bluuebunny/crossref_metadata_2025 huggingface dataset:
Interfamily timbral differences: A comparison of trombone and flute tone spectra
Authors: Beauchamp James W.
Published: October 2005

Time-averaged spectra for tones of the trombone and flute performed at the same pitch can be quite similar [D. A. Luce, J. Audio Eng. Soc. 23, 565–568 (1975)], and yet the tones sound quite different. Besides factors of attack rate and noise content, strong differences between the two types of sounds can be observed due to the manner in which partial amplitudes change as a function of overall (rms) amplitude. As rms amplitude increases, trombone partial amplitudes increase smoothly, resulting in an increasing spectral centroid, and as rms amplitude decreases, they follow approximately the same curves. On the other hand, amplitudes of flute partials tend to “turn on” at certain thresholds, depending on performed attack and overall amplitude, and follow different curves during ascending and descending portions of the tone. Using analysis and synthesis techniques, it is possible to create hybrid tones that blend some of the characteristics of the flute with those of the trombone, in order to evaluate the perceptual importance of distinctive parameters.
A comparison of musicians' preferences and acoustic measurements of different French horn mouthpieces
Authors: Plitnik George R. , Lawson Bruce A.
Published: October 2005

Impedance measurements were made for a selection of commercially sold mouthpieces attached to a test French horn. The resultant curves were analyzed for possible correlations with musicians' preferences as obtained from playing tests. It was originally believed that intonation is an important criterion for a good brass wind instruments and its mouthpiece, but it was not clearly evident from the data. Other parameters of the response curves analyzed included impedance peak amplitudes and Q ratios.
Wavelet methods for estimation of acoustically relevant parameters of musical signals
Authors: Kronland-Martinet R. , Guillemain Ph. , Grossmann A.
Published: October 2005

Wavelet transforms are a class of analysis and resynthesis methods that have been found useful in a variety of domains [Wavelets, edited by J. M. Combes, A. Grossmann, and Ph. Tehamitchian (Springer-Verlag, IPTI, 1989)]. They provide a two-dimensional (time and scale) description of one-dimensional signals. In this paper, wavelet transforms are used as the basis for an algorithm that represents an audio signal as a sum of “spectral components” (contributions obtained by amplitude modulation and slow frequency modulation around a frequency). A central role in this construction is played by the phase of the (complex-valued) wavelet transform; the interpretation of this phase is particularly simple and intuitive if the analyzing wavelet is chosen as an analytic signal (progressive wavelet). The parameters determined by the algorithm allow an additive reconstruction of the signal, as well as a variety of nonlinear modifications. These possibilities will be illustrated by several audio examples.
A discrete model for synthesizing guitar sounds
Authors: Chaigne Antoine
Published: October 2005

In an attempt to simulate guitar sounds, a discrete model was developed that consisted of real strings with stiffness and internal damping, the player's finger, the resonance box, and the modelization of the sound pressure. The vibrating string equation was solved in the time domain using the finite-difference method. The player's figure was modeled as a force density term in the string equation, which accounted for the initial plucking conditions. The resonance box was modeled as impedancelike boundary conditions at one end of the string. These end conditions were obtained under the assumption that the driving force exerted by the string at its ends excites a set of second-order resonances in parallel, the first resonance of the plate being coupled with the air resonance of the box. Finally, the sound pressure was modeled as the contribution of a set of monopoles, where each monopole corresponds to one given resonance. The radiation efficiency of each monopole was derived from a curve fitting on a guitar sound-pressure spectrum measured in an anechoic chamber. The waveforms obtained are in good agreement with experimental waveforms measured on real guitars. Sound examples will be presented high-lighting the realism of the simulation.
Statistical analysis of old Serbian music
Authors: Merkle Milan J. , Mijic Miomir
Published: October 2005

Being a link between mathematical scrutiny and a fuzziness of the real world, statistics is often used to help investigate relations between objects or phenomena that are intuitively similar (or dissimilar). In these problems, statistical methods set objective criteria of similarity, thus reducing the human proneness to errors. This work is an attempt to make a contribution to an objective analysis of music. It is not quite clear what are the main objective characteristics of a musical work, and much less is known about how to translate them to a mathematical writing such that when two songs sound similar, one gets two numbers that are close to each other. This is a problem of coding. In this work, the problem of coding is investigated, and the power of various statistical methods in comparison of songs is demonstrated. This is done using samples of an old Serbian chant from the 15th century. This kind of music is used because of its relative simplicity.
Surface acoustic wave technology and applications
Authors: Malocha Donald C.
Published: October 2005

