bluuebunny/crossref_metadata_2025 huggingface dataset:
Statistical energy analysis prediction of a liftoff vibration environment in six degrees of freedom
Authors: Powell Robert E.
Published: October 2005

The liftoff vibration environment for a guidance control sensor was predicted in three translational and three rotational degrees of freedom using statistical energy analysis (SEA) methods. Excitations from airborne sound pressure and mechanical transmission from solid rocket motor internal sources were applied to an SEA model of the vehicle and motor structure. Model predictions of translational response and wavenumbers for radial and inplane motions were combined to estimate rotational velocity environments. The rotational velocities from the vibroacoustic response represented out-of-band noise for the control system (that is trying to properly gimbal the motor nozzle) and required pre-flight estimation so that an appropriate filter could be provided.
What is a mode?
Authors: Barbone Paul E. , Crighton D. G.
Published: October 2005

The meaning of modes of vibration are discussed for elastic bodies submerged in an infinite acoustic fluid. In particular, it is shown that a submerged body has more modes than does the same body invacuo, that the individual modes of the fluid/solid system correspond generally to perturbations of either ‘‘fluid’’ modes or ‘‘solid’’ modes, and that the modes are not complete. Finally, it is shown that when the interaction between the wave types that comprise the different modes are strong, it may be impossible to identify a mode as originating with either the fluid or the solid. The results of this study have great general relevance, especially in the areas of statistical energy analysis (SEA), the singularity expansion method (SEM), and the resonance theory of scattering (RTS). [Work supported by ONR.]
Statistical energy analysis prediction of the interior low-frequency noise for a propeller-driven aircraft
Authors: Manning Jerome E.
Published: October 2005

Statistical energy analysis (SEA) is generally used to predict acoustic environments at high frequencies for random excitations. SEA can also be used to predict the environment at low frequencies for pure-tone excitations. The use of SEA for such problems requires consideration of both the mean and variance of the statistical prediction. Assuming a normal distribution of acoustic levels, 95% of measured data would be expected to fall within the limits set by the mean plus and minus two times the standard deviation. The use of SEA to predict both the mean and standard deviation of the cabin noise for a propeller-driven aircraft will be shown and comparisons with measured data will be presented. Methods to reduce the predicted variance will be discussed.
Application of statistical energy analysis for vibration and sound radiation prediction of multi-sectional cylindrical shells
Authors: Bernblit Michael V.
Published: October 2005

Statistical energy analysis (SEA) is a powerful tool for vibration and sound radiation prediction. The objective here is to evaluate acoustic power radiated by a large complex structure when its vibration is defined only within limited surface area in the vicinity of broadband sources. A tested structure is divided into a set of energy finite elements (EFEs) and the power balance matrix (PBM) equation is solved to obtain the ratio of EFEs rms normal velocity components. Acoustic power radiated by the whole structure is calculated as the sum of partial acoustic inputs of EFEs. Each power input is determined by radiation efficiency and surface averaged normal velocity of a structural element, enabling one to define generalized radiation efficiency of the total structure. A typical multi-sectional structure comprising cylindrical rib-stiffened shells with transverse partitions separating them was selected to utilize the developed approach. One of cylindrical subsets was driven by a normal broadband point force, and its vibration was measured in one-third octave bands and averaged over its surface area. Drops of surface averaged vibration levels between structural elements were computed using PBM equation. Radiation loss factors for ribbed cylindrical shells in air and water were investigated to obtain reliable data for PBM coefficients. Experimental data were in a good agreement with theoretical prediction.
Acoustic optimization of complex closed shells equipped with interior structures by numerical methods
Authors: Ionov A.
Published: October 2005

Complex closed shells divided into separate sections by partitions and decks with various kinds of interior equipment are used in the engineering practice quite frequently. The reduction of vibration and noise for these structures is very important at the stage of designing. The authors have developed an energy approach to an acoustic optimization of complex structures. It is based on a minimization of a total structure oscillation energy spectrum. It was indicated that the total oscillation energy estimates vibration and sound radiation of structures quite fully. As an example, the acoustic optimization of the multi-section cylindrical shell equipped with dynamically loaded interior solid bodies was produced with the help of FEM. The reduction of the total oscillation energy was received by changing of partitions stiffness, sizes of separate sections, and dislocations of interior solid bodies for a wide frequency range. The main rules of complex shell structure acoustic designing were formulated as the result of thorough computation investigations.
Reflection of a subsonic flexural wave on a plate at an air–water interface and far-field observations of the transition radiation in water
Authors: Matula Thomas J. , Marston Philip L.
Published: October 2005

