bharucha Jamshed J.
Published: October 2005

The acquisition of scale schemas through cultural exposure was modeled with neural net simulations. Associative networks employing the back-propagation learning rule were exposed repeatedly to scale sets of one culture and were then tested for their ability to complete scale subsets of either the same culture or a different culture. The scales were taken from Western and North Indian music. Each network completed the scale subsets so as to most closely match the scales of its own culture. When the foreign scale subset was sufficiently different from any scale to which the network had been exposed, the network showed no clear preference for scale completion. In some cases, the foreign scale was interpreted as being similar to a native scale with a shifted tonic. It is argued that acoustical patterns that occur with considerable regularity in the environment become encoded through passive exposure and serve to generate expectations in context. Reaction time tasks were employed to study these expectancies empirically, for both Western and Indian music.
Sound intensity measurements in a vehicle's interior sound field
Authors: Suzuki Ken'ichiro , Maki Tetsuo
Published: October 2005

Sound pressure and sound intensity in the passenger compartment of a vehicle with many standing waves are measured through piston-plate excitation. These measurements provide the characteristics of attenuation in the free field near the sound source, which is found to be approximately −6 dB per doubling of distance. At a certain distance from the sound source, sound intensity shows an average sound-pressure level between the maximum and minimum values in the passenger compartment. The reason for this is that, owing to the use of sound-absorbing materials, the sound field does not consist only of standing waves, It is revealed that progressive waves dominate near the sound source. As a result, sound intensity measurements in passenger compartments provides the precise location and contribution of the sound source, even when booming sounds are present. Moreover, using the modal analysis method to investigate the behavior of the body vibration from the enforced point to the sound source is effective in improving the body structure for reducing passenger compartment noise levels.
A method for evaluating the interior sound quality of automobiles
Authors: Yamashita Tsuyoshi , Niikura Hajime , Nakamura Mitsuo , Kitamura Otoichi
Published: October 2005

The A-weighted sound-pressure level has been the conventional method of evaluating interior car noise aimed at reducing loudness. However, it is difficult to evaluate the quality of sound only by loudness. The multivariate analysis method was applied using data obtained from a semantic differential questionnaire with over 20 items about four different car models' interior sounds. The results showed that the bias from the image of the models had a larger effect on impressions than the actual sound. This shows that a blindfolded test is very important. Thus, the test procedure was modified to obtain correct results, and a blindfold test was adopted. The test was done with 48 subjects monitoring both the actual sound of the automobiles and their recorded sound. The data were analyzed using the principal component analysis, and three main factors in the sound were revealed: “beautifulness,” “powerfulness,” and “metallicness.” These factors are similar to those already found in studies of other types of sound.
Decoupling fracture parameters from background transverse isotropy in orthorhombically symmetric media
Authors: Hood Julie A. , Schoenberg Michael
Published: October 2005

A system of plane parallel fractures or periodic fine layering renders any elastic medium seismically anisotropic. The anisotropy increases in complexity as the number of different systems incorporated into the medium increases. Using the group calculus formulation for layered media developed by Schoenberg and Muir [submitted to Geophysics], the effects of each individual system can be separated arithmetically after the properties of each system in the anisotropic medium are transformed to an element of an Abelian group. In an orthorhombic medium resulting from fractures orthogonally embedded in a transversely isotropic (TI) background, the contribution of the fractures to the stiffness matrix can readily be subtracted out, reducing the number of moduli from eight to five. The removed moduli provide information on the fracture properties, such as average fracture spacing, microcrack density, and average compliance of fracture-filling material. The remaining simplified (five-parameter) stiffness matrix can be analyzed further to infer the characteristics of the background TI medium. [Work supported by NSF.]
Computer synthesis of acoustic wave fields in laminated and fractured media
Authors: Mallick Subhashis , Frazer L. Neil
Published: October 2005