Since the advent of the interdigital transduction for efficient transducer of surface acoustic waves (SAW), SAW devices have provided a key technology for communication and signal processing systems. In order to obtain the required electrical terminal characteristics, a diverse number of SAW components have evolved. Some of these components include bidirectional and unidirectional transducers, couplers, beam compressors, waveguides, and reflectors. The various SAW components are monolithically integrated, using processors similar to those for integrated circuits, on a piezoelectric substrate to produce dispersive filters, linear phase filters, resonators, convolvers, and correlators. These devices are manufactured for a wide variety of applications which include TV-IF, cable TV, VCRs, fiber optic repeaters, radar, spread spectrum receivers, cellular radio, sensors, and a variety of communication receiver applications. The technological innovations and operation of the primary SAW components will be discussed. The presentation will focus on component implementations, their operation, and their most significant parameters. A review of some of the key developments of SAW technology and their applications will also be presented.
Shallow seismic survey, acoustic/seismic coupling, and surface impedance measurements on one site
Authors: Daigle G. A. , Verhaegen C.
Published: October 2005

At low frequencies, ground surfaces can be modeled as a poroelastic layer above an elastic substate. Theory predicts that the presence of seismic resonances within the layer should be observed in surface impedance measurements, due to acoustic/seismic coupling. A number of different measurements have now been made on one site. Core samples show that this ground consisted of a uniform layer of silt of thickness (1.6 ± 0.3) m over bedrock. Seismic velocity measurements were consistent with this simple structure indicating a layer thickness (1.9 ± 0.3) m. Quarter-wavelength related resonances in the silt layer should lead to maxima in the acoustic/seismic admittance spectrum at roughly 45, 135, and 225 Hz and minima at 90 and 180 Hz. Direct measurement of the acoustic/seismic coupling suggests maxima around 50, 135, and 225 Hz and a minimum around 90 Hz. The results also suggest other resonances [there are (a) peaks at about 65 and 185 Hz with a minimum at 130 Hz, and (b) a large peak at 105 Hz with a broad dip at 210 Hz] that are unrelated to the silt layer. Maxima in the acoustic reflection coefficient of the surface should be related to the minima of the acoustic/seismic admittance spectrum. Measurements [G. A. Daigle and M. R. Stinson, J. Acoust. Soc. Am. 81, 62–68 (1987)] of the reflection coefficient suggest peaks at 90–95 Hz, around 130 Hz, and at 200–210 Hz.
Preliminary studies on a bilateral cochlear implant user
Authors: van Hoesel R. , Tong Y. C. , Hollow R. D. , Huigen J. M. , Clark G. M.
Published: October 2005

A series of preliminary psychophysical and speech perception studies have been completed for a bilateral cochlear implant patient. The patient was implanted with a second cochlear implant after 5 yr experience as a monaural implant user. The patient appears to experience some, but not all of the effects of binaural hearing observed in acoustic stimulation. For an approximate matching of the places of stimulation on the two sides, the subject usually reported a single percept when the two sides were simultaneously stimulated. Lateralization was strongly influenced by amplitude differences between the two sides, but only weakly by interaural time delays. Interaural time-delay JNDs were large and were consistent with lateralization results. The amount of bilateral loundness summation for electrical stimulation was similar to that for acoustic stimulation in normally hearing subjects. Central masking was observed to be a function of the degree of place mismatch between the two sides. Preliminary speech testing has been completed. The subject used two independent, but identical speech processors (combined into one unit). The binaural advantage was observed particularly in noise.
Stability, double degeneracy, and chaos in cantilevered pipes conveying fiuid
Authors: Paidoussis M. P.
Published: October 2005

The nonlinear dynamics of planar motions of cantilevered pipes conveying fluid is studied via a tow-mode discretization of the governing partial differential equation, after first replacing inertial nonlinear terms by equivalent ones in the equations of motion through a perturbation procedure. For hanging cantilevers, as the flow velocity U is increased to a critical value, the undeformed vertical configuration of the pipe becomes unstable and bifurcates into stable periodic orbits through a Hopf bifurcation. For a “standing” cantilever, in which the flow discharges from the free, upper end, the pipe is statically unstable for small U if the pipe is sufficiently long; it regains stability through a subcritical pitchfork bifurcation at higher U, and this is followed by a Hopf bifurcation and periodic motions at still higher U. For certain parameter values these two bifurcations occur simultaneously (double degeneracy). By using center manifold theory and normal forms, it is shown that heteroclinic cycles exist in the reduced subsystem, suggesting the possible existence of chaotic behavior. Melnikov computations give guidance as to the likely location of chaotic regions in the parameter space. The numerical results, in which the existence of chaos is demonstrated, are presented in the form of phase portraits, bifurcation diagrams, and Lyapunov exponents.
Routes to chaos in a geared system with backlash
Authors: Kahraman Ahmet , Singh Raj
Published: October 2005