An investigation of radiation by subsonic flexural plate waves due to a discontinuity in fluid loading is described. A tone burst of flexural waves propagates down a plate, the lower section of which is submerged in water. Measurements indicate that there occurs a branching of energy as the flexural wave passes through the air–water interface, with little reflected energy. A portion of the transmitted energy continues along the plate as a subsonic flexural wave with an associated evanescent wave. A second acoustic wave (which is referred to here as term transition radiation) originates at or near where the plate crosses the interface, and propagates in water to the far field. The near-field interference between the two acoustic waves in water results in a series of pressure nulls [T. J. Matula and P. L. Marston, J. Acoust. Soc. Am. 94, 1877 (A) (1993); T. J. Matula, Ph.D. dissertation (1993)]. Far-field observations of the transition-radiation angular pattern resemble that of a line quadrupole (approximating the farfield radiation due to a line moment on an uniformly loaded thin plate). The agreement between measured and modeled patterns is improved, however, by displacing the quadrupole horizontally behind the surface of the plate. [Work supported by ONR.]
Response of a nonbaffled sandwich panel submitted to a reverberant chamber acoustic environment
Authors: Nelisse Hugues , Beslin Olivier , Nicolas Jean
Published: October 2005

A method has been developed to obtain the structural response of a sandwich panel placed in a large reverberant chamber excited by a so-called diffuse field. The approach is based on an integral equation method for the pressure response in the chamber. The Helmotz–Kirchoff theorem is expressed using a pressure jump function across the surface of the panel [see for instance D. Ouellet etal., J. Acoust. Soc. Am. 89, 2131–2139 (1991)]. A modal expansion over modes of rigid cavity and invacuo modes of the panel leads to a set of linear coupled equations reflecting the existence of fluid–structure interaction. Numerical resolution of the set of equations allows us to obtain physical quantities of interest, such as the deflection of the plate and the jump of sound pressure across the panel. It is found that the modal behavior of the acoustic excitation is of a major importance for the response of the panel. Results for large panels with added masses, used for instance in aerospace industry, will be discussed. Convergence of the solution, as well as its frequency range of validity will also be analyzed.
Beam vibration studies using measured bending wave impedance
Authors: Kim S. K. , Prasad M. G.
Published: October 2005

Acoustic impedance in a plane-wave field is obtained experimentally using the well-known two-sensor technique. Recently, in the literature, the application of this technique to determine bending wave impedance has been demonstrated. In transverse vibration, the impedance is defined in terms of force and moment and the corresponding linear and angular velocities, respectively. The determination of bending wave impedance is based on the reflection coefficient which is obtained from the measured transfer function. Thus impedance evaluated using the measured transfer function and reflection coefficient is useful in characterizing the effects of boundary conditions, discontinuities, etc., in vibrating structures. In this study, bending wave impedance in a transversely vibrating cantilever beam is determined using the two-sensor technique. In addition to impedance, the measured transfer function and reflection coefficient are also used to describe the characteristics of a fixed end and a joint. The applications of impedance characterization to vibration analysis of connected structures are discussed. [Work supported by Daewoo Motor Company, Seoul, Korea.]
Evaluating varying configurations of constrained layer damping on a plate using structural intensity
Authors: Spalding Amy B. , Mann J. Adin
Published: October 2005

The use of structural intensity to evaluate patches of constrained layer damping attached to a vibrating plate will be discussed. Measurements with several aluminum plates were performed using various damping configurations. The plates were excited with a chirp signal (0–500 Hz) and results were analyzed between 0 and 500 Hz. Measurements were made before and after constrained layer damping material was applied to each plate. The damping configuration was varied on each plate to observe the affect of damping placement. Structural intensity for both the damped and undamped plates will be compared. Specifically, the torsional, bending, shearing, and total structural energy in the plates will be discussed. In addition, the energy absorbed by each damping configuration will be evaluated. [Work supported by NSF, MSS-9103377.]
Attenuation of edge reflections in plates and bars in flexure by a graded impedance interface
Authors: Vemula Chandrasekar , Norris Andrew N. , Cody George , Storch Gerry V.
Published: October 2005

Experimental results will be presented showing that it is possible to reduce reflections at the edges of plates and bars by using a graded impedance interface. A perfect impedance mismatch at the boundaries results in 100% reflection of waves, hence the motivation for using a graded impedance interface. An aluminum bar, a lucite bar (2 ft×1 in.×1 in.), and a viscoelastic strip (2 ft×1 in.×0.5 in.) are glued at the edges of a steel plate (2 ft×2 ft×1 in.), in order to achieve the graded impedance interface. The center point impulse excitation response is measured, and the peaks in the velocity response give a measure of the intensity of the reflected waves. A reflection coefficient R is defined based on a one-dimensional wave model. R is as low as 0.17 for the steel plate with the graded impedance interface mentioned above. Similar results are obtained for a steel bar, for which a value of R=0.31 is obtained by the same type of impedance grading. The reduction in R for both bars and plates is superior to that obtained using dry or wet sand as a vibration absorption mechanism.
On the relation between spectral discrimination of wideband and narrow-band sounds
Authors: Berg Bruce G. , Turner Matthew , Southworth Curt
Published: October 2005

