bluuebunny/crossref_metadata_2025 huggingface dataset:
Breaking waves and bubble clouds: Acoustic observations of the upper ocean boundary layer
Authors: Farmer David M. , Vagle Svein
Published: October 2005

When surface waves break, they modify the acoustic environment of the upper ocean boundary layer in two important ways. The breaking events generate sound, and the injection of bubble clouds modifies the near-surface, sound-speed distribution. Individual breaking events contribute to a spatial and temporal pattern of sound in the ocean that can be used to infer their temporal and spatial distribution on the ocean surface. The bubble clouds are organized by the boundary layer turbulence. Reduced sound speed within the clouds can refract sound generated by the breaking events and even selectively trap it within the surface bubble layer. Modification of the sound generated by breaking events may therefore serve also as a probe of the bubble layer properties. Observations of these processes will be described. The measurements were obtained with a novel free drifting instrument that employs both active and passive acoustic sensors.
Underwater noise due to the interaction of single water drops with the air/water interface
Authors: Kim Eui Jun , Park Kwang Joon , Yoon Suk Wang
Published: October 2005

Underwater noise due to the interaction of single water drops with the air/water interface is experimentally studied. In an individual water drop experiment, the noise level is increased as the water drop becomes larger, and its impact velocity is raised. In a consecutive water drop experiment, it is observed that a critical falling rate of water drops exists for abrupt change in noise level. The noise due to water drops larger than 3.42 mm in diameter becomes abruptly quiet above their critical falling rate. However, the noise due to water drops smaller than 3.11 mm in diameter has peaks in the frequency range of 12–18 kHz above their critical falling rate. These experimental results confirm that the bubble induced by the drop impact is a more dominant noise maker in water than the impact of a water drop on the air/water interface.
Reduced ocean surface backscattering from under an artificial sea slick
Authors: Castile Brett D. , Rohr Jim , Glass Ray
Published: October 2005

Experiments were conducted in which acoustic backscattering from under an artificial ocean surface slick was compared with that in the absence of a slick. Initial tests were made with an AN/AQS-13 helicopter-dipping sonar. Subsequent tests were made using transducer heads from a MK-46 and a MK-48 torpedo mounted on a remotely steerable device that was suspended below a ship. The latter system allowed control over the acoustic frequency and angle of incidence at the sea surface and limited the size of the scattering area. At steep grazing angles of 40 deg or more, scattering strength under the slick was only about 2 dB less than that outside of the slick at 30 kHz, 4 or 5 dB less at 20 kHz, and 8 dB less at 10 kHz. The apparent reduction in surface scattering strength disappeared at shallow grazing angles. [Work supported by Office of Naval Technology.]
Use of high-resolution roughness power spectra in predicting bottom backscattering
Authors: Briggs Kevin B. , Stanic S.
Published: October 2005

Seafloor relief has been measured at very high (<1-cm) resolution with underwater stereo photogrammetry in the course of environmental characterization for high-frequency acoustic experiments. Information concerning bottom microrelief for the purpose of validating backscatter models has been heretofore expressed as rms height roughness. The validity of using Gaussian statistical methods to describe bottom relief is examined with the Hinich bispectrum test [Brockett et al., J. Acoust Soc. Am. 82, 1386–1394 (1987)] for normality in continuous data. Because bottom backscatter models such as the composite roughness model require the slope of the two-dimensional roughness power spectrum as an input parameter, generation of the two-dimensional relief spectrum is preferable to utilization of the rms relief or estimation from the one-dimensional spectra. Two-dimensional roughness spectra generated from photographs of rippled and smooth bottoms are compared to one-dimensional roughness spectra produced from the same photographs. Model and data comparisons are made for bottom backscattering using both roughness spectra.
Controlling an articulatory model
Authors: Ladefoged Peter
Published: October 2005

Nobody knows what the brain tells the vocal organs to do when producing speech. As a starting point, a computer model has been built in which there are the following independent articulatory parameters: height and backness of the body of the tongue, lip height, lip width, lip protrusion, jaw opening, tip of tongue curling, tip of tongue advancing, tongue root advancement, tongue bunching, and larynx raising. An alternative specification of the body of the tongue uses front raising and back raising gestures, each of which can be combined with the other in different degrees. There are also several options for the interdependent action of the parameters. Instead of being independently specified, the lower lip and the jaw can be determined by the gesture required for the tongue, or the tongue height and lower lip position can be determined by the jaw position. Lip protrusion can be considered to be a coordinative structure that also specifies lipwidth. Larynx height and tongue root advancement can be considered to co-vary. These and other possibilities will be demonstrated in a model of articulatory-acoustic relations in a (public-domain) program on a Macintosh computer.
Cross-language EPG data on lingual asymmetry
Authors: Marchal A. , Farnetani E. , Hardcastle W. J. , Butcher A.
Published: October 2005

