bluuebunny/crossref_metadata_2025 huggingface dataset:
Performance of adaptive beamforming (ABF) for fixed receiver bistatic active systems
Authors: Lee Yung P. , Hanna John
Published: October 2005

Data collected from a sea test have been processed and analyzed to examine the performance of adaptive spatial beamforming techniques in fixed receiver bistatic active systems. Adaptive beamforming using both the least-mean-squared (LMS) method and block sample-matrix-inversion (SMI) method have been studied. The LMS method uses the current input signal to recursively update the adaptive weight; it converges slowly and provides only a little adaptive gain (over the conventional beamforming) for reverberation suppression. On the other hand, the SMI method uses an estimate of the current covariance matrix to recursively update the adaptive weight; it converges rapidly and provides more than 10 dB adaptive gain. It is well known that adaptive processing is very sensitive to mismatch due to system errors or environmental fluctuations. To recover the signal lost in mismatch, two robust SMI adaptive beamformers, Feedback-Loop White-Noise-Constrained (FLWNC) method and Signal-Coherence-Constrained Reduced-Rank (SCCRR) method, were applied. They successfully recover signals and maintain excellent clutter suppression.
Dynamic stereo imaging of hard-skinned sonar targets
Authors: Henderson Terry L.
Published: October 2005

The visible portion of a hard-skinned sonar target can be scanned in xyz coordinates by beamforming a planar array, to give a 3-D representation. However, a simple line array running along the x axis can provide a projection view of a target that lies along the z axis, with its upper and lower surfaces being projected semitransparently onto the xz plane, giving a picture analogous to that of a biological specimen viewed with a slide microscope. If the line array is tumbled about the z axis, it will appear that the target is revolving around the z axis. When this tumbling view is presented 14° out of phase to the left and right eyes, a stereoscopic view of the tumbling target is obtained, with its 3-D form revealed by the brain’s innate tomographic signal processing capability. In-water experiments have been conducted for several target shapes. Other means of generating dynamic displays have been developed that are suitable for scanning objects on the seafloor. A videotape of results for several targets has been prepared. Further simplifications of the array structure and processing are possible, with some limitations and weaknesses. [Work supported by ARL:UT IR&D.]
Optimum detection of solid spherical elastic objects of uncertain composition in an uncertain reverberation-limited environment
Authors: Wazenski M. , Premus V. , Alexandrou D. , Nolte L. W.
Published: October 2005

Detection of solid spherical elastic objects is approached via optimal detection and estimation theory in conjunction with acoustic scattering models for both the object and the environment. The objective is the development of full field (amplitude and phase) optimal decision methods for high-frequency active detection of a target in the vicinity of the seafloor. This parametric approach incorporates the inherent uncertainty of both object composition and seafloor geomorphology. The acoustic model used for the spherical object is the well-known modal series solution and is parametrized by object size, density (object and medium), compressional wave speed (object and medium), and shear wave speed. This deterministic model is used to predict the target signature measured at an array of sensors. Acoustic modeling of the seafloor is performed by applying the 3-D Helmholtz–Kirchoff formulation to the anisotropic, power-law description of seafloor relief due to Goff and Jordan and is parametrized by correlation length (two directions), lineation direction, rms height, and fractal dimension. This model is used to predict the spatial and temporal coherence of the scattered sound field. Illustrative detection examples are presented along with receiver operator characteristic (ROC) performance. [Work supported by ONR: Ocean Acoustics.]
Reconstruction of a periodic time-domain signal ‘‘buried’’ in Gaussian noise
Authors: Sidahmed Stefan T. , Sarkady Antal A.
Published: October 2005

Power spectral analysis does a reasonably good job of signal detection but fares poorly in characterization because the phase information is ignored in the spectral computations. From spectral analysis, an initial or seed value for the period of the target signal is obtained. The original digitally sampled noisy time signal is resampled (up/down sample rate conversion) at such a rate that precisely an integer number of time samples are obtained on each cycle of the quasiperiodic target signal. These noisy cycles are synchronously averaged in the time domain and an estimate for the shape of the target signal is computed. The sample rate conversion, and the synchronous averaging is repeated until the rms value of the result is maximized (i.e., phase cancellation is minimized). The voltage signal-noise ratio of the reconstructed target signal is proportional to the square root of the number of target cycles averaged. This procedure was tested with laboratory type signals and found to have an excellent performance at signal-to-noise voltage ratios as low as −40 dB.
Asymptotic behavior of the matched-phase noise filter in the limit of low signal-to-noise ratio
Authors: McDonald B. Edward , Orris Gregory J.
Published: October 2005