The relation between listener’s abilities to discriminate spectral changes in wideband sounds and spectral changes in narrow-band sounds is examined. The standard in a 2IFC task consists of five, equal-intensity tones; the signal consists of an increment in the intensity of the central tone. Overall intensity is varied over a 20-dB range using a roving level procedure. Component frequencies are 200, 450, 1000, 2240, and 5000 Hz in the wideband condition, and 980, 990, 1000, 1010, and 1020 Hz in the narrow-band condition. Threshold estimates for 12 listeners show that performance levels in the two conditions are negatively correlated. Additional results suggest that listeners having the highest thresholds in the wideband condition extract envelope information over a greater range of frequencies than do listeners having the lowest thresholds. Theoretical implications concerning individual differences and an apparent trade-off in the ability to process acoustic information in temporal and spectral domains are discussed. [Work supported by ONR.]
Recognition of amplitude modulation patterns in the presence of a simultaneous distractor
Authors: Takeuchi Annie H. , Braida Louis D.
Published: October 2005

This experiment tested listener’s ability to compare the amplitude modulation pattern of 200- and 500-Hz signals when distractor signals that were also amplitude modulated were presented simultaneously. The amplitude envelopes of the distractor signals were either uncorrelated, partially correlated, or fully correlated with the amplitude envelopes of the comparison signals. Relative to the case of no distractor, performance tended to decrease when a distractor was present, and the degree of interference increased as the envelope correlations between the comparison and distractor signals decreased. Although the interference was greater when the comparison and distractor signals were separated by 50 Hz, there was also significant interference when the separation was 300 Hz. Whether the distractor was higher or lower in frequency than the comparison had no consistent effect on performance. Implications are discussed for the use of multiple band signals that carry the amplitude envelopes from different spectral regions of a speech signal to convey speech to severely hearing-impaired listeners. [Work supported by NIH.]
Discriminability of three-tone, common envelope signals
Authors: Anantharaman Jayanth , Feth Lawrence L. , Krishnamurthy Ashok K.
Published: October 2005

The IWAIF model [Anantharaman etal., J. Acoust. Soc. Am. 94, 723–729 (1993)] has been successfully used to explain the discriminability of two-tone complexes, called Voelcker pairs, which have the same envelope but differing fine structure [Feth etal., J. Acoust. Soc. Am. 72, 1403–1412 (1982)]. The addition of an equal amplitude third tone at the center frequency of the two-tone complex results in a three-tone complex pair which continue to have a common envelope. The IWAIF model predicts that the center tone acts as a masker, with the three-tone complex pairs being less discriminable than the corresponding two-tone complex pairs. Further the discriminability diminishes with increase in the level of the center tone. Thus to be equally discriminable, a larger frequency separation between the two extreme tones is needed for the three-tone pairs as compared to the corresponding two-tone complex. In this study the performance of listeners to such three-tone pairs at various center frequencies is investigated. The corresponding IWAIF predictions are compared with the subject data. [Work supported by AFOSR.]
Reproducible noise discrimination with concurrent narrowband noises
Authors: Sheft Stanley , Yost William A.
Published: October 2005

Reproducible noise discrimination ability was measured with a cued 2IFC procedure. Stimuli were either one, two, or three concurrent noise bands with bandwidths of either 12.5, 50, or 200 Hz and center frequencies of either 500, 1250, or 3125 Hz. In the multiband conditions, the concurrent noise samples either shared a common bandwidth and a common pattern of envelope fluctuation, had a common bandwidth with asynchronous envelopes, or had different bandwidths. Performance in the single versus multiband conditions was used to estimate cross-spectral correlations in the integration of envelope information. Single-band d′’s were inversely related to both bandwidth and center frequency. Performance in the multiband conditions was generally better than with the component bands presented individually. For subjects whose single-band d′’s were widely separated, greatest integration was obtained in the synchronous multiband conditions. Correlation estimates were significantly higher in the synchronous versus the two asynchronous condition sets. In all conditions, estimated correlation was inversely related to bandwidth. Results are used to model internal noise levels in cross-spectral envelope processing. [Work supported by NIH.]
Effect of coherent frequency modulation on grouping the harmonics of a vowel
Authors: Darwin C. J. , Sandell Gregory
Published: October 2005

When a single harmonic close to the first formant frequency is mistuned by about 8%, that harmonic makes a reduced contribution to the vowel’s first formant frequency as measured by a shift in the phoneme boundary along an F1 continuum between /I/ and /e/ [C. J. Darwin and R. B. Gardner, J. Acoust. Soc. Am. 79, 838–845 (1986)]. In the present experiment phoneme boundaries along an /I/-/e/ continuum were measured for vowels differing in F1 whose fourth harmonic (500 Hz) was mistuned by 0, ±3, ±6 or ±9%. All the harmonics of a vowel (including the mistuned one) were given either no FM or coherent FM at a rate of 6 Hz and modulation depth ±5%. The results replicated the previous findings, but found little evidence for coherent FM preventing the segregation of the mistuned harmonic from the vowel. [Work supported by SERC.]
Preliminary measures of the focusing of attention on auditory streams
Authors: Botte Marie-Claire , Drake Carolyn , Brochard Renaud , McAdams Stephen , Argenti Anne-Marie
Published: October 2005