Frequency-wavenumber integration is used to compute time-domain sonic waveforms at any point in a stratified medium. The source is either a point force, or a point dislocation, at any depth. The medium consists of welded homogeneous layers with all 21 elastic constants varying independently from layer to layer. Currently, a free-surface boundary condition at the top of the medium and a radiation condition at the bottom is used, but these boundary conditions can be easily changed. Although final results are in the time domain, the basic calculations are carried out in the frequency domain so anelastic effects are incorporated by use of complex elastic constants. (Usually, instead of directly specifying the elastic constants in each layer, Schoenberg algorithms were used to “fracture” a layer or to “construct” a layer from many thin microlayers of simpler material. These algorithms have a group property [Schoenberg and Muir, 1988] which makes it easy to generate fractured laminates.) For elastic materials with azimuthal anisotropy, two wavenumber integrations are required. If the medium is many wavelengths thick, and the source and receiver(s) are many wavelengths apart, then several hours of Cray CPU time are needed to generate a complete response. [Work supported by Geo-Pacific Corporation.]
Scholte-Gogoladze-like waves at the interface between a fluid and an anisotropic layered composite
Authors: Braga Arthur M. B. , Herrmann George
Published: October 2005

It is now well understood that two types of plane harmonic waves can propagate freely along the plane interface between a homogeneous isotropic elastic solid and an acoustic fluid. The first, with decaying amplitude as it propagates along the interface, and which is equivalent to the Rayleigh wave in the solid as the fluid density becomes negligible, is called a leaky or generalized Rayleigh wave. The other, named here the Scholte-Gogoladze wave after the two researchers who first, independently, uncovered its existence in 1948, propagates unattenuated parallel to the boundary, while decaying exponentially in both directions away from the interface. In the present contribution, the existence of Scholte-Gogoladze-like waves at a fluid/layered composite interface is investigated. The composite is made-up of an arbitrary number of homogeneous anisotropic layers stacked in a unit cell that repeats periodically. An extension of Stroh's sextic matrix formalism to periodically layered media is employed to carry out the analysis. The associated eigenvalue problem is solved in closed form, and the frequency equation for the interface waves is formulated in terms of the surface impedance tensors for both the fluid and the layered composite. In contrast to the homogeneous medium, the Scholte-Gogoladze-like waves are now dispersive and, for certain combinations of the material parameters, the dispersion spectrum consists of more than one branch in the frequency-wavenumber space. [Work supported by ONR.]
Exterior acoustic scattering/radiation without interior eigenvalue problems
Authors: Shaw Richard Paul
Published: October 2005

The use of integral equation formulations, such as boundary-element methods or T-matrix methods, for the solution of time harmonic exterior acoustic radiation or scattering is hindered by the presence of the “fictitious interior eigenvalue” difficulty that causes the basic integral equation formulation, and thus any numerical scheme based on such a formulation, to fail at certain frequencies related to the interior acoustic problem. A number of techniques have been developed in the last 20 years to circumvent this difficulty; these involve additional computational effort. A method is presented here that eliminates this difficulty analytically by embedding the original obstacle in an infinite acoustic fluid bounded by a simple surface of sources and/or doublets. Such an approach moves the eigenvalue failures from those for the original shape, which are in general unknown, to those for a simple shape which are known. These eigenvalues can be carried through the formulation analytically and canceled exactly, thereby eliminating any subsequent numerical failures.
Measurement of the temperature dependence of the piezo-optic coefficients of liquids using the submersible electrostatic acoustic transducer
Authors: Yost William T. , Cantrell John H.
Published: October 2005

The development of a liquid-immersible megahertz-range electrostatic acoustic transducer was previously reported. The transducer consists of a thin conductive membrane stretched over an electrically isolated cylindrical electrode recessed approximately 10 μ from the membrane inner surface. The transducer is shown to be capable of measuring absolute ultrasonic displacement amplitudes in the subangstrom range. This capability, combined with Raman-Nath light diffraction measurements, permits the measurement of the piezo-optic coefficients of liquids. Measurements of the piezo-optic coefficient in distilled water as a function of temperature are reported. The results are compared with values obtained from other techniques.
Propagation of energy in nonspreading wave packets
Authors: Ziolkowski Richard W. , Lewis D. Kent , Cook Bill D.
Published: October 2005