A lumped parameter torsional-bending model of a spur gear pair with backlash and time-invariant mesh stiffness is used to examine chaos. It can be shown that chaos typically exists in a lightly loaded and lightly damped gear pair when excited by the static transmission error. These results are qualitatively, but not quantitatively, similar to the studies reported on Duffing's oscillator. When radial clearances in rolling-element bearings are introduced, a three degree of freedom nonlinear model is used to identify two different routes to chaos: (i) period-doubling route and (ii) quasiperiodic route. Predicted time histories, phase planes, Poincaré maps, and Fourier spectra will be illustrated to examine these. Limited experimental data. as available in the literature will be cited in evidence. [Work supported by NASA Lewis Res. Ctr.]
Small Business Innovation Research (SBIR) funding: A case study in bringing a computer-based speech training aid into the marketplace
Authors: Kewley-Port D. , Watson C. S. , Maki D.
Published: October 2005

University research projects occasionally involve applications of speech technology that are intended to produce a commercial product, such as the development of a microcomputer-based speech training aid (ISTRA) at Indiana University. Even though initially funded as basic research, such projects, when successful, eventually require an alternative source of funds. SBIR programs in most federal agencies have been mandated by Congress to support research and development in small businesses. This presentation will describe SBIR programs and the funding process, from the perspective of investigators who attempt to form small businesses outside of the university womb. [Research support by NSF and NIH, SBIR Phase I grants.]
From theory to practice: A 10-yr path
Authors: O'Malley M. H.
Published: October 2005

Berkeley Speech Technologies, Inc. has been developing commercial text to speech synthesis technology for over 10 yr. What started out as a quick “technology transfer” has grown to become a complex body of “intellectual property” that has been realized in such products as a 100 000-word talking dictionary, a telephone response system with 16 T-T-S lines on one board, a satellite communication system for trucks, and a portable talking computer for blind users. Practical considerations caused modification of the initial theoretical assumptions. From the beginning, it was assumed that high intelligibility and high phoneme accuracy were essential, but it was soon learned that 700 words per minute with a 25-ms start and stop are equally important for blind users. Similarly, academic research had assumed wide bandwidth and low noise, but telephone systems require that all of the speech information be packed into a 3.5-kHz telephone bandwidth. Initially, the choice was made to use demi-syllable synthesis because it seemed to be an “engineering shortcut” that might cover gaps in standard scientific descriptions. As the technology developed, however, the decision was made to convert to a more scientifically based synthesis model because it offered higher quality, greater flexibility, and faster development, especially of new languages. Our 10-yr development could not have been justified on the basis of expected financial return. However, it was and is fun.
Issues for bringing speech technology to the marketplace
Authors: Boll Steven F.
Published: October 2005

ITT A/CD is one of a few companies that has taken on the challenge of transferring speech technology from an abstract algorithm to a working system. In the mid-1980s IIT A/CD developed moderate vocabulary, speaker-dependent, connected speech recognition, and produced a series of single-board recognizer products compatible with various personal computers and workstations. To date, over 400 single-board recognizers have been ordered by over 30 customers. From these efforts a number of issues and experiences have surfaced that are critical to their successful operation. Some of these include: (1) robust performance across channels and microphones (every change to the microphone, analog recording setup, and room environment, will affect, and, most likely, lower performance); (2) training sensitivity (every shortcut taken in training, i.e., using fewer tokens, cross-channel recordings, out-of-vocabulary training phrases, etc., will lower performance); (3) man-machine interface (use psychologists to design user interfaces rather than engineers; otherwise they end up being just too complicated); (4) application specification (select only those applications that receive an overwhelming and compelling advantage from the technology).
Error correcting alternatives in the machine recognition of speech
Authors: Bellegarda Jerome R. , Kanevsky Dimitri
Published: October 2005