The focusing of attention on an auditory stream was studied by asking subjects to detect a temporal irregularity in the interonset interval between two successive tones of a regular sequence in a 2AFC procedure. Eight subjects heard the sequence with the irregularity presented concomitantly with two other regular sequences, each sequence having its own specific tempo, number, and frequency of tones. One of these sequences, presented alone as a cue for the target, preceded two observation intervals containing the three concurrent sequences. The size of the irregularity detected at a 90% level in a target sequence was only detected at chance level in both nontarget sequences. Increasing the level of a probe sequence above the level of both the target and the other nontarget sequences resulted in a significant improvement in detection for five subjects. Detection performance in the probe sequence was equivalent to that in the target sequence when the probe level was increased by 18 dB. For the three other subjects, detection in the probe sequence remained at chance, whatever its level. Therefore, the ‘‘attenuation’’ of unattended auditory streams can attain as much as 18 dB for some subjects, probably being higher for others.
Iterated rippled noise: Testing temporal theories of complex pitch
Authors: Yost William A. , Sheft Stanley , Patterson Roy
Published: October 2005

Two versions of iterated rippled noise were generated by cascading a delay (T), attenuate (g), and add network n times. In one version, the new sum was delayed, attenuated, and added to the original noise iteratively. In the other version, the new sum was delayed, attenuated, and added back to itself iteratively. Pitch varies with T; and pitch strength varies with g and n, increasing as g decreases and n increases. Listeners discriminated between two iterated rippled noises based on pitch strength, and performance was measured as a function of T, n, and g. The data were accounted for by peaks in the autocorrelation functions. Listeners were often at or near chance in discriminating one noise from another when their spectra were different, but the autocorrelation functions shared some similarities. The auditory image model [R. D. Patterson etal., in AuditoryPhysiologyandPerception, edited by Y. Casals, L. Demany, and K. Hoerner (Pergamon, Oxford, 1992)] performs an operation similar to autocorrelation, and this temporal model accounts for the pitch and pitch strength of rippled noise and other versions of complex pitch. [Work supported by NIDCD and DRA Farnborough.]
The influence of temporal deviations on the perception of auditory pattern components
Authors: Kidd Gary R.
Published: October 2005

Previous studies of the effect of deviations from pattern-based temporal expectations revealed increases in detection and discrimination thresholds for temporally displaced components in sequences of tones as the amount of displacement was increased [G. R. Kidd, J. Acoust. Soc. Am. 93, 2315(A) (1993)]. Further studies have examined a variety of temporal contexts to determine the conditions necessary for the establishment of an effect of deviation from temporal expectations. Expectations were induced by repeated presentations of a standard pattern on each trial. Temporal deviations were introduced in comparison patterns by advancing or delaying the onset of a single ‘‘target’’ tone while maintaining the temporal relations among the context tones. The effect of temporal deviations was found to be considerably more robust for frequency discrimination than for detection. Frequency discrimination was consistently more sensitive to small temporal displacements and large effects were obtained in rhythmic contexts that produced little or no effect on detection. The results suggest a rhythmic heightening of attention, based on pattern timing, that enhances resolving power for events within a fairly narrow temporal window. Alternative explanations for the greater sensitivity of frequency discrimination to temporal deviations will be evaluated. [Work supported by NIH.]
The measure and modelization of tempo sensitivity in auditory sequences
Authors: Drake Carolyn , Botte Marie-Claire
Published: October 2005

The ability of listeners to detect slight changes in the tempo of isochronous auditory sequences was investigated by measuring differential thresholds for tempi of sequences with interonset intervals (IOI) ranging from 50 to 1500 ms. The number of intervals they contained was varied systematically. A multiple-look model presented previously predicts that sensitivity increases as the number of intervals in the sequence increases, since multiple observations afford the elaboration of a more precise mental representation involving the coding of the average duration and degree of dispersion of the intervals in the sequence [C. Drake and M.-C. Botte, Percept. Psychophys. 54, 277–286 (1993)]. The results of the present study provide additional support for the multiple-look model and suggest that another parameter, that of the total duration of the sequence, should be incorporated. The multiple-look strategy can only function efficiently within a limited ‘‘temporal window’’ which depends on the tempo of the sequence. The upper limit of this temporal window is about 1 s for sequences of 50- to 300-ms IOI and about 2.5 s for sequences of 500- to 1500-ms IOI.
Auditory discrimination based on the physical dynamics of a tuning fork
Authors: Lutfi Robert A. , Oh Eunmi
Published: October 2005

