摘要:

The coupling loss factors are important in SEA (statistical energy analysis) for predicting vibrational energy distributions and power flows between coupled structures. The traditional empirical and analytical methods for determining these parameters were either for high‐frequency range or for coupled structures with very simple geometric shapes. Analytical results are presented in this paper, which can be used to determine the energy distributions and power flow between coupled structures and so the coupling loss factors at any frequency and for any structure shape. The numerical tests for the two coupled plate system is conducted by using the analytical results, and are compared with the SEA predictions and previous experimental results in the high‐frequency range.

ISSN:0001-4966    

DOI:10.1121/1.2028911

出版商:Acoustical Society of America    

年代:1990

数据来源: AIP

摘要:

Foamed aluminum is a promising candidate in hydroacoustics for pressure‐release material in the dry state, and for sound absorption, when saturated with a viscous liquid. Previously the dynamic shear modulus measurement of this material has been reported, as a function of density and pore size [Proc. IEEE 1985 Ultrasonics Symp.2, 1052–1055 (1985)]. To characterize the elastic properties, at least one other modulus measurement is required. It is difficult to conceive how the bulk modulus could be determined. The dynamic Young's modulus may be found by measuring the extension of a beam by means of accelerometers. In the present experiment the dynamic Poisson's ratio and Young's modulus of the same sample were determined by the use of laser Doppler velocimetry. The results are given for foamed aluminum of varying porosity and pore size, and the consistency with measurements of the shear modulus is discussed.

ISSN:0001-4966    

DOI:10.1121/1.2028913

出版商:Acoustical Society of America    

年代:1990

数据来源: AIP

摘要:

The use of the cochleagram and the correlogram in speech and sound recognition is discussed. The cochleagram represents a sound as a pattern of neural firing probabilities at places along the basilar membrane versus time. It is roughly analogous to the spectrogram and its benefits have been described in several papers. But using the cochleagram as a basis for speech recognition is only a weak way to use knowledge of human auditory processing A richer representation of speech and sound is called the correlogram. Sound is represented as a two‐dimensional picture versus time—the extra dimension allowing several interesting perceptual experiences to be modeled. Assembling correlograms into a movie and synchronizing them to sound allow the auditory and visual percepts to be compared. The correlogram is more useful than a cochleagram (or spectrogram) because it shows an orthogonal dimension that represents the fine‐time structure and pitch in the auditory signal. The extra dimension provides the information necessary for auditory grouping and speaker separation. This talk will emphasize the advantages of a two‐dimensional representation of sound and describe several auditory maps that might be used by the brain to do auditory scene analysis. Videotapes will be shown to demonstrate the advantages of the correlogram representation.

ISSN:0001-4966    

DOI:10.1121/1.2028916

出版商:Acoustical Society of America    

年代:1990

数据来源: AIP

摘要:

The range of sounds that are exploited phonetically by the world's languages represents only a portion of what humans are capable of producing vocally. Moreover, among attested phonetic segments there is enormous variation in the frequency of occurrence across languages: Most segments are relatively rare, while a few occur almost universally. A major task of phonetic theory is to explain these patterns of selection. Traditionally, many phoneticians have believed that two principles—articulatory economy and perceptual distinctiveness—play a role in shaping sound patterns and segment inventories. However, these principles have not often been formulated with sufficient precision to have genuine explanatory content. The focus of this presentation is on the role of auditory factors in structuring vowel systems. First, attempts to predict vowel inventories on the basis of a principle of auditory dispersion (i.e., sufficient auditory contrast) are reviewed. Second, a corollary of the dispersion principle, the auditory enhancement hypothesis, that provides a general account of certain widespread patterns of phonetic covariation in the production of vowels is explored. Finally, how the notion of sufficient contrast may explain some puzzling acoustic differences between male and female tokens is considered.

ISSN:0001-4966    

DOI:10.1121/1.2028918

出版商:Acoustical Society of America    

年代:1990

数据来源: AIP

摘要:

The spectral acoustics of the sounds of speech as measured upon production are well known. But the spectral features of these sounds as they exist in the ear canal immediately prior to auditory reception are somewhat different from those typically cited, due most likely to the resonance characteristics of the external meatus. A recent pilot study using probe microphonics and spectral analysis revealed that the formant structures of the vowel sounds /i, u, ɜ/ as recorded in the ear canal were similar to those recorded outside of the ear, but the amplitude of the second formant was reduced in each case. These findings have potential implications for the importance of the second formant in the perception of these and other vowel sounds. The spectral structure of more vowels, consonants, and sounds in speech are yet to be explored when recording is performed from inside the external ear canal.