It has been shown elsewhere that information about the shape of the noise spectrum with no phase knowledge can be used to construct a noise rejection algorithm which we call the matched-phase filter. Here, it is demonstrated that to first order in the signal-to-noise ratio (SNR), the matched-phase filter is capable of improving an arbitrarily low SNR (≪1) to order unity when the spectrum of the noise is known exactly. This analysis accounts for properties of numerical results with computer-generated signal and noise, with SNR values below −100 dB. Since computers have dynamic ranges far in excess of audio equipment, these results in an actual experiment would be limited by the dynamic ranges of the equipment used. Also shown are examples illustrating the performance of the algorithm when the noise spectrum is known only approximately.
The development of speech perception: Beyond infancy
Authors: Walley Amanda C. , Michela Victoria L. , Flege James E.
Published: October 2005

In comparison to what is known about speech perception in infancy and adulthood, much less is known about perception in childhood. Nevertheless, there is a growing consensus that phonetic/phonemic segments are not present in the ‘‘initial state,’’ but only emerge gradually over the course of childhood as units of speech representation and/or processing. How perception may become more segmentally based by virtue of vocabulary growth will be considered—as the mental lexicon grows in overall size, as individual lexical items become more familiar, and as particular regions of the lexicon become more ‘‘crowded’’ (i.e., as the acoustic–phonetic similarity among different words increases). The research to be presented includes recent data bearing on children’s perception of native and non-native vowels in word and nonword contexts. Age-related changes in spoken word representation and processing may have important consequences for other developmental achievements, such as the growth of metaphonological knowledge/skills and early reading ability, as well as for success in learning a second language. More generally, the purpose is to help set an agenda for future research that will provide a more complete picture of the developmental trend for speech perception. [Work supported by NIH.]
A multiresolution, likelihood-based approach to pattern classification with application to characterization and automated recognition of marine mammal sounds
Authors: Hayward Thomas J.
Published: October 2005

A multiresolution, likelihood-based statistical approach is presented for characterizing labeled classes of samples (e.g., measured time series or time-frequency distributions of acoustic transients) and for classifying new samples based on this statistical characterization. The labeled classes are characterized by a histogram associated with a multiresolution decomposition of the data in each class. Classification of a new sample is then performed by calculating, for each labeled class, the conditional probability of the sample given the statistics of that class. These conditional probabilities, which are interpreted as relative likelihoods that the sample belongs to each of the classes, are calculated in a recursive computation that proceeds from coarse to fine resolution. A simple, efficient computer implementation using associative arrays is presented. Successful classification of both time series and time-frequency distributions of marine-mammal vocalizations is demonstrated using relatively small numbers of labeled samples (∼10 per class). [Work supported by ONR.]
Time domain based optimal a posteriori source localization in a shallow-water environment
Authors: Shorey J. A. , Nolte L. W.
Published: October 2005

An optimal aposteriori probability approach to matched field processing using time domain data in the form of arrival times is implemented. The inherent robustness of this technique is demonstrated with respect to conventional narrow-band frequency domain based matched field processing techniques. A ray-tracing method and a wideband normal mode method are compared as modeling mechanisms for this application. Localization performance for a shallow-water source localization problem is presented using probabilities of correct localization. Specific results for this shallow-water environment are presented as a function of signal-to-noise ratio. [Research sponsored by ONR Ocean Acoustics and NUWC.]
Prosodic cues for phrasal boundaries in productions by children and adults
Authors: Verma Sushama , Mannering Anne M. , Kornell Branden M. , Katz William F. , Beach Cheryl
Published: October 2005

Although recent evidence suggests that young children use prosodic information to ‘‘bootstrap’’ themselves into syntactic comprehension, little is known about how children use prosody to signal syntactic boundaries in their spoken utterances. A speech elicitation task examined the extent to which children and adults control duration and intonation to mark phrase boundaries in spontaneous speech. Adults and young children (ages 5 and 7) were asked to describe groupings of colored blocks in a manner such that a blindfolded listener could tell ‘‘which blocks go together.’’ This procedure elicited utterances corresponding to three syntactic bracketings of the phrase ‘‘pink and green and white.’’ Acoustic analysis of subjects’ productions indicated that adults reliably controlled duration (i.e., syllable lengthening, pauses) and intonation fall–rise patterns to signal phrase boundaries. In contrast, children’s productions were highly variable and showed no evidence of intonation cues being involved. Individual subject analyses suggested that some of the 7-year-old children used duration cue to signal phrase boundaries, although the contrasts were not as marked as in the adult data. The results were interpreted as showing that by age 7, most children have not yet mastered prosodic cues for phrasal boundaries in their speech production.
Application of Navy IUSS for whale research
Authors: Clark Christopher W. , Mellinger David K.
Published: October 2005