A method is described for measuring the strategies listeners employ in the determination of simple resonant sources. To demonstrate this method the principles of theoretical acoustics were applied to reconstruct the sound-pressure waveform at the ear as would be generated by a common tuning fork. The result is an inharmonic sum of damped sinusoids whose acoustic parameters covary in lawful relation to the physical dynamics of the source. Listeners were asked to discriminate these tuning fork sounds from a set of nontuning fork sounds for which the lawful covariation among parameters was randomly perturbed from trial to trial. The lawful covariation had to be detected to perform significantly above chance. Finally, individual correlations were computed between the listener’s response and the values of the acoustic parameters across trials to determine precisely which aspects of lawful covariation listeners use to identify the source. Discrimination performances was found to be quite good for a tuning fork whose physical dimensions remained constant across trials (driving force varies), though it was exceedingly poor when just one physical dimension was allowed to vary from trial to trial. Individual correlations revealed that the covariation exerting the greatest influence often involved the one spectral component having the most acoustic energy in the signal. [Research supported by NIDCD.]
A 350-m depth capable, air-compensated baffled transponder for the deep water portable tracking system
Authors: Menoche R. K. , Reid R. J. , Stein P. J.
Published: October 2005

Here we describe a novel baffled central up-link subsurface link (SSL) for the deep water portable tracking system. The central SSL is deployed at depths up to 350 m and collects the acoustic information from bottom moored tracking transponders. These transponders can be deployed to depths up to 6000 m. Degradation of the up-link SSL performance is generally due to support ship radiated noise and surface generated ambient noise. The sources of interference can be suppressed with a downward looking receiver. This receiver must have a wide beamwidth (140 deg) and its response must fall off rapidly outside the beam to reject ambient noise. The device built consists of a hydrophone and projector mounted on the face of a steel signal conditioning plate (SCP). The plate is designed to have low vibration sensitivity and edge diffraction. Behind the plate, and extending beyond the SCP edges is a fiberglass air chamber. To prevent crushing, it is fed by an air-compensation bag. Extension of the air-backing beyond the SCP is important to ambient noise rejection. This soft surface helps to reject noise which diffracts around the edge of the baffle. The air-chamber design was chosen over material solutions because it was uncertain if, at the deeper depths, the latter could provide the pressure release surface required. Both theoretical and experimental results of baffle performance are presented.
Sharpness: A perceptually based measure of the spectral dimension of musical timbre
Authors: Goad Pamela
Published: October 2005

In order to assess the perceptual relationship between musical timbre and its physical properties, an effective measure should combine those factors considered important to perception with the common attributes of musical timbres. Sharpness is a perceptually based measure of timbre originally tested and put forth by Bismarck [Acustica 30, 146–172 (1974)] and later refined by Aures [Acustica 59, 130–141 (1985)] which applies a weighting to higher frequencies of a loudness pattern. Two-octave ranges from the brass, clarinet, saxophone, double reed, and string families of musical instrument timbres were analyzed across a musical dynamic range of pianissimo to fortissimo. Particular attention was paid to similarities in sharpness in three categories: Musical instrument families, instruments of like voice, and performance comparisons on the same instrument with different performers. Evidence suggests that most similarities in sharpness occur at the lower playing levels in every category. In addition, sharpness proved to be sensitive to idiosyncrasies of instrumental design and performer-specific techniques. Some comparisons to spectral centroids and centroids of loudness patterns [Grey and Gordon, J. Acoust. Soc. Am. 63, 1493–1500 (1978)] will be presented in addition to implications for application to subjective responses of musical blend.
Residual minimization in a musical signal model based on a deterministic plus stochastic decomposition
Authors: Serra Xavier
Published: October 2005

Spectral modeling synthesis (SMS) is a spectrum modeling technique that models time-varying spectra as (1) a collection of sinusoids controlled through time by piecewise linear amplitude and frequency envelopes (the ‘‘deterministic’’ part), and (2) a time-varying filtered noise component (the ‘‘stochastic’’ part). This technique has proved to give general, high quality transformations for a wide variety of musical signals. As part of the analysis process of SMS the deterministic component is subtracted in the time domain from the original sound resulting into a residual signal. For a good deterministic-stochastic decomposition this residual should be free of the partials of the analyzed sound and a minimization criteria of the residual can be establish to measure the quality of the decomposition. There are many control parameters in the analysis process and their impact in the synthesized sound can be measured numerically by studying their effect on the residual. This paper will discuss the effect that parameters like size and type of analysis window, hop size, or FFT size have on the residual and analytical ways to get the best analysis for a given sound.
Time-scale modification of complex acoustic signals
Authors: Quatieri T. F. , Dunn R. B. , McAulay R. J. , Hanna T. E.
Published: October 2005