Is there anything new about the wave equation? Some investigators have found new, unusual solutions. Brittingham started it all in 1984 by publishing solutions describing localized packets of energy that propagated without spreading. A controversy arose as Brittingham's solutions had questionable features including infinite energy, a backward wave component, and a questionable launchability. Ziolkowski demonstrated that by combining many modes of Brittingham's solutions, these problems disappeared. This paper describes a brief history, the basic concepts, the objections, and the major breakthroughs to date. Experimental evidence suggests that a new solution can be launched. [Work performed under the auspices of the U.S. Dept. of Energy by Lawrence Livermore National Laboratory under Contract W-7405-Eng-48.]
Concert acoustics: The performers' perspective
Authors: Kirkegaard R. Lawrence , Mahoney Robert F. , Kirkegaard Dana L.
Published: October 2005

Following each performance during the San Francisco Symphony's recent tours of Europe, Asia, and North America, musicians completed questionnaires evaluating the stage acoutics of the hall in which they had just performed. The questionnaire's format was developed with the input of the Symphony and in deference to an earlier study by A. C. Gade [A. C. Gade, “Musicians' Ideas About Room Acoustical Qualities,” Rep. No. 31, Acoustics Lab., Techn. Univ. of Denmark (1982)]. Subjective parameters included the ability to hear each section of the orchestra (including one's own), the ease of ensemble, the ease of tone production, adequacy of dynamic range, loudness, and the support and responsiveness of the room. These data were compared with echograms recorded at five locations, i.e., 25 communication paths, on each stage. Interpretations of these data have been compared with the published results of research with laboratory-simulated stage acoustics and are presented in this series of papers addressing loudness, communication, and support and responsiveness.
The acoustic design of partially enclosed orchestra pits
Authors: Mackenzie Robin K.
Published: October 2005

The trend during recent years for musical directors of touring opera companies to require productions to be played to a fuller score has resulted in problems for the orchestral musician. Research has found that musicians frequently complain of (a) lack of contact with the stage, (b) diminished ensemble within the orchestra, and (c) lack of feedback from the auditorium. The geometric design of the orchestra pit has been investigated and optimum parameters have been established. A system of electroacoustic reinforcement to improve communication between the stage and pit is described. [Work supported by S.E.R.C.]
Experimental validation of the MAE theory of sound propagation over two consecutive ridges of finite impedance
Authors: Kearns James A. , Berthelot Yves H. , Zhou Ji-xun
Published: October 2005

A difficult problem within the realm of outdoor sound propagation is predicting the acoustic field associated with long-range sound propagation over hilly terrain. Such terrain is often of irregular geometry with variable finite impedance. An appealing approach to handling this problem is to consider the local field around a particular ridge with the hope of splicing together a global solution from a set of local solutions. An initial study was made of the diffraction of a plane wave by a single cylindrical ridge of finite impedance. The method of matched asymptotic expansions (MAE), for large values of kR (where k is the wavenumber and R is the radius of curvature of the ridge), was employed to obtain a solution valid on, near, and behind the ridge. Predictions based upon this solution compared very well with the results of scale model experiments [J. Acoust. Soc. Am. Suppl. 1 83, S93 (1988)]. Recently, scale model experiments have been conducted to study the propagation of sound over two consecutive identical ridges. Preliminary results indicate that the MAE theory is indeed a powerful tool for the analysis of the diffraction of sound over complicated boundaries. [Work supported by NASA Langley Research Center.]
A categorical factor analysis of vowel distribution based on the modified qualification theory
Authors: Kobayashi Tetsunori , Matsuda Toshiyuki
Published: October 2005