Conventional x-ray views of the vocal tract give only limited information about the lingua-palatal contacts occurring during speech production. From these types of data, it is not really possible to evaluate the real size and the exact shape of the vocal tract. When needed, the area at different sections must, in fact, be reconstructed using a mathematical model [S. Maeda, J. Acoust. Soc. Am. Suppl. 1 65, S22 (1979)]. A review of literature indicates that little attention is paid to the potential asymmetry of the articulatory gestures. This phenomenon appears, however, to be particularly crucial in speech pathology [Suzuki, Jpn. J. Oral Maxillo. Fac. Surg. 30, 45–54 (1984)], where lingual and palatal asymmetry is often very severe and causes a poor speech intelligibility. But, even for normal speech, there is some evidence that lingual asymmetry may occur [Hamlet et al., J. Acoust. Soc. Am. 79, 1164–1169 (1986)]. Moreover, it has also been suggested that an asymmetrical tongue placement would be necessary to achieve the expected acoustic output for certain consonants [C. H. Shaddle, J. Acoust. Soc. Am. Suppl. 1 77, S85 (1985)]. To assess lingual asymmetry quantitatively, it is proposed that the following index be computed: Ias = (Na − Nb)/(Na + Nb), where Ias is the index of asymmetry, N is the number of contacts, a is the right side, and b is the left side. The details of tongue to palate contacts for consonants and clusters in English, French, and Italian in nonsense syllables or words embedded in sentences were investigated. They were read by seven native normal speakers. The data demonstrate for all speakers an asymmetrical lingual pattern. Asymmetry is not language bound and seems to reflect a more general phenomenon. Alveolars appear to be more asymmetrical than velars. For stops, the asymmetry is most important at the articulatory release.
Effects of speaking rate on formant trajectories and their interspeaker variations
Authors: Imaizumi Satoshi , Kiritani Shigeru
Published: October 2005

Variations in formant trajectories due to speaking rate were investigated for /VCVCV/ utterances, Vε[a,i] and Cε[b,d,g], which were recorded from three normal speakers and two Broca's aphasics with apraxia of speech. For the normal subjects, the timing of the F3 transition toward the following stop consonants relative to the onset of the vowels, and also the transition rate of F3, showed large variations depending on the speaking rate. The magnitude of the timing and rate changes varied depending on the subjects. As for F1 and F2, variations in the onset frequencies and the transition rates due to the speaking rate were smaller than the interspeaker variations. For the aphasics, the formant trajectories showed large interutterance variations, but no systematic variations as a function of speaking rate. [Work supported by Grant-in-Aid for Scientific Research on Priority Areas, the Ministry of Education, Science and Culture, Japan.]
Preference test of seven European concert halls
Authors: Morimoto Masayuki , Maekawa Zyun-iti , Tachibana Hideki , Yamasaki Yoshio , Hirasawa Yoshio , Pösselt Christoph
Published: October 2005

Preference tests of seven European concert halls using musical motifs recorded through a dummy head were performed. Test signals were reproduced binaurally. Eighty-eight students served as subjects. The results show that (1) there was no significant difference between preference scores for these halls when averaged over all subjects, and (2) subjects could be divided into several groups with respect to preference. In conclusion, even though each subject was free to prefer hall A to hall B, the acoustical quality of a certain hall could not be solely evaluated using the preference score. [Work supported, in part, by The Kajima Foundation.]
Speech coding by Model Reference Adaptive Control
Authors: Hashimoto Kiyoshi , Yasuhara Makoto
Published: October 2005

Model Reference Adaptive Control (MRAC) coding [M. Yasuhara, Trans. IEICE E71 (1), 34–42 (1987)] was applied to speech signals. The primary feature that characterizes MRAC coding is that it is not a predictive coding, but control coding. It determines the control signal so the output of a plant (an ARMA speech production model) follows the speech waveform. The control signal is quantized and transmitted to the receiver. Since at the receiver only a noisy state vector of the plant is observable, a Kalman filter is embedded in the plant to estimate the optimal state vector. Experiments were conducted to evaluate the transmission performances in the S/D ratio for the MRAC coding in comparison with those for the forward ADPCM. With a rate of 2.25 bits/sample, the AR order of the plant between 5 and 12, and the MA order of 1, the average performance for the MRAC coding is around 20 dB, which is 6 dB higher than for the FADPCM.
On improving voice periodicity prediction in codebook-excited LPC coders
Authors: Lin Daniel
Published: October 2005

Codebook-excited predictive coders are able to synthesize high-quality speech at bit rates of 8 kbits/s and above. However, at lower bit rates (e.g. 4.8 kbits/s), speech enhancement techniques, such as adaptive post-filtering, are needed to improve subjective performance [cf. P. Kroon and B. S. Atal, Proc. ICASSP (1987)]. The main source of quality degradation at 4800 b/s is due to the reduced transmission bandwidth of the excitation parameters, which results in an inaccurate representation of the LPC innovation signal. In this paper, methods are proposed for improving the voiced speech excitation and prediction in the codebook-excited coder. The modification uses a composite (i.e., product) excitation codebook together with a close-loop pitch prediction filter that is optimized relative to the source codebook. Several optimization creiteria for the source and predictor are compared in terms of performance and computational complexity. The best result shows a segmental SNR improvement of nearly 2 dB over the conventional stochastically excited LPC coder.
A theory for the use of windows in cepstral processing
Authors: Fuller Chris R. , Hardin Jay C.
Published: October 2005