A new approach is introduced for time-scale modification of short-duration complex acoustic signals. The method preserves an approximate time-scaled temporal envelope of the signal, thus capitalizing on the perceptual importance of the signal’s temporal structure, while also maintaining its spectral character. The basis for the approach is a subband signal representation, derived from a filterbank analysis/synthesis, whose channel phases are controlled to shape the temporal envelope of the time-scaled signal. To account for a stochastic element in the signal, channel amplitudes and filterbank inputs are selected to shape the spectrum and correlation of the time-scaled noise component. The phase, amplitude, and input control are derived from locations of events which occur within filterbank outputs. A frame-based generalization of the method imposes phase consistency and noise continuity across consecutive synthesis frames. The approach is applied to synthetic and actual complex acoustic signals consisting of closely spaced sequential time components and is applicable to music, biological, and underwater sound processing.
Classification of music type by a multilayer neural network
Authors: Matityaho Benyamin , Furst Miriam
Published: October 2005

Western people who are exposed to different types music, such as pop or classic, can easily distinguish between them after listening to any composition for a few seconds. It is still unclear what the features are in the music that allow people this quick recognition. In the present study a model is proposed that distinguishes between two classes of music, pop and classic. The model is a decision making system that was implemented by integrating outputs of a multilayer neural network (NN). The input nodes to the NN were obtained by a preprocessing algorithm that included the following steps: (1) Dividing the musical composition to intervals of 16.8 ms; (2) applying spectral analysis on each interval; (3) combining the spectral components of T successive intervals into F divisions, which were obtained by dividing the logarithmic of the audiometric frequency range into equal F parts. As a result of the preprocessing algorithm interval of T*16.8 ms of musical signal was presented by F*T input values to the NN. Our results show that it is enough to train the NN with a group of short intervals (i.e., 0.3 s), and to represent its spectrum by about 20 values in order to obtain a 100% success in distinguishing between two types of music.
Pitch and timbre influences on the short-term memory for timbre
Authors: Starr Gary E.
Published: October 2005

Memory for pitch is susceptible to same-dimension interference as a function of pitch distance, while remaining relatively free from cross-dimensional (e.g., timbre) interference [C. Semal and L. Demany, J. Acoust. Soc. Am. 89, 2404–2410 (1991)]. This study investigated whether analogous effects occur in memory for timbre. Subjects discriminated the timbre of a standard and a comparison tone. The tones were separated by a 5-s retention interval that was filled with silence or a six-tone sequence that jdiffered from the standard in pitch, timbre, or both. There were two levels of each variable, perceptually close or far to the standard. Results indicated that timbre is qualitatively different from pitch, since any sequence, independent of same-dimension similarity, produced interference.
Wave propagation with local conservation laws in discrete Space-Time
Authors: Nagem Raymond J.
Published: October 2005

The equations of three-dimensional linear acoustics are formulated using a complex state vector and a first-order Lagrangian. Local conservation equations for energy, linear momentum, and angular momentum are derived in a unified and systematic way from gauge transformations. The first-order Lagrangian description is shown to lead naturally to a set of discrete space and discrete time field equations. Gauge transformations are again used to derive local conservation equations for the discrete field variables. Dispersion relations for the discrete equations are derived, and are used to analyze the spatial isotropy of propagating waves on the discrete space-time lattice. The solution of the discrete initial value problem is discussed, as well as its implementation on a massively parallel computer.
Acoustic pulse effects for media obeying a frequency power law absorption
Authors: Szabo Thomas L.
Published: October 2005

The effects of phase velocity dispersion accompanying attenuation can alter both the shape and time delay of finite length acoustic pulses. For attenuation described by a frequency power law with an exponent y, a time causal theory [T. L. Szabo, J. Acoust. Soc. Am. 93, 2277 (A) (1993)] has shown that dispersion is greatest [in the range (0≤y≤2) when y=1 and is zero for y=0 and y=2. These results are in contrast to the approximate nearly local Kramers–Kronig theory [M. O’Donnell etal., J. Acoust. Soc. Am. 69, 696–701 (1981)] which predicts increasing dispersion as y increases from 0 to 2. The consequences of dispersion from the two theories are compared for simulated finite-length pulses for different values of y. For values of y=1, where the theories agree, the effects of bandwidth on pulse shape and delay are illustrated. An approximate general scaling law for pulses propagating in lossy media is presented.
Measurements of the modal decay rates of a gas/particle suspension confined in a cylindrical tube: Relationship to particle concentration
Authors: Moss Simon H. , Attenborough Keith
Published: October 2005

A novel method to measure the particle concentration in a gaseous suspension has been investigated. The measured impulse response across a section of cylindrical tube containing the suspension has been processed to determine the decay rates of radial modes in the tube. These decay rates are sensitive to the attenuation coefficient of the suspension, and hence to the particle concentration. The decay rates observed with and without suspended particles have been used to establish an empirical relationship between the particle concentration and the model decay rates. The frequency range used was 1.5–15 kHz, and the particle volume concentration has been varied between 0.7–10 ppm. The mean diameter of the particles was 6 μm. [Work supported by the S.E.R.C.]
Influence of finite impedance walls on the sound field in an enclosure
Authors: Chandra Kavitha , Thompson Charles , Manley Martin G.
Published: October 2005