The recent application of Navy Integrated Undersea Surveillance System (IUSS) data (Whales ‘93) for detecting and tracking low-frequency sounds from whales has led to a renewed interest in the use of these arrays for marine mammal research. Previous work with sparse arrays has limited the detection and tracking to ranges of tens of kilometers, while the IUSS allows for orders of magnitude greater coverage. SOSUS data are processed via computer systems specifically developed for detecting and tracking whales, as well as through annotations, to determine individual, geographic, and seasonal variability in vocalizations for at least four species (blue, finback, humpback, and minke), and there are also numerous sounds from unidentified whales. Given that these animals use these deliberate patterns of intense low-frequency sounds for either communication, navigation, or food finding, verification of sound function will be deduced through careful examination of behavioral context and other environmental conditions such as local productivity, bathymetry, and oceanography. Results from the Whales ‘93 and continuing Dual Use projects will be presented. This will include descriptions of signal variability for the different species of marine mammals and preliminary interpretation of the results in terms of sound function. [Work supported by ONR and NRL.]
Auditory-evoked potentials obtained using continuous amplitude-modulated tones
Authors: Dolphin William F.
Published: October 2005

Auditory-evoked potentials (AEPs) recorded from the scalp surface are neurogenic potentials arising from the massed discharge of neural populations in response to acoustic stimuli and provide a ‘‘window’’ on the physiological basis of auditory processing. AEPs, in particular the transient-evoked auditory brain-stem response (ABR) have been widely used in human clinical settings for the evaluation of auditory functioning. Recently, several new techniques using continuous amplitude-modulated tonal stimuli have been investigated. Many of these techniques take advantage of auditory nonlinearities which result in the auditory system acting as an envelope extractor. This envelope following response (EFR), phase locking of neural discharge to waveform periodicities, has been measuring using two-tone, sinusoidal amplitude-modulated, and multitone, multiple envelope components signals. The EFR offers great promise for frequency specific threshold measurements, especially for low-frequency audiometry and for the examination of auditory processing of complex sounds.
A portable system for physiological assessment of hearing in marine animals
Authors: Carder D. A. , Ridgway S. H.
Published: October 2005

A compact field portable system has been developed and tested for presenting sound stimuli (20 Hz–200 kHz) and acquiring and analyzing sound and microvolt-level electrophysiological responses in marine animals. Stimulus sound is controlled for frequency, amplitude, duration, rise/fall times, interstimulus interval, phase, and masking noise. Analysis is by synchronous averaging, with artifact rejection, and is displayed, stored, and may be printed, if desired. The weather resistant system includes hardware based on a 486 with fast bus and software for presenting shaped tones and pulses, acquiring up to eight channels of data for display, storage, and analysis. The system includes all necessary functional units (suction cup electrodes, high gain (X500K) amplifiers, analog-to-digital and digital-to-analog converters, fiber-optic intermodule connection, computer with LCD screen that can be seen in bright sunlight, an internal 1.2-gigabyte hard disk, power amplifier, transducers, power source) and is constructed in a set of small cases that can be hand carried to remote sites. To date, the system has been used for hearing tests in dolphins, Tursiops, white whales, Delphinapterus, and a beached pygmy Sperm whale, Kogia. [Work supported by ONR.]
Rhythmogram analysis of human and synthetic performance
Authors: McAngus Todd Neil P. , Clarke Eric , Windsor Luke
Published: October 2005

Previously, a method of performance analysis has been described which is based on a multitimescale decomposition of the acoustic signal [Todd, ‘‘Wavelet analysis of rhythm,’’ J. Acoust. Soc. Am. 93, 2290(A) (1993)]. This analysis is sensitive to a variety of expressive devices employed by musical performers including tempo, dynamics, and articulation. The output of the analysis is a ‘‘rhythmogram’’ which resembles the time-spanreduction of the theory of Lerdahl and Jackendoff. In this paper, the results of some analyses of real and synthetic performances are presented [Todd, ‘‘The dynamics of dynamics: a model of musical expression,’’ J. Acoust. Soc. Am. 91, 3540–3550 (1992)].
Musicians’ tendency to stretch larger-than-octave melodic intervals
Authors: Rakowski Andrzej
Published: October 2005