Cepstral processing is often used to recover the spectrum of signals in which echoes or multipath propagation is present. Often the form of removal of echo information from the cepstral domain takes the form of a boxcar window. However, there is presently no theory to predict the influence of a cepstral window on the form of the recovered spectrum. Consequently, the use of a cepstral window is often based on a purely qualitative procedure. This paper introduces a theory that predicts the effect of a boxcar cepstral window on recovered spectra. The theory demonstrates that the actual spectrum can be recovered from the windowed spectral estimate via a purely multiplicative frequency-dependent correction factor. The theory is then applied to a signal often encountered in practice, consisting of an echoed narrow-band peak on noise. The bounds over which the cepstral window can be used with reasonable accuracy are established. Numerical simulations are then used to test the accuracy of the theory and shown to be in good agreement. [Work supported by NASA Langley.]
Optical modal analysis compared to modal analysis with digital electronic equipment
Authors: Jansson Erik V. , Molin Nils-Erik
Published: October 2005

Hologram interferometry and speckle interferometry have introduced powerful tools for “optical modal analysis.” The vibration amplitudes at a single frequency are recorded simultaneously for all points of a surface. The excitation position and frequency are adjusted to give a normal mode by watching the vibration patterns in real time. In the common modal analysis with digital electronic equipment, the vibrations of a single point as a function of time are recorded for all frequencies simultaneously via the impulse response. Impulse responses for a large number of points are measured, analyzed, and recalculated to extract the normal modes. Investigations of two modes of a complicated object (a violin) using both methods showed differences. The position of the dominating antinodes agreed, but the distribution of the vibration amplitudes and especially the nodal lines did not always agree. The differences seem to depend mainly on methods. The differences imply that great cautiousness is motivated in interpreting unknown vibration modes of objects with complicated structures and boundaries. The real time observation with the optical modal analysis offers hereby advantages.
Real-time two-dimensional blood flow imaging using the ultrasonic Doppler effect
Authors: Kasai Chihiro
Published: October 2005

A newly developed two-dimensional blood flow imaging system that uses the ultrasound Doppler effect is presented, in which blood flow behavior in a given cross section of a living organ is displayed in real time. To perform this, an autocorrelation technique is employed. In the technique, ultrasonic pulses are transmitted several times in the same direction and the phase shifts of the returning echoes from moving blood cells are extracted. The blood flow data obtained are displayed in color on a TV screen superimposed on tissue images that are displayed in black/white. The flow direction, speed, and variance are expressed by differences in color and brightness. Experiments were conducted with a mechanical and an electrical scanner, and good agreement with the theory was obtained. Studies on clinical significance have also been carried out at hospitals, and the usefulness of the system has been demonstrated. The system is now becoming an indispensable tool in diagnosing circulatory diseases such as heart diseases and vessel diseases.
Noise suppression effect of a linear predictor and a coding scheme for narrow-band speech communications
Authors: Arai Yasuhiko , Yagi Toshio
Published: October 2005

The relation between the perceptual noisiness of quantization error and the number of prediction coefficients was experimentally studied. It was revealed that a speech coder (a kind of APC) with ten coefficients shows considerably less noisiness than that one with four coefficients. Though improvement of the segmental SNR was only 1 dB, perceptual noisiness was improved by an equivalent value to a quantizing accuracy of 1.5 bits (approximately 9 dB). Taking this result into consideration, a coder scheme was proposed for special-purpose, narrow-band, wireless communcations. The narrow-band speech coder to be used in very noisy channels having a transmission error rate of 10−2 or 10−1 requires good intelligibility and less perceptual noisiness. It was confirmed that a source coder having eight or ten prediction coefficients and an error correction code for side information would meet the requirements. Even if the residual information fails to be transmitted, the source decoder is capable of synthesizing intelligible speech from only the side information including coefficients.
Effects of degradation in vowel steady-state and transition on the perception of coded speech
Authors: Ozawa Kazunori , Hanada Eisuke
Published: October 2005

In order to investigate the perceptual importance of the vowel steady state and transition on the perception of coded speech, and to obtain cues for further reducing the bit rate in speech coding methods, coded speech was systematically deteriorated during vowel steady states and transitions. Multipulse excited speech coding (MPC) [T. Araseki and K. Ozawa, Proc. GLOBECOM, 794–798 (1983)] at 16 kb/s was used as a speech coding method. Perceptual experiments were carried out using 18 kinds of meaningless sequences having a Japanese CVCV structure including 5 vowels, 2 fricatives, and 6 stops. The intelligibility of the vowels and consonants was measured. The results showed that the intelligibility for vowels was the same as that of the original 1 kb/s MPC, even when all of the multipulse excitation information during the vowel steady states was deleted and the multipulse information during transition parts was reduced to 13 of the 16 kb/s MPC. The intelligibility began to decrease when the multipulse information during the transitions was reduced to 16 of the 16 kb/s MPC. The results, including the intelligibility of the consonants, will be discussed in detail.
Development of a model for multidimensional identification experiments
Authors: Braida Louis D.
Published: October 2005