A numerical technique is presented for computing the pressure distribution in the interior of a rectangular enclosure. The analysis models the effect of a nonuniform distribution of wall impedance. The normal modes for this problem are evaluated by solving Green’s integral equation. The wall admittance function is represented as the sum of a mean value superimposed by spatially varying fluctuations. An asymptotic series solution for the pressure about the mean value of the wall admittance is not guaranteed convergence for functions that exhibit large perturbations from the mean. The convergence of the series is shown to be improved by transforming the pressure to a rational functional representation. The superposition of normal modes determined by this technique serves as Green’s function that allows the acoustics of the enclosure to be modeled for a variety of sources. The effect of the wall impedance on the transient response and the reverberation time of the room is presented. [Work supported in part by DOE.]
Quo Vadis: whither goest thou
Authors: Gough Peter T. , Hayes Michael P.
Published: October 2005

Most synthetic aperture (side scan) sonars, SAS, assume that the towfish travels along a straight track. Unfortunately, even a sub-wavelength sideways movement (sway) causes significant errors. For SAS to reach its potential, methods need to be found to monitor and correct these errors. Three techniques exist to estimate the unknown movements of the towfish. The first is to use precise navigation instruments, the second is to correct the image using point-spread-variant image enhancement—relying on dominant point reflectors in the object field. The third is the topic of the simulations reported in this paper. This technique makes one assumption: that in the absence of any dominant echoes, the only parameter common to all ranges is the movement of the towfish. Thus after all the range-gated echoes have been averaged in some fashion, any residual bias is due to the movement of the towfish. Here, a series of simulated trials to test this hypothesis is reported. It is shown this technique fails when a dominant reflector is introduced in the clutter-limited object field. The conclusion is that a combination of techniques two and three should enhance the quality of SAS image reconstruction under most realizable situations.
Compensation of medium-induced fluctuations for synthetic aperture sonar
Authors: Huxtable Bartou D.
Published: October 2005

There are two primary obstacles to achieving aperture coherence in synthetic aperture sonar (SAS); motion compensation (MOCOMP) accuracy, and medium fluctuations along the acoustic propagation paths. The feasibility of MOCOMP for high-resolution SAS using a combination of commercial MOCOMP sensor components with existing autofocus processing algorithms has been previously assessed. This paper discusses the experimental measurement of acoustic fluctuations induced by inhomogeneities in sea water, their impact on SAS imagery, and whether autofocusing algorithms can compensate for medium-induced fluctuations. Spatial as well as temporal fluctuations over propagation path lengths of 1000 ft and frequencies of 100 kHz are relevant to high resolution SAS. A historical survey discovered no data in this parameter regime. Therefore, a joint IR&D effort, the Acoustic Medium Stability Experiment (AMSE), was conducted to collect appropriate data. AMSE’s unique data set provides space-time statistical descriptions of the medium-induced phase fluctuations. Using the AMSE data and an end-to-end SAS simulation capability, it was found that medium-induced fluctuations do degrade SAS imagery significantly. However, available autofocusing algorithms are capable of estimating and compensating the medium-induced phase errors (in addition to MOCOMP errors), allowing fully focused imagery to be obtained.
Synthetic aperture experiment in the Hudson Canyon area
Authors: Doutt James A. , Rajan Subramaniam D. , Carey William M.
Published: October 2005

In 1988, narrow-band sound propagation measurements were performed in 73 m of water on the New Jersey Continental Shelf over a frequency range from 50 to 600 Hz. A mid-depth source was towed past a 24-element hydrophone array out to ranges of approximately 4 km. Precision time bases controlled the source and were recorded on one of the data channels. A Del Norte radar system measured the range to within several meters accuracy. Plots of the residual phase of the sound pressure as a function of range demonstrate the high quality of the data. A synthetic aperture technique was used to produce horizontal wave-number spectra at each of the hydrophone depths. Plots of the magnitude of the horizontal wave number as a function of depth and wave number clearly demonstrate the modal nature of the propagation. The sound speed in the water column was calculated from CTDs taken during the experiment, and the second speed in the bottom was derived using an acoustical inversion technique. The measured acoustical data along with the measured and derived environmental information provide a unique data set for testing processing and modeling algorithms. [Work supported by ONR.]
High-resolution sub-bottom profiling and reverberation mapping using reflection seismology and synthetic aperture techniques
Authors: Rogers Andrew K. , Yamamoto Tokuo
Published: October 2005