Four musicians experienced in identifying musical intervals, but not possessing absolute pitch, tuned a pure-tone oscillator in individual sessions to obtain various melodic intervals with a standard pure tone 500 Hz. Within a single task the standard and variable tones were interchanging until the intended value of a melodic musical interval was obtained. Stimuli were presented via earphones at a loudness level 50 phons. An isosonic filter was used to maintain equal loudness of tones. The variable oscillator was set at either very low or very high frequency at the beginning of each task. The range of 3 octaves above and 2 octaves below standard was investigated and each subject tuned each of the 60 musical intervals within this range 10 times. The sequence of tunings within each octave was quasirandom. The results are presented as mean deviations of each interval from its equally tempered value. The dispersion of results as well as intersubject and intrasubject variability is shown. A tendency appears toward stretching large melodic intervals; they are stretched the more the larger they are. Maximum stretch across 5 octaves investigated is about 1 semitone. [Work supported by the Polish National Committee for Scientific Research.]
Real-time computer simulation of concert hall acoustics: Arbitrary geometries
Authors: Kuyel Turker , Hixson Elmer L. , Pinkston Russell
Published: October 2005

Artificial reverberation is a challenging application of the computer technology in the fields of musical acoustics, architecture, multimedia, or even in home audio. Real-time operation and high perceptional quality are the two challenges for artificial reverberator design. In this paper, the theory and the implementation of ‘‘real-time computer simulation of concert hall acoustics’’ is discussed. Efficient real-time algorithms based on theoretical models have been built. The memory and computational requirements of these algorithms have been determined. Successful implementation of these theoretical models and software algorithms have been implemented in UT Computer Music Studios using ‘‘Accelerando’’ audio processor. ‘‘Accelerando’’ implementation assumes arbitrary room geometries for the early reverberation response, and rectangular room geometries for the late reverberation response. Extensions of the model and algorithms are made to enable the effective simulation of the late reverberation responses of concert halls with arbitrary geometries. For real-time implementation of the extended algorithms, MasPar MP1 massively parallel computer is used.
Noise: Progress report and discussion on the continuing activity on ASA’s role in noise and its control
Authors: Hoover Robert M.
Published: October 2005

A discussion meeting is being sponsored by the Technical Committee on Noise to review progress made to date on the actions initiated by the Technical Committee on Noise at the Denver 1993 meeting to increase the role of the ASA in noise and its control. Members of the steering committee will each review the specific activities undertaken in the areas of education, collaboration with other societies, increasing public awareness of noise and the establishment of a task force to determine feasibility of establishing an ASA clearinghouse on noise.
Cavitation in water generated by pulse Doppler ultrasound
Authors: Roy Ronald A. , Fowlkes J. Brian
Published: October 2005

A recently published study reports the direct observation of transient microcavitation in water produced by clinical diagnostic ultrasound [Holland etal., IEEE UFFC 39, 95–101 (1992)]. Results are presented from a similar study in which an aqueous suspension of 0.2-μm-diam polystyrene spheres was insonated by an ATL UM9 scanner operating in the pulse Doppler mode at 3.5 and 5.0 MHz. The suspension was degassed to approximately 90% of saturation. Using a computer automated active cavitation detector [Roy etal., J. Acoust. Soc. Am. 87, 2451–2458 (1990)], cavitation production rates were obtained for a variety of pulse parameters such as intensity, peak negative pressure, pulse length, and pulse repetition frequency. Results suggest a cavitation threshold which is fairly well predicted by the mechanical index calculated from the various pulse waveforms obtained insitu. [Work supported by NIH through Grant No. RO1 CA39374.]
Anatomic development of the vocal tract: Implications for speech motor control and acoustic properties of speech
Authors: Kent Ray D.
Published: October 2005

It has been proposed that models of speech production should be developed with consideration of age and gender differences. Although some important age differences in anatomy and acoustic patterns of vocalization have been described between infants and adults, much less attention has been given to the anatomic development of the speech production system between infancy and adulthood. This talk reviews the anatomic development of the craniofacial, oral, and laryngeal systems of speech production between birth and young adulthood. Implications of the developmental patterns are considered for the ontogeny of speech production, especially speech motor control and acoustic patterns of speech. Consideration also is given to gender and race as they relate to vocal tract anatomy and its development in children. Topics to be reviewed include: craniofacial skeleton, nasopharynx, tongue, lips, and larynx. The overall pattern of development will be summarized for each system and intervals of especially rapid growth will be identified. A major hypothesis to be evaluated is whether growth is harmonious across the major anatomic systems.
Development of skilled speech production in children
Authors: Baum Shari R. , Katz William F.
Published: October 2005