When several aspects of a stimulus are identified simultaneously, the pattern of errors made by an observer is generally not well described by a unidimensional Thurstonian decision mode. Although such data are often interpreted with the aid of multidimensional scaling techniques, this approach describes only very crude aspects of confusion matrices. A multidimensional generalization of the Thurstonian decision model has been developed that permits relatively precise predictions of error patterns to be made from parameters that describe sensitivity and bias. This model has been applied to identification experiments using vibrotactile stimuli varying over large ranges of intensity, frequency, and contactor area [Rabinowitz et al., J. Acoust. Soc. Am. 82, 1243–1252 (1987)]. Analysis reveals, for example, that sensitivity for a given stimulus dimension is generally reduced when both that dimension and another dimension are identified. This reduction is partially responsible for the failure of multidimensional stimuli to transmit as much information as performance for the separate dimensions would suggest. Also, for vibrotactile stimuli, the perceptual coordinates corresponding to frequency and contractor area are relatively orthogonal while that corresponding to intensity is skewed relative to these. [Work supported by NIH.]
Pitch of complex tones
Authors: Ebata Masanao , Matsuyama Youichi , Suzuki Shin'ichi , Usagawa Tsuyoshi
Published: October 2005

There has been much research on low pitch, but the mechanism of the perception of low pitch has not yet been made clear. In this paper, the characteristics of the pitch of complex tones were compared with those of pure tones. The effect of duration on pitch discrimination for complex tones was almost the same as that for pure tone, but the slope of the DL curve as a function of the duration was a little slower for the complex tone than for the pure tone in the region of shorter duration. The pitch shift due to the auditory fatigue observed for the pure tones was also observed for the complex tones. The effect of a preceding tone on the pitch of a complex tone was also observed. The pitch of the complex tone was influenced by a preceding tone with the same frequency as one of the components of the complex tone. Based on these observations, the process of pitch extraction for complex tones was discussed.
Comodulation masking release with listening-condition uncertainty
Authors: Wright Beverly A. , McFadden Dennis
Published: October 2005

Detectability of a 1250-Hz tonal signal, masked by 50-Hz-wide noise bands centered at 850, 1050, 1250, 1450, and 1650 Hz, was determined for two listening conditions (correlated and uncorrelated) under three types of uncertainty about the listening condition. Here, “correlated” means that the temporal envelopes of all five noise bands were the same, and “uncorrelated” means that only the band centered at 1250 Hz had a temporal envelope different from the other noise bands. In the no-uncertainty condition, all trials of a block consisted of correlated, or uncorrelated, waveforms. In the random-by-trial condition, correlated and uncorrelated trials were interleaved at random. In the random-by-observation-interval condition, the nonsignal interval of each trial contained the same type of waveform as the signal interval only by chance. Levels of detectability, and thus the magnitude of the comodulation masking release (CMR), were essentially the same in the first two conditions. However, in the third uncertainty condition, the CMR was practically abolished, owing almost entirely to a 6–8-dB decrease in detectability with the correlated waveforms. These results suggest a contribution of cognitive listening strategy to signal detection in comodulation masking release conditions. [Work supported by NINCDS Grant NS15895.]
Temporal effects in vowel and consonant-vowel masking
Authors: Blake Patricia E. , Bacon Sid P.
Published: October 2005

Masking patterns were determined using 300-ms synthetic vowel (/u/, /a/, /i/) and consonant-vowel (/bi/, /gi/) stimuli as maskers in a two-interval forced-choice experiment. Thresholds were obtained for 10-ms sinusoids presented at 0- or 145-ms delays relative to masker onset for the vowel (v) masker, and at 0-, 40-, or 145-ms delays for the consonant-vowel (CV) masker. Signal frequencies ranged from 250–3600 Hz. In general, the masking patterns reflected the shape of the masker's short-term spectrum computed at the time of signal presentation. The 0-ms signal delay masking patterns for /bi/ and /gi/ clearly revealed the second-formant differences in these two stimuli. The 40-ms delay masking patterns were similarly different, though less so, even though 40 ms corresponded to the onset of the steady-state portion of the vowel (and thus at this delay the stimuli were identical). Formants were resolved the best for both the V and CV maskers with the 145-ms signal delay, suggesting that the auditory system requires a certain amount of time to represent most accurately the acoustic spectrum of the masker. These temporal effects are probably related to the temporal effects observed previously with pure-tone maskers. [Work supported by NIH.]
Low commonality between tests of auditory discrimination and of speech perception
Authors: Espinoza-Varas Blas , Watson Charles S.
Published: October 2005