Recent experiments in the Atlantic margin have measured the acoustic response of the shallow water environment. A near-surface orthogonal towed array was used to record the response of the environment to broadband (75 Hz to 1.6 kHz) acoustic excitations produced by a piezoelectric source at timed intervals. Approximately 30 km of data were taken. Reflections from the bottom and subbottom layering have been measured yielding information about the layering and sediment properties of the area. The bistatic configuration of the source receiver and source firing position pairs invited the investigation of the reverberation or scattering field over a large coverage area using beamform and aperture enhancement processing. Array dimensions have been extended to 24×24 element and may be increased further. The effective large apertures provide array gain and beam focusing over a 360°×180° region. The combination of synthetic aperture processing and the three-dimensional characteristics of the experimental setup provide excellent resolution and survey rate. Both the water column and subseafloor sediment regions are investigated. Results of the processing of the measured data are presented. [Work supported by ONR.]
Passive synthetic aperture processing as a Kalman filter problem
Authors: Sullivan Edmund J. , Candy James V.
Published: October 2005

Passive synthetic aperture processing of a towed array is basically the coherent processing of the array data over a given length of time which, based on the speed of tow, translates into an equivalent increase in aperture length, and therefore a concomitant increase in spatial gain. The Kalman filter formalism is particularly suited to this task for four reasons. First, it coherently updates the measurements based on a comparison of their predicted values, based on a signal model, and subsequent measurements taken at a later time, in a recursive manner. Second, it allows in principle any signal model, i.e., not simply plane waves. Third, it generates a minimum variance estimate not only on the desired bearings, but any desired (observable) model parameters. Finally, it avoids the explicit construct of a beamformer, thus permitting the precision of the estimates to have no lower limit (such as the spatial bin size of a beamformer) other than the minimum variance obtainable under the given signal to noise conditions. Samples of multiple bearing estimations based on synthetic data will be shown. Also, a general formalism will be presented that allows direct estimation and update of certain signal model parameters.
Using a spatially referenced towed array for estimating the bearings of narrow-band acoustic sources
Authors: Edelson Geoffrey S. , Tufts Donald W.
Published: October 2005

Insights, simplified formulas, and figures are provided to aid in the design of unconventional towed array systems; specifically, for the case of a spatially referenced towed array (SPARTA). The SPARTA consists of two subarrays, one of which is moving and one of which is fixed in position. The estimated bearing of an acoustic source and the resolution in bearing of multiple acoustic sources can be improved if a SPARTA is used instead of a conventional towed array (CTA) or a fixed-position array (FPA). The CRLB provides a framework within which different array configurations can be compared in order to point the way toward better design possibilities. The capabilities attributable to the SPARTA are emphasized because of its overall potential to provide lower variance on bearing and bearing resolution than a FPA or CTA. A summary is given of the varying abilities of towed array configurations to estimate bearing, determine when the use of a SPARTA is desirable from the perspective of accurately estimating bearing, and help to build an understanding of design principles for passive towed line array systems. Important properties are derived, which characterize the ability of these configurations, with appropriate signal processing, to jointly estimate the bearing and frequency of each source.
Implementation of a synthetic aperture processing scheme in a towed array sonar system
Authors: Stergiopoulos S. , Trider R. C.
Published: October 2005

This experimental paper is concerned with the implementation of a passive synthetic aperture processing scheme in a towed array sonar system. It addresses theoretical and practical concerns, and the experimental results suggest that the concept of a synthetic aperture towed array system complements the conventional beamformer’s processing results. The acoustic synthetic aperture technique, which has been considered in this study, is the ETAM algorithm [i.e., extended towed array measurements, J. Acoust. Soc. Am. 87, 158–171 (1989)]. Theoretical and experimental studies have shown that the above technique compensates the random phase fluctuations in the data that are introduced by the tow path. This results is an effective extension of the physical aperture of a deployed towed array and consequently provides improved angular resolution performance equivalent to a fully populated long array. Also included in this presentation is a discussion of the development of the computing architecture requirements related to this kind of implementation. It is shown that the development of a next generation signal processor (NGSP) by the Defence Research Establishment Atlantic (DREA) overcomes the limitations normally associated with the implementation of nonlinear algorithms into parallel FFT-based vector processors.
Multipath signal modeling for electroacoustic transducer calibration in reverberant underwater environments
Authors: Ainsleigh Phillip L. , George James D.
Published: October 2005

A new method was developed for modeling acoustic signals in reverberant environments for application to underwater electroacoustic transducer calibration. Specifically, a multipath exponential model was used to represent the response signals encountered during stepped-sinusoid testing of transducers in bounded test facilities. In these tests, the received signal is the hydrophone’s response to the acoustic wave emitted by the projector and traversing both the direct and the reflected propagation paths from the projector to the hydrophone. By modeling the response to both the direct and reflected waves, the desired echo-free steady-state response can be determined from the estimated model parameters corresponding to the earliest arrival time. The method was applied to calibrating a low-frequency high-power tonpilz projector in a small pressurized tank where the echo-free time is shorter than the transient response of the projector. The estimated results agree very closely with measurements made in a large lake where the echo-free steady state can be observed directly. [Work supported by ONR.]