Vowels in continuous speech have a very wide distribution in spectral space. However, if the phonemic environment is known, it is possible to compensate for the contextual variation in the spectrum and to make the distribution compact. In this paper, an attempt is made to determine what kind of categorical item (factor) should be considered to make the distribution compact using a modified qualification theory (MQT). Although original qualification theory has the fault in that it cannot deal with correlative relations between items, this MQT can do this because it has a self-organizing function for correlative relation. The results of an analysis of vowels in 492 words are as follows. To compensate for the variance, in /u/, it is most important to know whether the preceding sound is alveolar or not. In case the preceding sound is a labial, the place of the articulation of the next preceding sound is very important.
Phonetic feature extraction based on mutual information
Authors: Aoki Noriyuki , Hosaka Naoki , Shirai Katsuhiko
Published: October 2005

A novel method of feature extraction for phoneme recognition in continuous speech is proposed that employs mutual information between acoustic features and phonemes. Various acoustic features are coded by the vector quantization (VQ) method and a method to discriminate phonemes by the effective combination of these VQ codes is developed. To construct an optimal algorithm for phoneme discrimination, entropy and mutual information, in addition to conditional probability, between phoneme labels and features are also taken into consideration. The effectiveness of each acoustic feature for describing the characteristics of the phoneme in a given environment is evaluated based on the mutual information. The LPC mel-cepstrum. its pattern of temporal changes over frames, and power are used as acoustic features. Three experiments were conducted. The first was on the optimization of the frame labeling. The second was on the detection of the vowel using a sequence of frame labels. The third was on word discrimination. The effectiveness of the proposed method was verified by these experiments.
Word recognition using a two-dimensional mel-cepstrum in noisy environments
Authors: Kitamura Tadashi , Hayahara Etsuro
Published: October 2005

This paper presents a word recognition method using a two-dimensional mel-cepstrum (TDMC) in noisy environments. The TDMC is defined as the two-dimensional Fourier transform of mel-frequency scaled log spectra in the frequency and time domains, and it consists of the average features and dynamic features of two-dimensional mel-log spectra. In comparison with the standard method using a one-dimensional mel-cepstrum, this method gives better recognition rates. Furthermore, the selection of the size of the TDMC used to describe dynamic features, the optimum weighting between the average and dynamic features, and reference noise patterns are taken into consideration in order to improve this method. Speaker-dependent word recognition experiments using 10 Japanese digits uttered by six male speakers indicate that recognition error rates lower than 2% can be obtained for noise-free speech and white-noise-added speech of 20 and 10 dB SNR.
Sentence understanding of spoken Japanese using phrase spotting and dependency grammar
Authors: Ito Tatsuji , Nakagawa Sei-ichi
Published: October 2005

In order to recognize spoken Japanese sentences, a speech understanding system that consists of four stages was constructed. First, the system hypothesizes syllable candidates using an HMM-based syllable spotter. Second, it concatenates the syllables to obtain word candidates, checking the tree-structured dictionary. Third, it connects word candidates to get phrase candidates using a Japanese phrase spotter. At this stage, a finite state automaton that represents the Japanese intra-phrase grammar is used. Finally, the system selects the best phrase sequence as a sentence recognition result using a backward parsing algorithm. This algorithm makes it possible to analyze the syntactic structure of a phrase sequence or a partial sentence using Japanese dependency (kakari-uke) grammar. Japanese sentences for the recognition task were inquiries on the UNIX system and spoken by six male speakers at natural speed. Syllables extracted from the continuous speech were used as training data for the HMM syllable spotter. The system's performance was evaluated from these speech data. The semantic analysis of this system is also described.
The Carnegie-Mellon Portable Speech Library
Authors: Alleva Fileno A. , Thayer Eric H.
Published: October 2005