Because for many users the machine recognition of realistic, large vocabulary speech tasks typically produces recognition rates only in the neighborhood of 80%, the use of automatic speech recognizers (ASR) requires spotting and correcting the errors introduced during decoding. As a result, error correction is an important consideration in bringing current speech recognition technology to market. Standard methods for error correction include overwriting using a keyboard and selecting from a list of related candidate words with a mouse. In addition, a new alternative is considered: overwriting using a pressure-sensitive pen on a transparent paper-like interface (PLI) capable of recognizing handwriting online. These three alternatives are compared in terms of relative speed and ease of use, by measuring the average time that it takes to correct speech recognizer errors using each of the alternatives, and by recording the overall impression of each of the participants in the experiments. Keyboard entry was viewed as slow and distracting, because the attention must shift from screen to keyboard. Mouse selecting was faster but inappropriate for out-of-vocabulary words, such as proper nouns. Overall, PLI came across as a very viable means of correcting ASR errors, even in writer-independent mode. This is because the correct word often differs from the wrongly decoded one by only a few characters, thus enabling the use of (robust) discrete character recognition. In a typical experiment, users were able to correct all ASR errors after a maximum of two passes at each wrongly decoded word.
Hidden Markov model based automatic keyboard spotting for automating operator services
Authors: Modi P. , Wilpon J. G.
Published: October 2005

A basic assumption for most current speech recognition systems is that the speech to be recognized consists solely of words from a predefined vocabulary. For speech recognition applications in the telephone network, where any person can pick up any telephone at any time from anywhere, it is naive to assume that users will adhere strictly to this protocol. For example, when a customer is asked to say only collect, calling-card, third-number, person, or operator, they may instead say, “I want to make a collect call please.” In Wilpon et al., a hidden Markov model based keyword spotting algorithm was presented, which recognizes keywords from a pre-defined vocabulary list spoken in an unconstrained fashion. In this talk, there will be discussion of several advances to the previous algorithm including: (1) improved rejection criteria and (2) improved modeling techniques. Results will be presented from evaluations on a five-word vocabulary used to automate operator-assisted calls. Using wordspotting techniques to recognize speech, a recognition accuracy of 99.8% is currently being achieved, while rejection is only 5.0%, regardless of whether the user speaks only vocabulary words or not.
What does a matched-field processor try to do?
Authors: Kuperman W. A.
Published: October 2005

Matched-field processing (MFP) presents a fertile area for applying, developing, and understanding new neural network architectures. Orders of magnitude in computational efficiency are still required to fully take advantage of MFP. Essentially, MFP is a signal processing technique that uses received acoustic data across an array to locate the position of radiating sources in, at best, an imperfectly known ocean environment. This ocean domain is huge. The concepts of MFP are reviewed emphasizing those aspects of the problem that would make it an attractive candidate for applying neural networks. The importance and subsequent treatment of the environment, which is characterized by oceanography, bottom geophysics, and ambient noise, are probably the most central issues from which the benefits of neural networks might emerge. Also relevant is the connection between MFP and remote sensing that further magnifies the impact of new computational structures.
Artificial neural networks for field reconstruction and source localization
Authors: Chin S. , Beran M. , Howard J. , Steinberg B.
Published: October 2005

Neural networks have been used to successfully solve the inverse problem of determining the source location based on field or coherence measurements. The conditions and types of neural networks that are effective in solving this problem are discussed. Neural network topology will be considered, as well as the source-array geometry and its effect on performance. Simulation results will be provided, which demonstrate under what conditions interpolation is possible. Finally, a discussion of other source localization algorithms will be given. These methods will be contrasted to the neural network methodology.
Neuronal contributions to auditory brainstem potentials (ABR)
Authors: Starr A. , Zaaroor M.
Published: October 2005

ABRs were studied in cats for up to 45 days after kainic acid injections into the superior olivary complex (SOC) or cochlear nucleus (CN), which destroyed neurons while sparing fibers of passage and the terminals of axons of extrinsic origin. With destruction of the CN neurons, there was a loss of components P2 (III), and P3 (IV) and only a modest attenuation and prolongation of the latency of components P4 (V) and P5 (VI). Unexpectedly, to stimulation of the ear contralateral to the injection side, waves P2 (III), P3 (IV), and P4 (V) were also attenuated and delayed in latency. The sustained potential shift from which the components arose was not affected. Following bilateral SOC destruction, there was a moderate attenuation of wave P2 (III), and a marked attenuation of up to 80% of P3 (IV), P4 (V), P5 (VI), and the sustained potential shift. No component of the ABR was totally abolished. The results are compatible with multiple brain regions contributing to the generation of the components of the ABR beginning with P2 (III) and that components P3 (IV), P4 (V), and P5 (VI) and the sustained potential shift depend particularly on the integrity of the neurons of the SOC bilaterally.
Neural nets for seafloor classification: A computer simulation study
Authors: Alexandrou D.
Published: October 2005