ISSN:0001-4966    

DOI:10.1121/1.2028921

出版商:Acoustical Society of America    

年代:1990

数据来源: AIP

摘要:

The perceptual segregation of simultaneous speech is known to benefit from differences in pitch or loudness between the speech sounds. It is likely that different cues operate on different time courses. A series of experiments examined subjects' ability to identify both members of simultaneously presented steady‐state vowel pairs. The duration of each vowel pair varied from 1 to 8 cycles. A “dominant” vowel was identified with near 100% accuracy even in the single‐cycle vowel pair condition (one pitch period), whereas the recognition of the “nondominant” vowel showed a slow improvement up to the 8‐cycle condition. Asynchrony between the pitch periods of the vowels improved performance when they had a low fundamental frequency (80 Hz), but about 3 cycles of the vowels were necessary for this advantage. The effects of changing the amplitude of one vowel relative to the other and the effects of differences in frequency modulation were also studied. The results support previous research in showing that phonetic decisions “await” organizing mechanisms that operate in the auditory periphery. [Work supported by Esprit BRA 3207.]

ISSN:0001-4966    

DOI:10.1121/1.2028927

出版商:Acoustical Society of America    

年代:1990

数据来源: AIP

摘要:

Volume acoustic scattering by internal waves causes a mismatch between the true environment and the idealized environment assumed in matched‐field processing. To assess the effect of internal waves, the performance of various matched‐field algorithms has been tested using simulations based upon the parabolic approximation. Internal wave and sound‐speed parameters appropriate to the deep ocean have been used. Using a combination of theory and simulated results, the dependence of processor performance on range, frequency, and aperture size is discussed. [Work supported by ONR.]

ISSN:0001-4966    

DOI:10.1121/1.2028932

出版商:Acoustical Society of America    

年代:1990

数据来源: AIP

摘要:

Matched‐field processing can be used to locate a sound source in the ocean by finding the best correlation between the signals received by a set of hydrophones and expected signals that would be received if the sound source were at a particular location. When standard (Bartlett) correlations are used, the sidelobes of a strong source may obscure the main response of a weaker source. A simple palliative for this problem is to calculate the elements of the cross‐spectral matrix associated with the strongest signal and subtract these from the observed cross‐spectral matrix. This process can be repeated for the next strongest signal, etc. This method has been suggested before for use with conventional plane‐wave beamforming [H. P. Bucker, J. Acoust. Soc. Am.62, 1222–1225 (1977)]. In this report, the sidelobe reduction is quantified as function of mismatch in the environmental parameters and the proposed subtraction method is compared to high‐resolution adaptive matched‐field algorithms.

ISSN:0001-4966    

DOI:10.1121/1.2028936

出版商:Acoustical Society of America    

年代:1990

数据来源: AIP

摘要:

Time‐domain beamforming has been formulated as an optimization problem with simulated annealing applied to search the large parameter space of source bearings and time series [Kupermanet al., “Optimal time‐domain beamforming with simulated annealing including application ofa prioriinformation,” J. Acoust. Soc. Am. (in press)]. Both the generality and the efficiency of this signal processing algorithm have been improved. The time‐domain beamformer was originally applied to problems involving waveforms of compact support. For more general types of signals and multiple sources, an additional constraint of minimizing the correlation between signals is required to suppress ambiguous periodic functions in the time series (each of which is common to signal pairs). The efficiency of the simulated annealing algorithm for time‐domain beamforming has been improved using the concept of latent heat release at phase transition. Simulated annealing is analogous to the cooling of a liquid to form a perfect crystal and involves artificial quantities analogous to temperature and energy. With an adaptive latent‐heat cooling schedule in which the temperature depends on the energy, the system has a greater tendency to remain in low‐energy states. The convergence rate of the optimization procedure is thus significantly increased.

ISSN:0001-4966    

DOI:10.1121/1.2028944

出版商:Acoustical Society of America    

年代:1990

数据来源: AIP

摘要:

Synthetic aperture sonars (SAS) or radars (SAR) that transmit narrow‐band signals are sampled arrays and therefore must satisfy sampling requirements to avoid aliasing. The sampling requirements for a narrow‐band SAS or SAR may be relaxed if a broadband signal is used. From a beamforming view, broadband transmit signals result in a mainlobe that always points in the same direction and has a width that varies as the transmit signal sweeps through the broadband, but having aliasing lobes that change in both width and direction. A composite broadband beam pattern would show constructive mainlobes but smeared aliasing lobes. Broadband undersampled synthetic aperture images similarly show a target image having the samex–yimage space position for all frequencies, but alias target images that have azimuth positions that vary with frequency. A synthetic aperture image uses the entire transmit band, and so, targets appear sharp and aliases become smeared. An approximate relation for the level of smearing of the alias targets may be shown, which depends on the signal bandwidth, the platform speed, the sonar along‐track length, center frequency, and the pulse rate. This allows the design of broadband undersampled synthetic aperture sonars with higher speeds and greater mapping rates. [Work supported by the Charles Stark Draper Laboratory, Cambridge, MA.]

ISSN:0001-4966    

DOI:10.1121/1.2028949

出版商:Acoustical Society of America    

年代:1990

数据来源: AIP