In order to promote the dissemination of research results and to help advance the general state of the art in speech recognition, a public domain library of plug-compatible subroutines, modules, and programs has been created. This paper describes the Carnegie-Mellon Portable Speech Library (CMPSL). CMPSL provides algorithms written in C that can be used to quickly prototype state-of-the-art speech recognition systems that are useful in their ability to perform the speech recognition task while also providing a basis from which to pursue further research. CMPSL's chief contributions are to make available recent advances in speech recognition at Carnegie-Mellon in the context of Lee's large-vocabulary, speaker-independent recognition system, SPHINX [K.-F. Lee and H.-W. Hon, Proc. IEEE ICASSP-88, 123–126 (1988)], and to suggest a framework within which speech researchers can make the results of their work available to other members of the automatic speech recognition community. The scope of CMPSL includes digital signal processing, training, recognition, and evaluation algorithms. [Work supported by DARPA.]
Consonant discrimination using the formant coarticulation model
Authors: Chiba Shigeru
Published: October 2005

Formant trajectories in the vicinity of consonants in continuous speech are observed to discriminate the place of articulation of the consonants. The coarticulation model [S. Chiba, Proc. IEEE ICASSP-86, Paper 49-10 (1986)] with a target formant frequency for consonants is developed based on the observation of the formant trajetories in natural VCV utterances. The coarticulation model proposed here is as follows: Fb − Ft = a(Fp − Ft) + b(Fe − Ft) + c(Ff − Ft) + E, where Fb, Fp, Fe, and Ff are the second formant frequencies at the centers and the consonant boundaries of the initial and final vowel segments in a VCV sequence, Ft is the target second formant frequency of the consonant, and E is the estimation error. The parameter values of a, b, c, and Ft are estimated by the method of least squares using actual values. Three coarticulation models for labials, alveolars, and velars are estimated and these models are applied to the discrimination of place of articulation of consonants in continuous speech. Experimental results show that the models for labials and alveolars can estimate the formant trajectories accurately and that a high-discrimination rate of 94% is obtained by comparing prediction trajectories with actual values. In the case of velars, however, the discrimination rate is low.
SAIPH: A segmentation system for automatic labeling of a large speech database. Application to speech recognition
Authors: Junqua Jean-Claude , Wakita Hisashi
Published: October 2005

Several attempts for automatic segmentation and labeling of large speech databases have been made. Those previous studies proposed to (1) align the input utterance with the manually labeled reference utterance using dynamic time warping, (2) segment and label the utterance into broad phonetic classes prior to time alignment, and (3) use trained reference patterns. In some eases, phonetic or heuristic knowledge has been used to refine the boundaries. This approach does not use reference units and thus can be applied also to speech recognition. A transition measure derived from the perceptually based linear predictive analysis (PLP) was defined. Basic segmentation is obtained by use of heuristic knowledge based on this transition measure in addition to the energy and the zero-crossings parameters. An evaluation on a keyboard database (104 words) spoken by two speakers (one male and one female) showed that more than 92% of the good segments are obtained after this stage. A broad classification, knowledge of the expected phonetic string, and phonological rules are applied to deal with the oversegmentation and undersegmentation for further improvement.
Speech recognition using least-squares models for the contextual variations in the acoustic characteristics of phonetic segments
Authors: Abe Yoshiharu , Nakajima Kunio
Published: October 2005

This paper describes a speech recognition method using linear models to predict the variations in a feature vector caused by the phonemic context. The feature vector in a phonetic segment is decomposed and formulated by the sum of a context-independent vector, a context-dependent vector, and an estimation error. The second component is given by the product of a weight matrix and an environment vector. Least-squares estimations of the context independent vectors and the weight matrices for top-down, bottom-up, and mixed models are obtained algebraically. The top-down model obtains the environment vector from the phoneme string transcribed in the lexicon, while the bottom-up model obtains it from the input feature vectors. Speaker-dependent recognition experiments were carried out using an open vocabulary of 200 words spoken by two male and two female speakers. The average word error rate of 19.1% without any contextual models was reduced to 7.4%, 5.3%, and 3.3% by the top-down, bottom-up, and mixed models, respectively.
Speech interface in a dialog system
Authors: Kobayashi Yutaka , Niimi Yasuhisa
Published: October 2005