The relation between measures of speech processing and of fine auditory discrimination abilities was investigated by means of factor analyses of the performance of normal listeners on the Test of Basic Auditory Capabilities, TBAC [Watson et al., J. Acoust. Soc. Am. Suppl. 1 71, S72 (1982)]. The eight subtests of the TBAC include syllable identification in noise and discrimination of (a) changes in frequency, intensity, and duration in 1.0-kHz sinusoids; (b) “jitter” within a pulse train; (c) temporal order for sinusoids and nonsense syllables; and (d) detection of single-component changes in word-length tonal patterns. Data were collected in a field (n = 127) and using earphones (n = 119). Test reliability was high in both presentation conditions. Average performance was slightly higher with earphones for discrimination of intensity, of tonal patterns, and in the two speech subtests. Both groups showed large ranges of individual performance on all subtests. Factor analyses revealed similar structures for both groups, consisting essentially of two factors loaded on the nonspeech tests (complex sound/pitch and duration/intensity) and a third that is mainly a speech factor. This result, and the associated inter-test correlations, points towards only a very weak relation between listeners' measured abilities to hear out fine details of acoustic waveforms and their skill at identifying speech sounds. [Work supported by NIH and AFOSR.]
The content of “timbre”
Authors: Kitamura Otoichi
Published: October 2005

The definition of timbre has been established by standards such as the American National Standards. According to the results of factor analytical research on sound color, the content of sound color can be well expressed by such descriptive adjectives as soft, rich, clear, etc. This aspect of sound color can be called the “impression expressed by a descriptive adjective of sound color.” Furthermore, there exists another aspect of sound that identifies the sound source such as a piano, violin, etc. Generally speaking, sound impressions are considered to occur in the following order: (1) identification of the sound source, followed by recognition of detailed sound properties such as (2) the impression expressed by the descriptive adjective of the sound color, (3) loudness, (4) pitch, and (5) sound duration. The two aspects of sound, (1) and (2), are usually recognized as sound color or timbre. Accordingly, it is desirable that the content of timbre should be understood more widely as mentioned above.
Continuous multidimensional assessment of musical performance
Authors: Namba Seiichiro , Kuwano Sonoko
Published: October 2005

This study introduces a new method for the assessment of the emotional aspect of music developed by the present authors and termed the “method of continuous judgment of selected description.” This is a method for the multidimensional assessment of a series of instantaneous impressions of a musical performance. Performances of “Pictures at an Exhibition,” composed by Moussorgsky, were used as stimuli. In presenting the results of this experiment, two main points will be discussed. The first is the relation between instantaneous impressions during performances and the overall impression when they come to an end. The second is a comparison of the structure of the music (from an analysis of the performances using the score) with the series of impressions that the subjects registered.
Recent trends in articulatory studies in Japan
Authors: Kiritani Shigeru
Published: October 2005

Ultrasonic measurement, which at present appears to be the most convenient method for recording tongue contour, lacks an effective means of visualizing the hard palate. Techniques have been developed both at the Research Institute of Logopedics and Phoniatrics and at NTT Research Laboratories to mark position of the cranium by monitoring the relative position of the head and an ultrasonic probe. Other techniques employing magnetic or optoelectric devices are also being used for recording articulatory movements. It is expected that a new high-performance magnetic system will enable recording of positions of several magnetic pellets simultaneously through multipoint field recording and analysis. In addition to progress in articulatory measurements, new studies are emerging on the detailed measurements of acoustic effects of several articulatory factors, such as vocal tract wall radiation, the shape of the lip opening and dental structures. A compact, high-speed digital image recording system, developed and now being used at the Research Institutes of Logopedics and Phoniatrics, is also expected to be useful for the analysis of the relationship between vocal cord vibration and the acoustic characteristics of speech output.
Imaging techniques of the tongue and vocal tract
Authors: Stone Maureen
Published: October 2005

The four most well-known imaging techniques that have been applied to tongue and vocal tract research are x-ray, computer tomography (CT), magnetic resonance imaging (MRI), and ultrasound (US). Imaging systems provide very different articulatory information from flesh point tracking systems. They provide information about the shape and position of the entire structure being imaged, rather than a single point on that structure. In addition, the images usually can be made in multiple planes providing midsagittal, parasagittal, coronal, transverse, and oblique sections of the structure. There are, however, several limitations to these systems. CT and MRI create static images only. Ultrasound cannot image hard tissue; x ray can only be used in limited amounts. This paper will discuss the advantages and limitations of imaged data, as well as the uniqueness of the measurements made from them.
Feature extraction of articulatory motion based on the speech production model
Authors: Shirai Katsuhiko
Published: October 2005