The point scattering model of reverberation offers estimates of a large number of reverberation statistics. However, fitting the model to data is not immediately meaningful owing to the model's weak physical connection. The approach taken here utilizes parametrizations of the reverberation probability density function (pdf) to create acoustic “feature vectors” representative of different seafloor types. The problem of seafloor classification is then tantamount to a problem of pattern recognition. Computer simulations are used to create a number of different scatterer distributions representing different seafloor regimes. Several flexible pdf families, including the most general pdf supported by the point scattering model and the generalized Gaussian pdf, are used to parametrize synthetic reverberation data. A neural net classifier is trained to recognize the prevailing seafloor type based on its acoustic signature and is tested for its ability to discriminate among different scatterer distributions. A number of neural net algorithms are evaluated in this setting, including the linear perceptron and a backpropagation of error algorithm. [Work supported by ONR.]
Pulse propagation in coupled one-dimensional dynamic systems
Authors: Dickey J. , Maidanik G.
Published: October 2005

A pulse propagating in a complex of connected one-dimensional dynamic systems is formulated using delay operators. These operators are used to construct an impulse response operator for the complex in the temporal domain. This impulse response operator is then used to describe the response of the complex at a localized position in space and time due to an external drive applied to another position localized in space and time. Simple examples are presented and discussed for complexes composed of one and two dynamic systems that are connected through junctions characterized by reflection and transmission coefficients.
Numerical analog to plate equation-including shear and rotary inertia for discontinuous materials
Authors: Wendlandt B. C. H.
Published: October 2005

A numerical analog to the wave equation for a plate is presented and results of propagation studies of Gaussian and sinusoidal wave packets across the plate reinforced by ribs of four times the thickness of the plate are discussed. Results are presented for selected rib numbers, widths, and spacings. The wave equation is based on theory presented by Skudrzyk, modified to consider Kelvin-type viscoelastic behavior and discontinuous material properties and plate thickness. The plate equation includes consideration of shear and rotary inertia. It is able to describe the propagation of flexural waves whose wavelengths vary from being long compared to the plate thickness to waves whose wavelength is 20% the plate thickness. Plate distortions are assumed to be antisymmetric about a plane through the center of the plate, the points of this central plane moves up and down, but not to the left or right. The wave equation in terms of the plate displacement w along the direction normal to the plate, of thickness h, is in the operator form, Λ2Λ1w + ρh [1 + 〈βμ〉 (∂/∂t)] (∂2w/∂t2) = Υq, where t is time and the operator Λ1 = ∇〈μ*〉h⋅∇ − ρh(∂2/∂t2), the operator Λ2 = ∇D*⋅∇1〈μh〉 − 〈ρh2〉12〈μ〉 ∂2∂t2(1 + 〈βμ〉∂∂t), and the source operator acting on applied pressure q is Υ = 1 + 〈βμ〉∂∂t − ∇D*⋅∇1〈μh〉 + 〈ρh2〉12〈μ〉 ∂2∂t2(1 + 〈βμ〉∂∂t), where 〈 〉 indicates that the material property has been averaged over a small volume of material which can be justified from the integral definition of the divergence theorem and represents a first-order approximation to variations in the material property. The variable D* is defined by D* = Y*h3/[12(1 − v2)], where v is Poisson's ratio and Y* = Y[1 + βy(∂/∂t)] is Young's modulus. The shear modulus μ* = μ[1 + βμ(∂/∂t)]. The effects of viscoelastic relaxation of the material in the Kelvin form are expressed by βy(∂/∂t) and βμ(∂/∂t) where the β's are the viscoelastic relaxation times of the material. The material and plate parameters are a function of spatial position. A satisfactory numerical analog of the plate equation above has been developed by approximating the spatial and time derivatives by centered finite differences that divide the time domain into equal levels of time and the spatial domain into a grid where the functional values are computed at the grid nodes. The numerical analog to the plate equation is evaluated by the alternating direction implicit (ADI) technique. Current studies aim to identify the more significant parameters that determine the transformation of acoustic waves at beam/beam and plate/beam junctions.
A simplified absolute method for point impedance measurement
Authors: Ge Li-Feng
Published: October 2005