A speech interface for a database is being developed. The system tries to interpret sequences of natural language queries from a user by utilizing a wide range of knowledge sources. The acoustic analyzer transforms the speech signal into a phonetic lattice using the phonetic HMM-based analysis. The syntactic and semantic analyzer predicts those words that might appear next to partial sentence hypotheses. Instead of matching word by word, the unit of matching against the lattice is a part of concatenated word templates bounded by robust phones for dealing with coarticulation effect across word boundaries. The dialog component includes a user model and a topic management mechanism, so as to keep track of the user's intention and to alternate the conversation initiative between the system and the user. Knowledge sources are being refined and statistics are being gathered for a set of queries on sightseeing, where the vocabulary size is 500 × 1000 and the speech corpus contains 120 sentences from each of six speakers. Recent progress is reported in this paper.
Word boundary placement in faintly perceived speech
Authors: Cutler Anne , Butterfield Sally
Published: October 2005

The recognition of words in continuous speech is made more difficult by the absence of explicit word boundary markers. Cutler and Norris [J. Acoust. Soc. Am. Suppl. 1 77, S39 (1985)] proposed that listeners compensate for the absence of explicit word boundary cues by adopting a strategy of assuming each strong syllable to be a potential word onset. Cutler and Carter [J. Acoust. Soc. Am. Suppl. 1 81, S67 (1987)] showed that the structure of the English vocabulary is conducive to the application of such a strategy. In the present experiment, listeners' speech reception thresholds were individually estimated; the listeners were then presented with short unpredictable utterances (e.g., “never just convict them”) close to this threshold. Exactly as predicted by the proposed segmentation strategy, erroneous boundary insertions occurred more often before strong syllables (i.e., syllables containing a full vowel) than before weak syllables, while erroneous boundary deletions occurred more often before weak syllables (i.e., syllables containing a reduced vowel) than before strong Syllables. Word frequency analyses showed that this result was not an artifact of a bias towards high-frequency responses. [Research supported by the Alvey Directorate, U.K.]
Resonance and directionality of linear arrays of monopole scatters
Authors: Tolstoy I. , Tolstoy Alexandra
Published: October 2005

Linear arrays of n identical resonators (bubbles, gas-filled balloons, or very thin shells) spaced a distance l apart and insonified by a plane wave at frequencies close to or below the monopole resonance (the radial “bubble” frequency), display resonances due to multiple scatter interaction. Whereas such systems exhibit true normal modes of vibration (superresonances) only near elastic boundaries in a fluid, they exhibit strong partial resonances (quasiresonances, or QRs) in fullspaces, for both finite and infinite n, with absolute amplifications relative to the free-field amplitude of up to 7 × 102 inside and upon the surface of the resonator. These QR effects have strong directionality: an important result of the multiple scatter interaction is to make the response of individual elements—their effective monopole strengths—very sensitive to the direction of arrival of the plane wave. Theoretical directionality factors of the order of 200 are achieved in this way prior to any form of phasing, filtering, or processing of the monpole outputs qua elements of an array.
Benchmark comparisons of underwater acoustic scattering models
Authors: Norton Guy V. , Werby Michael F. , Collins Michael D.
Published: October 2005