This paper deals with two problems that exist in the transformation from an acoustic signal to the articulatory motion. The first concerns the description of articulatory motion during the utterance of voiced sounds. The configuration of the articulatory mechanism is expressed using a rather small number of articulatory parameters, and assuming simple dynamics. It is shown that articulatory movements can be produced by the step inputs for the system, and, since the coarticulation effect is reduced in the stepwise movement of the fictitious target values for each articulatory organ, vowel discrimination can be performed more easily by using these values. The second problem concerns the production model for consonants. Based on this model, the precise vocal tract shape and the position of the noise sound sources can be estimated. The estimation of the vocal tract shape, sound source, and articulatory motion using an articulatory model is a kind of inverse problem of articulatory to acoustic transformation, and requires a nonlinear optimization techinque for its solution.
Elastic waves in porous and particulate solids: Resonances and dead zones
Authors: Junger Miguel C.
Published: October 2005

Simple asymptotic descriptions of elastic wave motions in representative micrononhomogeneous solid media are presented with the purpose of highlighting some physical phenomena. For porous media, the crucial role of the Poisson's ratio is explored: When that ratio lies between 13 and 12, the medium displays a “dead zone” where it cannot sustain wave motion. The propagation of dilatational and shear waves associated respectively with rigid-body translational and rotational resonances is described. Test results illustrate energy dissipation associated with relative motion between inclusions and the viscoelastic matrix.
Wave propagation on thin-walled curved elastic plates with truncations
Authors: Felsen L. B. , Lu I. T.
Published: October 2005

Pierce [ITUAM Elastic Wave Symposium, Galway, Ireland (March 1988)] has recently described the dynamics of thin-walled uniform cylindrical shells by analogy with propagation in an infinite two-dimensional uniform anisotropic medium. This analysis is generalized here, first, by allowing for edge truncations and joints, thereby generating finite or joined cylindrically curved plates. and second, by allowing for weak deviations from circular cylindrical curvature and constant thickness. The resulting equivalent anisotropic medium models involve, respectively, terminated or joined half-spaces, and spatial inhomogeneities. The analytical tools used for the generalization under time-harmonic point force excitation include infinite angular space representations, adiabatic Fourier transforms, reduced spectra (in terms of characteristic plate waves) tied to a preferred spatial coordinate, and anisotropic ray asymptotics. These analytical constructs are given a physical interpretation by recourse to the dispersion surfaces for anisotropic waves. [Work supported by David Taylor NRDC and ONR.]
Relationship between the stored energy density and intensity
Authors: Maidanik G. , Dickey J.
Published: October 2005

A model of a complex, composed of coupled one-dimensional dynamic systems, is proposed. An analytical procedure is presented that yields the steady-state stored energy density vector ε(x) = {εj(xj)} and the net power flow vector ι(x) = {ιj(xj)}, where εj(xj), ιj(xj), and xj are the stored energy density, the net power flow, and an observation position in the (j)th dynamic system, respectively. Under appropriate averagings and assumptions, the equation that describes the stored energy vector can be reduced to the central equation in the statistical energy analysis. On the other hand, the net power flow vector is the intensity. Using the analysis, one may argue that there could exist special situations in which some corresponding elements in ε(x) and ι(x) provide substantially identical information. However, in general, the two quantities are supplemental in the investigation of the energetics of a complex. Notwithstanding that, situations may arise in which the information provided by one or the other quantity may be deemed the more significant in seeking answers to specific questions.
Power flow finite element analysis of beam networks
Authors: Nefske Donald J. , Sung Shung H.
Published: October 2005

A power flow analysis was previously developed for predicting the vibration response of structural and acoustic systems to high frequencies at which the traditional finite element method is no longer practical [D. J. Nefske and S. H. Sung, ASME NCA 3, 47–54 (1987)]. As compared to statistical energy analysis, which predicts an overall vibration response of each individual subsystem, the power flow analysis enables one to predict the spatial variation of the vibration response within each subsystem, as well as the power flow and vibration response throughout the entire system. The present paper considers application of the power flow method to built-up frame structures in the form of beam networks. The power flow analysis considers excitation of multiple wave types in the individual beams and utilizes transmission efficiencies to couple the wavetypes at the beam junctions. The power flow solution is obtained by modifying the properties of a standard structural finite element model to form a power flow finite element model, which is then solved using MSC/NASTRAN. The predictions of the power flow finite element method are compared with the measured vibration on an actual structure and with the predictions of the traditional finite element method.
Axisymmetric vibrations of an elastic fluid-loaded cylinder
Authors: Photiadis Douglas
Published: October 2005

The response of an infinite elastic cylindrical shell immersed in an acoustic medium is examined analytically. The relatively low frequency range, (ka) < 3–4, is investigated. The cylinder is subject to a localized axisymmetric excitation, a ring drive. Starting from the known integral expression for the Green's function, approximation techniques were applied to extract the dominant behavior of the surface field. Expressions for the response are given for distances greater than an acoustic wavelength from the source location.
Experimental identification of a noise prediction model of a vibrating structure using optimal estimation techniques
Authors: Aquilina R. , Flandrin J. C. , Gaud E.
Published: October 2005