A previous work has indicated that an electromechanical transducer, i.e., an electrodynamic-type or piezoelectric-type vibration exciter, can be used for determining the point impedance by the reciprocity principle, and a general formula was given [L.-F. Ge, J. Acoust. Soc. Am. Suppl. 1 87, S208 (1990)]. The present study develops a simplified and practical method. It is only needed to measure four complex electrical impedances of the exciter, W1, W2, W3, and Wx, corresponding to three known different mass loads, M1, M2, M3, and an unknown structure or material coupled successively on the exciter table while it is vibrated at a frequency of interest. The driving point impedance of the structure or material at the frequency f can be found from the equation: Zx = jω(M2M31W2Wx1 − M1M32W31Wx2)/(M31W32Wx1 − M32W31Wx2), where j2 = −1, ω = 2πf, double subscripts mean differences, such as M31 = M3 − M1, Wx1 = Wx − W1. The equation would be further simplified by the selection of M1 = 0, i.e., no-load, to: Zx = jωM2(1 − M32W31Wx2/M3W32Wx1)−1. Measurements performed on a piezoelectric vibration calibrator have verified the method. Better results can be obtained from an optimized set of known mass loads.
An approximate mode-to-mode power flow formulation for line-coupled structures
Authors: Bremner Paul , Pan Jie
Published: October 2005

It is often desirable to compute mode-to-mode power flow between substructures. Indeed it is necessary when low modal populations in the analysis bandwidth cause excessive variance in statistical energy analysis (SEA) predictions of vibrational energy levels. An important class of such problems occurs for substructures connected along continuous line junctions. Examples include simple plates and beams (for which analytical modal solutions are known), but can include quite general finite-element substructures (for which numerical modal solutions can be computed). This paper proposes and evaluates a simplified mode-to-mode power flow formulation, which uses a complementary set of constrained and free uncoupled modes to approximate power flow between both resonant and nonresonant modes of the coupled substructures. Comparison with analytical solutions demonstrates the usefulness of this formulation.
Auditory phonetics
Authors: Diehl Randy , Lindblom Björn
Published: October 2005

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.
Automatic identification of place of articulation of nasal consonants using the PLP method
Authors: Qi Yingyong
Published: October 2005

Until recently, techniques on speech analysis have been built around the all-pole linear predictive model. This study demonstrates the advantage and effectiveness of using the perceptual linear predictive (PLP) method in analyzing nasal consonants. Six speakers (three men and three women) produced 300 CV syllables with initial nasal consonants /m/ and /n/. A threshold-based boundary detection algorithm was developed to extract nasal segments from CV contexts. Poles of the fifth-order PLP model were calculated to characterize the place of articulation of nasal consonants. Results indicated that the frequencies for the transformed poles, particularly for the second pole, were significantly lower for /m/ than for /n/ and were independent of factors such as vowel context and gender of the speaker. A nasal identification rate of 86% was obtained based on the frequency of the second pole. The use of the PLP method clearly has overcome difficulties associated with the antiresonance in analyzing nasal consonants.
Acoustic phonetic property development in a neural net model of early auditory learning
Authors: Seebach Bradley
Published: October 2005

The existence of phonetic property detection by early infancy has been held as evidence for the existence of a genetically specified language acquisition device. A biologically plausible neural network was used in order to determine whether phonetic property detection might be the result of inductive learning in fetal humans, rather than genetic specification. Synaptic modification in the net is a function of presynaptic activity, synaptic strength, instantaneous cell activity, time-averaged cell activity, and global inhibition. Using critical band analysis and series of overlapping time windows, stop-vowel syllables from several speakers were transformed into lower-dimensional energy surfaces and used to train and test the network. Results suggest that detection of some phonetic properties may develop via inductive learning in auditory cortical connections. Neurons became selective for subsets of phonetic feature spaces, and developed representations of distinctive features that match classically described distinctive features for place of articulation. The network's solution makes use of the affinity structure of consonants.
Multisource detection using matched-field processing and scientific visualization techniques
Authors: Chin Steven H. , Schuette Lawrence C. , Heitmeyer Richard
Published: October 2005

The matched-field processor has the ability to provide good source location estimates for a single source in a given search volume. If the scenario is complicated by the presence of multiple sources, the detection/localization capability of the matched-field processor will be degraded from the case where only a single source is present. The sensitivity of localization estimates to varying configurations (e.g., number of sources, source levels, spatial distribution) involving multiple sources are considered. The Bartlett and maximum likelihood processors have been applied, and new three-dimensional visualization tools have been developed in order to address these issues. Simulated results are presented to illustrate the efficiency of the technique. [Work support by NRL.]
The effect of internal waves on matched-field processing
Authors: Jackson Darrell R. , Ewart Terry E.
Published: October 2005

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.]
A “true” maximum likelihood method for directional wave spectra estimation and matched-field source localization
Authors: Baggeroer Arthur B.
Published: October 2005