Benchmark calculations for waveguide scattering problems in realistic ocean environments will be presented. The three models used in the study are based on Huygens principle [M. F. Werby and R. B. Evans, IEEE J. Ocean. Eng. OE-12, 380–394 (1987)], a projection approach [G. V. Norton and M. F. Werby, Math. Comput. Model. 11, 81–86 (1988)], and the method of matched asymptotics [M. D. Collins and M. F. Werby, J. Acoust. Soc. Am. (submitted)]. The first two methods utilizes the normal mode solution for a point source in a waveguide along with the T matrix that describes scattering from an axially symmetric object in free space, in order to obtain the scattered field near the object. The method based on Huygens principle uses this scattered nearfield, along with the Green's function that satisfies the waveguide's boundaries to solve for the scattered field far from the object. The projection method, however, projects this scattered nearfield onto the directional multipole solution for the waveguide and the resulting field can then propagate out into the waveguide. The asymptotic method involves the parabolic equation method and is based on a simple approximation of the waveguide Green's function in the nearfield. This method does not require a decomposition of the incident field, which results in computational efficiency. [Work supported by the Naval Ocean Research and Development Activity.]
Exact solutions for sound diffraction by truncated wedges
Authors: Tolstoy I.
Published: October 2005

A previously described method for obtaining exact, explicit solutions for the diffraction by truncated wedges [I. Tolstoy, J. Acoust. Soc. Am. Suppl. 1 82, S61 (1987)] was valid for arbitrary vertex angles θω, θω′ providing these were irrational fractions of π. This has now been extended to the case where these angles can be rational fractions of π. As before, the method combines exact, explicit representations of the field diffracted by a single line vertex with the classic multiple scatter self-consistent algorithm to formulate the interaction between two vertices. Thus there is a general exact and explicit solution for the problem of diffraction of a plane wave by a truncated wedge of any shape. This has been applied to the case of a thick rectangular sound barrier, in the wavenumber range 0.2 ≲ kl ≲ 2.8, where l is the barrier thickness, to calculate the sound field in the shadow. The results are seen to agree well with published experimental results [Medwin et al., J. Acoust. Soc. Am. 72, 1005–1013 (1982)]. The method is quite general and may, in principle, be extended to more complicated polygonal and broken-line profiles with any number of vertices. [Work supported by ONR.]
Acoustic surface reflection loss at low grazing angles in the 0.95–1.35-kHz frequency band—A deep ocean measurement
Authors: Christian Raymond J. , Klingbeil Delia
Published: October 2005

An experiment to measure acoustic reflection loss at the ocean surface was conducted in the North Sargasso Sea in December. A sonobuoy hydrophone received direct path and surface-reflected signals from a dipole element transmitter suspended from a research vessel. Both source and receiver had similar depths of approximately 80 m in the surface layer and separation ranges from 0.5–3 km. Pulsed linear frequency modulated transmissions from 0.95–1.35 kHz of 15-ms duration were used to perform matched filter correlations between the direct and surface reflected paths. Multipath separation was observed and surface reflection loss computations performed at specular grazing angles greater than 5 deg. Surface reflection loss versus grazing angle results under different sea-state conditions are used for comparison with those of the Bechman-Spezzichino scattering theory.
Military aircraft overflight and wilderness perceptions
Authors: Dunholter Paul H. , Harrison Robin T.
Published: October 2005

With the passage of Public Law 100-91, which mandates study by land management agencies of the impact of aircraft overflights on wilderness areas, has come increasing public scrutiny of military aircraft operations over such areas. Under the sponsorship of the U. S. Forest Service and the National Park Service, a preliminary study of such impact is being completed. Measurements of aircraft sound in the wilderness environment and assessment of park management perception of the effects of such sound indicate that such impact is not in itself adequately described by traditional metrics. The effect of sound onset rate on visitor reaction is also explored. Nonacoustic factors such as listener expectations and startle effect play an important role in visitor response to aircraft overflights. Indeed, the current work raises the question whether any objective acoustic measurement correlates with recorded visitor reaction.
The impact of aircraft noise on elementary schools
Authors: Lewis Nelson D. , Russell William A.
Published: October 2005

The noise environment at elementary schools located under the flight paths at two military installations has been measured. These measurements showed that the day-night sound level (DNL) is a poor metric to use to describe this noise environment. The measured and predicted DNL's showed that these schools were located in an area described as acceptable for noise sensitive land uses. However, the measurements show that it will be difficult to maintain speech intelligibility in the classroom in excess of 25% of the time due to aircraft overflights.
Hong Kong International Airport noise study
Authors: Gharabegian Areg
Published: October 2005