A novel statistical method of estimating a noise prediction model of a vibrating structure has been developed using optimal estimation techniques. The aim of this method is to derive the optimal transfer functions between a set of vibratory information taken on the structure, and the radiated noise. An underlying problem to be solved is then the selection of the minimal acceleration subset necessary to “explain” this radiated noise. The statistical approach lies in the achievement of an important number of experimental excitations of the structure and the measurements of the frequency response functions (FRF) of the model inputs (accelerations) and the output (radiated noise). Each noisy measurement is then regarded as a random variable, and the optimal prediction model is obtained by minimizing the variance of residual errors after the selection of the more significant inputs using partial correlation analysis. The measurement noise characteristics are derived from the coherence function information relative to each FRF measurement. Very good experiment results are presented concerning a submersed vibrating shell over a wide frequency range.
A recursive estimation of ARMA parameters based on a robust time varying model for speech analysis
Authors: Serizawa Masahiro , Miki Nobuhiro , Nagai Nobuo
Published: October 2005

This paper presents a recursive estimation of ARMA parameters based on a robust time-varying model for speech analysis. This algorithm is basically similar to the recursive least-squares estimation (RLS), but it is different in that the time variation of the ARMA parameters is dependent on past ones. This is based on the assumption that the speech production process does not vary instantly. This method has two linear estimators: an input estimator and a parameter estimator for known input. The variation of the parameters is estimated by using the likelihood function. The proposed method is equivalent, under certain conditions, to the RLS with the forgetting factor. However, using the proposed method, this factor can be estimated as the value that represents the variation of the parameters. Finally, the proposed method was applied to a synthetic speech and real speech. The results show that the estimated spectra sufficiently represent the dynamic movement of formants without jitters or extreme estimation errors.
Bottom sediments—Viscoelastic, poroelastic, or poro-viscoelastic?
Authors: Stoll Robert D.
Published: October 2005

The velocity and attenuation of seismic waves are known to be frequency dependent in virtually all marine sediments. The frequency dependence results from various kinds of viscous losses in the pore water as well as from intergranular friction in the skeletal frame. The viscous losses fall into two broad categories—losses due to overall motion of the fluid field relative to the skeletal frame and local fluid motion (near the grain contacts) and in cracks and crevices. The cumulative effect of these kinds of energy loss are generally described by a phenomenological model (i.e., viscoelastic, poroelastic, poro-viscoelastic, etc.) with input parameters that are related to measurable physical properties of the sediment. Practical geoacoustic models may range from a simple viscoelastic model with a single relaxation time, valid over a narrow frequency band, to very complex models based on the more general Biot theory that require many more input parameters. In this paper some of the many factors that influence the choice of a practical model are discussed and illustrations of both “simple” and complex cases are given. [Work sponsored by ONR, code 11250A.]
Acoustic propagation in inhomogeneous underwater acoustic waveguides
Authors: McDaniel Suzanne T.
Published: October 2005

The propagation of sound in horizontally stratified media may be treated using a variety of techniques. Realistic shallow-water environments may, however, exhibit significant departures from this ideal geometry. This paper reviews various methods, both deterministic and stochastic, for handling range-dependent shallow-water propagation. Stochastic coupled-mode theory is applied to consider the effects of sea surface scattering. It is shown that for low enough frequencies, adiabatic normal mode theory provides a sufficient treatment, in accord with the findings of both laboratory and at sea measurements. This same technique is also applied to consider the effects of inhomogeneities within the seabed and water column on the propagated field and its spatial coherence. Implications for matched field processing are discussed, and methods of estimating performance degradation due to rough boundary interaction are presented.
A comparison between broadband and narrow-band measurements of bottom properties in shallow water in the Gulf of Mexico
Authors: Lynch James F. , Frisk George V. , Rajah Subramaniam D.
Published: October 2005

In order to assess the relative merits of two different types of acoustic measurements in determining bottom geoacoustic properties, both narrow-band and broadband data taken at the same site in the Gulf of Mexico near Corpus Christi, TX have been inverted. The narrow-band data, taken by Lynch and Frisk, were obtained over a well-sampled 5-km track at 50 and 140 Hz. These data were then processed using the Hankel transform to display the modal eigenvalue structure of the waveguide at those frequencies [G. V. Frisk and J. F. Lynch, J. Acoust. Soc. Am. 76, 205–216 (1984)]. The broadband data were taken by L. A. Rubano [J. Acoust. Soc. Am. 67, 1608–1613 (1980)] in a band between 20 and 250 Hz and yielded a group velocity dispersion curve for the first normal mode. Both types of data were then inverted using perturbative inverse techniques developed by Rajah et al. [J. Acoust. Soc. Am. 82, 998–1017 (1987)], producing both bottom geoacoustic models and error estimates (resolution and variance) for the bottom models. Comparisons of the geoacoustic models with each other, as well as with a third model generated by a NORDA group using a compilation of archival records [Matthews et al. NORDA Report 120 (June 1985)] are made. [Work supported by ONR.]
Propagation loss at Arctic under-ice cover
Authors: Kuo E. Y. T.
Published: October 2005