Most methods of estimating directional spectra involve a step wherein the cross spectral covariance of the signal field over the array elements must be estimated, When the arrays are large and the data sparse, this estimate is singular or poorly conditioned. Several methods of mitigating this, including diagonal loading, eigenvalue thresholding, and subspaces have been traditionally used to circumvent these singularities. The fundamental problem is that an arbitrary covariance matrix has many more degrees of freedom than the data can constrain. A new algorithm is introduced that starts directly from the data to form an estimate of the covariance matrix that is constrained by the wave equation describing the propagation of the directional signals. It is found that a “true” maximum likelihood estimate (not a minimum variance, distortionless filter in the guise of maximum likelihood) can be specified and an iterative algorithm for implementing it can be derived. The results are similar in structure to those derived by Snyder and Miller (Proc. IEEE, July 1987) for estimating power densities by imposing a Toeplitz constraint. The algorithm can be extended to matched-field processing for localizing-independent sources. One of the advantages is that a priori information about the sources can be used in estimating their distribution.
Reduction of matched-field sidelobes by subtraction of cross-spectral matrix elements associated with strong signals
Authors: Bucker Homer P.
Published: October 2005

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.
Shallow water waveguide: Modal analysis of acoustic measurements taken on the New Jersey continental shelf
Authors: Bishop Judith , Doutt James , Carey William
Published: October 2005

A calm sea state, low-frequency sound transmission experiment was conducted in shallow water on the New Jersey continental shelf in the vicinity of the AMCOR 6010 borehole [L. Maiocco, W. Carey, E. Parssinen, and J. Doutt, J. Acoust. Soc. Am. Suppl. 1 86, S8 (1989)], where bottom topology and composition have been previously characterized. A source towed at 12- and 34-channel depth was used to transmit cw (two families of four tones, frequency range 50–600 Hz) and pulsed (Gaussian, LFM) signals to a 24-element vertical hydrophone array deployed in 73 m of water. Measurements were obtained along a constant depth radial at tow speeds of 5 kn, 2 kn, and adrift, and at 5 kn along a varying-depth radial course. An internally consistent description of normal mode propagation was sought at 50 and 75 Hz from a combination of (a) horizontal wave-number spectra obtained from the Hankel transform inversion [G. Frisk and J. Lynch, J. Acoust. Soc. Am. 76, 205–216 (1984)] of the complex pressure field at 24 separate depths, (b) the vertical wave-number spectrum determined as a function of range from the spatial transform of complex pressure over the real aperture formed by the vertical array, and (c) the normal mode shapes and amplitudes constructed from the depth dependence of pole peak amplitude in the wave-number domain and of pulsed transmission amplitude in the time domain. A unique feature of the analysis is the successful generation of well-resolved wavenumber spectra from the sound field projected by a moving source in the presence of multiple receivers. These spectra have been filtered and inverted to yield noise-free estimates of signal level and transmission loss.
An adaptive cooling schedule for simulated annealing with application to multiple-constraint time-domain beamforming
Authors: Davis Nolan R. , Berkson Jonathan M. , Perkins John S. , Collins Michael D. , Kuperman W. A.
Published: October 2005

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 [Kuperman et al., “Optimal time-domain beamforming with simulated annealing including application of a priori information,” 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.
The underwater sound generated by light and heavy rain
Authors: Nystuen Jeffrey A.
Published: October 2005

The physics of sound generation from light and heavy rain will be discussed in the context of the measurement of rainfall rate by monitoring underwater ambient sound levels. The physics of sound generation by small drops/light rain at normal and oblique incident will be reviewed. These results will be contrasted with preliminary field measurements of the sound produced by heavy rain during the Acoustic Rain Experiment at the Ocean Test Platform in the Gulf of Mexico. The underwater sound spectra generated by heavy rain will be explained using new laboratory measurements of the sound generated by individual large drops showing acoustic energy from initial impacts (at 10–15 kHz) and from entrained bubbles at lower frequencies.
Resolution performance of the overlap-correlator synthetic aperture technique
Authors: Edelson Geoffrey S. , Sullivan Edmund J.
Published: October 2005