Existing noise problems associated with Hong Kong International Airport (HKIA) are severe. Noise levels from airport operations currently exceed government guidelines for at least a quarter of a million people. The measure used to date in Hong Kong to assess noise from airport operation is the noise exposure forecast (NEF). The baseline and future NEF contours were developed using the PC version of the integrated noise model (INM). Noise contours and flight tracks were generated by a computer-aided drafting (CAD) system using the INM data. Noise contour generated with CAD were superimposed on a digitized map of the Hong Kong planning areas to determine the population of each planning area that was affected by noise levels. Due to the unique landing pattern at HKIA, the INM noise profiles were calibrated for several major commercial aircraft. The future noise impact and recommended improvements will be discussed. Information will also be provided about using CAD and INM and calibrated noise profiles.
Noise criteria methodology for site selection of the proposed Hokkaido Space Center in Japan
Authors: Hale Marlund E. , Ikegami Mori
Published: October 2005

The U. S. space program has produced an enormous amount of data useful to environmental studies. Much of the launch noise data, however, is lacking correlation with such propagation influences as meteorological conditions and terrain effects. Thus, otherwise identical noise sources, such as the Space Shuttle, yield wide variations in measured and predicted launch noise. In addition, adequate criteria for assessing launch noise impacts remain an elusive issue within the several agencies and departments responsible for the evaluation of proposed new generation launch vehicles and possible locations for future space launch facilities. In order to develop a site selection criterion based on launch noise for the proposed Hokkaido Space Center, the limited acoustic, meteorological, and terrestrial information available from launches in Florida and California was evaluated, correlated, and scaled to designated launch vehicles and potential sites. The resulting launch noise site selection methodology was used in locating an acceptable site for both equatorial and polar orbit launches while providing consideration for facility support buildings and the surrounding region of potential acoustic impact.
Cross-cultural study on noise problems in apartment houses in Japan, West Germany, the U.S. A., and China
Authors: Kuwano Sonoko , Namba Seiichiro , Schick August , Florentine Mary , Da Rui Zheng
Published: October 2005

Neighborhood noise has become a serious problem in many countries, and it is difficult to regulate it by physical means alone. A cross-cultural survey was conducted in four countries, Japan, Germany, the United States, and China on the residents of apartment houses in order to examine the cultural differences involved. Interesting differences were found in preferred countermeasures, highly annoying sounds, attitudes against noise, expectations for laws, etc. The use of loudspeakers in public places was accepted in the four countries when they were used for conveying necessary information. The results suggest that it is important to take cultural or social backgrounds into consideration in order to find suitable countermeasures.
Real-time pitch detection with a digital signal processor
Authors: Mito Michiharu , Takahasi Kiyoshi , Kurokawa Syuji , Nakamura Syogo , Kubota Tadahiro
Published: October 2005

Pitch detection is an important and essential technique in speech analysis, synthesis, and so on. There are many methods to extract voice pitch, but it is difficult to perform pitch extraction in real time. This paper describes a simple real-time pitch detection algorithm which directly estimates the interval between peaks of the waveform. This algorithm consists of the following two parts: one is the peak emphasis of voiced signals and the other is the pitch detection. The peak emphasis is obtained by running a DFS (discrete Fourier series) and a window operation. It is important to determine the peaks for the pitch measurement because the pitch period is obtained by estimating the interval between successive peaks representing the pitch period of the voiced signal. The peaks related to pitch are determined using a few simple rules. Since the voiced signal waveform includes several extra complicated peaks, these rules are constructed taking into account the characteristics of the voiced signal. Other peaks, which do not correspond to the pitch period, are rejected by a simple logical judgment. A real-time pitch detection algorithm has been realized using a conventional DSP.