The propagation loss of low-frequency acoustic energy at Arctic under-ice cover depends on various parameters. This talk gives the combined effects of (1) the under-ice and over-ice roughnesses; (2) the ice layer thickness; (3) the shear wave speed and the absorption represented, respectively, by the real and imaginary part of the longitudinal plate wave speed; and (4) the physical properties of ice and water, At low frequencies, the Arctic ice cover can be approximated by a rough thin plate. Similarly, the roughness scattering process can be approximated by a small perturbation theory [H. Marsh, J. Acoust. Soc. Am. 33, 330–333 (1961); E. Y. T. Kuo, J. Acoust. Soc. Am. 36, 2135–2142 (1964) and 81, 1762–1766 (1987)]. Compared to experimental propagation loss, the new results improve considerably from the previous predictions that were based on a pressure release boundary condition [F. R. DiNapoli and R. H. Mellen, NUSC TM851130 (1985); E. Y. T. Kuo, NUSC TM871118 (1986)]. It is found that the combined effect of roughness scattering and ice absorption dominates over that of the shear wave speed at grazing angles. [Work supported by IR/IED program at NUSC.]
Evidence for the 3-Bark integration interval
Authors: Hanson Brian A. , Javkin Hector Raul
Published: October 2005

It is generally accepted that sound energy within a 1-Bark interval is integrated by the ear. Chistovich et al. [in Frontiers of Speech Communication Research, edited by Lindblom and Ohman (Academic, New York, 1979), pp. 143–157] found evidence of a larger integration interval, of approximately 3 Bark, for vowel perception. Klatt [J. Acoust. Soc. Am. Suppl. 1 77, S7 (1985)] found that listeners could distinguish between vowels whose formant differences were compensated by bandwidth differences (so that cue trading did not occur) and concluded that the 3-Bark interval was not supported. The present paper analyzes the effect of different intervals on the center of gravity analysis described in Jaykin et al. [J. Acoust. Soc. Am. Suppl. 1 82, S81 (1988)] adapted from Chistovich and Chernova [Speech Commun. 5, 3–16 (1986)], and compares that analysis with a perceptual experiment testing the effect of harmonics on the perception of formants. The results support the hypothesis that the minimum interval is about 3 Bark. Our model may explain the lack of cue trading between formants and bandwidths.
Learning to identify phonemic orders
Authors: Brubaker Brad S. , Warren Richard M.
Published: October 2005

Several laboratories have reported that the order of items within repeating vowel sequences cannot be identified at item durations below 100 ms. This is puzzling, since phonemic sequences in speech consist of items that are considerably briefer. The present study demonstrates that listeners can, without feedback or knowledge of results, readily learn to name phonemic orders at item durations even briefer than those of normal speech. The listeners first identified different orders of components in repeating three-item vowel sequences at a few hundred ms per item (an easy task), and were then required to identify orders of these sequences at regularly decreasing item durations. By a series of successive generalizations to ever shorter items, order was named down to about two glottal pulses/vowel (the shortest duration used). Within the durational range of vowels in speech, our listeners distinguished different arrangements by their resemblances to particular words. Below this range, listeners employed qualitative differences of a nonverbal nature. Implications for theories of speech perception will be discussed. [Work supported by NIH.]
Rate-dependent perception of interpersonal characteristics
Authors: Feldstein Stanley , Dohm Faith-Anne , Crown Cynthia L.
Published: October 2005

The hypothesis, derived from the similarity-attraction literature, was that listeners describe speakers in more positive ways when they judge speakers' global speech rates to be similar to their own. Forty-five male and female listeners judged the global rates of three male and three female speech samples and how those rates compared with their own rates. The speaker of each sample was evaluated in terms of ten unipolar, adjective scales, each of which ranged from 0 to 9, with the higher score having the higher valence. The scores of the ten scales were then averaged to provide a total attribution valence score, and were also divided to provide “competence” and “affability” factor scores. The scores were subjected to appropriate regression analyses that included as independent variables speaker and listener gender and the perceived and actual differences between the speaker and listener rates. In support of the hypothesis, listeners assigned more positive total attribution values to those speakers whose rates were similar to their own, although their gender and the actual differences between their rates and the speakers' rates jointly influenced their evaluations of competence.
Identification of Polish words with non-neutralized word-final segments
Authors: Slowiaczek Louisa M. , Szymanska Helena
Published: October 2005

Research examining the phonetic characteristics of a number of neutralization rules has found that underlying contrasts that should be neutralized are phonetically preserved. In particular, earlier results [L. Slowiaczek and D. Dinnsen, J. Phonet. 13, 325–341 (1985)] found evidence that the rule of word-final devoicing in Polish is not neutralized. The present investigation extended these production results by testing whether the acoustic measures identified in productions from the original study are functional in perception. Native Polish and English listeners identified Polish monosyllabic words using a two-alternative forced-choice procedure. Results for the Polish subjects indicated a bias to choose the voiceless alternative and suggested that Polish listeners are unable to perceive differences in the minimal pairs examined in the production study. [Work supported by a small grant from Loyola University of Chicago.]