There has been much recent interest in the manner in which children develop mature speech production capabilities. Investigations of both normal speech acquisition and developmental speech production impairments have contributed to our current knowledge base. A consistent finding is that young children’s speech is highly variable, both in its temporal and spectral attributes, and that this variability diminishes gradually with age. However, the exact manner in which articulatory precision emerges (or fails to emerge in speech disorders) continues to be the subject of debate. For example, some research on anticipatory coarticulation suggests a developmental progression from syllable- to segment-based timing strategies while other research suggests the opposite. Similarly, some studies of compensatory articulation show comparable degrees of motor equivalence in young children and adults, while others show that this aspect of motor control emerges gradually with maturation. This presentation will review key experiments (primarily based on acoustic analyses), consider potential explanations for the controversial data, and discuss methodological alternatives that may assist in the resolution of outstanding questions. Recent evidence indicates that specific motor patterns of different articulators will play a major role.
Evaluations of 1–3 piezocomposite transducers for underwater applications
Authors: Ting Robert Y. , Howarth Thomas R.
Published: October 2005

This presentation begins with discussions on 1–3 piezocomposite transducer designs and how the designs may be tailored for specific underwater acoustic applications. For instance, 1–3 piezocomposite materials with cover plates for actuator applications shall be shown and compared with other 1–3 piezocomposite transducers that feature finer scale piezoceramic rods for wideband underwater acoustic imaging applications. Other 1–3 piezocomposite transducers to be shown include multilayer plates with elements stacked in mechanical series and wired electrically in parallel as well as monolithic structure designs with elements etched into the piezocomposite structure. Underwater electroacoustic evaluations of each of the 1–3 piezocomposite transducers will be presented. The in-water measurements include the transmitting and receiving performances as well as directivity patterns over extended frequency bands. Discussions on how the 1–3 piezocomposite transducers may be applied towards future underwater acoustic systems shall conclude the presentation. [Work sponsored by ONR.]
1–3 composite transducer for partial discharge detection in high-voltage transformers
Authors: Kurusingal Valsala
Published: October 2005

Ultrasonic techniques alone or in combination with electrical methods are increasingly being used for the detection and/or location of partial discharge (PD) in high-voltage transformers and other electrical plants. Ultrasonic transducers used for fault location are traditionally coupled externally on to the transformer wall. Mounting transducers inside the transformer steel tank has certain definite advantages like better noise immunity, increased sensitivity due to better acoustic matching to oil, and less complicated signals (only longitudinal waves). Transformers are filled with mineral oil whose temperature could rise up to 110 °C during normal operation. Also, there are large electric and magnetic fields and strong mechanical vibrations as well inside the transformer. The authors has developed 1–3 piezopolymer composite transducers which are capable of operating in this adverse environment. These transducers were life tested in mineral oil at 130 °C for over 500 days and their kt values monitored as a function of time. Life test and vibration test data for several transducers as well as sensitivity, frequency response, pulse width, and constructional details are presented.
A study of lip vibrations in a trombone
Authors: Copley David C. , Strong William J.
Published: October 2005

More than 50 years ago, Martin investigated lip vibrations in a coronet mouthpiece using stroboscopic photography [D. Martin, J. Acoust. Soc. Am. 13, 305–307 (1942)]. Since then, several researchers have based lip models on Martin’s data. Unfortunately, due to the quality of the photographs, it is difficult to obtain anything more than a limited quantitative description of the lip motion. The purpose of this study is to obtain more detailed photographic sequences and lip motion data on which new models may be based. The trombone was selected as representative of the lip reed family. A computer-controlled fiber optic stroboscope was used to capture the motion of a player’s lips on video. By inserting the optic bundle through small holes drilled in the mouthpiece, lip motion was observed from the front and side for six notes (Bflat2, F3, Bflat3, D4, F4, G4) played at loud and soft dynamic levels. Video sequences and resulting lip motion data will be presented and discussed.
Brass sound simulation with a lip vibration model having two degrees of freedom
Authors: Adachi Seiji , Sato Masa-aki
Published: October 2005