The resolution capabilities of the overlap-correlator technique for passive acoustic synthetic aperture array processing are considered. The overlap-correlator method [S. Stergiopoulos and E. J. Sullivan, J. Acoust. Soc. Am. 86, 158–171 (1989)] can successfully lengthen the effective aperture of a towed array in the passive mode. In this present work, previously reported results on the performance bounds of the overlap-correlator bearing error variance for a single acoustic source as a function of signal-to-noise ratio, the number of elements in the physical army, and the size and number of the overlaps [J. Acoust. Soc. Am. Suppl. 1 87, S155 (1990)] are extended. In addition, the performance of the technique in resolving multiple sources is investigated. This performance is compared to the corresponding Cramer-Rao bounds for static line arrays and towed synthetic aperture line arrays. Results using simulated data are presented.
An experimental study in forming a long synthetic aperture in a nonisotropic ocean
Authors: Urban Heinz , Stergiopoulos Stergios
Published: October 2005

The formation of a passive synthetic aperture in a nonisotropic ocean is a complex problem that includes two main areas of investigation. The first one, which is discussed in another study, is related to the limitations imposed by the temporal and spatial coherence properties of the underwater medium on the size of an effective synthetic aperture. The second area of investigation is related to the development of an algorithm that will synthesize coherently the successive spatial information of a small moving line array into an extended aperture. The synthetic aperture techniques, that have been discussed in the open literature, however, have not been fully tested with experimental results including many targets. The present study is concerned with an experimental attempt in forming a long (i.e., 100 λ) synthetic aperture in an ocean where the spatial and temporal coherence properties of the acoustic field are known. For a multitarget environment, results indicated that an optimum synthetic aperture processing scheme has to correct, (1) for the defocusing effects due to the relative movement of a source with respect to the towed array, and (2) for the sources that are in close range and the plane-wave approximation is not valid any more. A review of the existing synthetic aperture techniques and an approach for optimizing their performance are also discussed.
Limitations on towed-array gain imposed by a nonisotropic ocean
Authors: Stergiopoulos Stergios
Published: October 2005

This study is concerned with the limitations imposed by the characteristics of the ocean on the gain of hydrophone arrays. The multielement physical aperture of a passive array operates in a nonisotropic noise field and in a medium of limited spatial coherence length. In particular, it is assumed that, due to distant shipping, the noise field is partially directive and imposed upon an isotropic noise background, and in the rough-bounded transmitted medium, many multipaths exist that are closely spaced in arrival time and arrival angle. In the theoretical development of this study, a processing scheme is suggested for the derivation of the array gain from the received signal and the directivity pattern of the noise field, which is assumed to be known. Results from simulations of this processing scheme have indicated that a comparison of the experimental estimates of towed-array gains with the above array gain predictions for different values of the spatial coherence length can provide an approximate estimate of the coherence length of the medium. Results from real-data applications of the theoretical development indicate that the coherence properties of the ocean can be sufficient for effective long (80λ) towed-array applications.
Arrival time tracking of partially resolved acoustic rays with application to ocean acoustic tomography
Authors: Chaulk Edwin K. , Hippenstiel Ralph , Miller James H.
Published: October 2005

Accurate estimation of the arrival time of a ray is an important aspect of ocean acoustic tomography. A straightforward method of arrival time estimation, based upon locating the maximum value of an interpolated arrival, was used with limited success for analysis of data from the December 1988 Monterey Bay Tomography Experiment. Close examination of some data revealed multiple, closely spaced arrivals of similar amplitude, only partially resolved in many returns. A modification to the original tracking algorithm succeeded in improving the estimates and lead to the development of a tracker, based upon a linear predictive filter that provides reasonable estimates for the arrival times of poorly resolved peaks. In addition, the filter coefficients provide estimates of the dynamic spectrum of the sequence of arrival times. At present, the tracker utilizes only magnitude information. A possible extension may refine estimates of poorly resolved arrivals by incorporating phase information. [Work supported by the NPS Research Council and the Monterey Bay Aquarium Research Institute.]
Articulation index: Importance function in the intensity domain
Authors: Boothroyd Arthur
Published: October 2005

In the classical derivation of the articulation index (AI) account is taken of the relative contributions of different parts of the acoustic spectrum in the frequency domain, as represented by the importance function. In the intensity domain, however, it is assumed that there is a constant contribution per decibel over a range of 30 dB. In fact, the intensity domain can be treated like the frequency domain, assigning different weights to different parts of the intensity distribution. Moreover, by integrating the logarithm of the error probability over intensity, it is possible to predict the shape of the performance/intensity (PI) function that relates recognition probability to signal-to-noise ratio. Under the assumption of a quasi-Gaussian importance function in the intensity domain, predicted PI functions are very similar to those obtained experimentally. Modifications to the parameters of the importance function permit modeling of some of the effects of changes of vocal effort, talker distance, and sensorineural hearing loss.