Brass sound simulation is carried out with the use of a two-dimensional lip vibration model, where lips execute both rolling and reciprocating motions. This model allows lips to operate on both the lower and higher frequency sides of the air-column resonance frequencies. Oscillations generated by the total sound production system are on both the lower and higher frequency sides in the first and second resonance modes, while in the higher modes they are realized only on the lower side. The harmonic series selected from these oscillations in the second and higher modes to achieve the optimum pitch turns out to comprise an oscillation on the higher side of the second resonance frequency and oscillations on the lower sides of the higher resonance frequencies. This result closely matches the transition of lip vibration states from the outward-striking vibration at the second mode (i.e., the lowest mode among the ones used musically) to the vocal-cord-like vibration at the higher modes, which is observed in the simultaneous measurement [S. Yoshikawa, submitted to J. Acoust. Soc. Am.] of mouthpiece pressure and lip vibration.
The developmental role of acoustic boundaries in speech perception
Authors: Ohde Ralph N.
Published: October 2005

During production, there are frequent abrupt changes in the amplitude or in the spectrum of the sound, and these variations are regarded as boundaries between speech sounds. The information appears rich in cues for phonetic features of the segments occurring within 10 to 30 ms of the acoustic boundaries. Recent research examining children’s speech shows that the information contained within the acoustic boundaries of their productions provides important cues for consonant place of articulation. In addition, interspeaker variability for these segments is low, as compared to properties such as formant transitions. Current studies also reveal that for both adult and child speech a principal mechanism involved in processing acoustic boundary cues involves short-term memory, and that the elimination of boundary information negatively affects perception of place of articulation, particularly for children’s speech. Furthermore, the importance of acoustic boundaries in children’s speech extends to their perception of place of articulation of both consonants and vowels. Young children accurately process place of articulation from very short duration stimuli, which includes the period of rapid spectrum change. In the current paper, this evidence supporting the salience of acoustic boundary information in children’s speech and children’s perception of speech will be presented. [Work supported by NIH, DC00464.]
Wave propagation in truss structures
Authors: Guo Yueping
Published: October 2005

A wave-based approach is used to study truss vibrations. Each truss member is treated as a wave-bearing system so that the number of unknowns is the same as the number of wave types, independent of the physical dimension and frequency. A global matrix system is then formed to solve the overall problem, which, because of the wave approach used for the truss members, is very sparse and numerically absolutely stable. Thus even large-scale trusses can be studied with computation that is almost trivial. An example is given for a truss with 35 joints, connecting 109 beams, which is also used in our experimental studies. Results are obtained to discuss features such as pass/stop bands, modal density, and energy sharing between different wave types. [Work supported by ONR.]
Changes in sound phenomena of a beam due to rib and weld properties
Authors: Carney Matthew S. , Mann J. Adin , Driesch Patricia L.
Published: October 2005

Experiments will show how rib and weld discontinuities on a beam effect wave propagation and far-field sound radiation. Numerous aluminum and steel beams were constructed using a different rib and weld configuration. The beam dimensions were 5 cm × 0.635 cm × 2.44 m and the ribs had T-shaped cross sections of slightly varying dimensions. The beams were then suspended and sound pressure 2.5 cm from the beam surface was measured with a microphone. By changing the physical construction of the attachment area of the ribs it was possible to isolate and analyze the effects of the different construction parameters. Also, the existence and effect of residual stresses in the beams due to the construction process was investigated. Analysis of the data showed that both the ribs and the welds produced changes in the far-field sound radiation and the phase speed of the propagating waves. Also, the geometry and the means of attachment of the rib affected the sound field. Transmission and reflection coefficients were calculated, along with travel times of the waves, to verify the changes in phase speeds that were found. [Work supported by ONR, Contract No. N00014-93-1-0493.]
Wave propagation in thermoporoelastic plate
Authors: Paul H. S. , Murali V. M.
Published: October 2005

Governing equations for wave propagation of a thermoporoelastic plate are derived. The material obeys the theory of Pecker and Dersiewicz [C. Pecker and H. Dersiewicz, Acta Mech. 16, 45–64 (1973)]. The temperatures of solid and liquid phases are assumed to be different. Due to temperature difference in both phases at every point in the medium, there is a coupling parameter in heat conduction equations. The frequency equation is obtained for stress-free and thermally insulated boundary conditions. Numerical results are calculated for isothermal and adiabatic wave propagation corresponding to kerosene filled sandstone. Phase velocities and the attenuation factor are plotted against frequencies for symmetric and antisymmetric mode. In an isothermal case, the phase velocity oscillates sharply for the symmetric mode but it is not so sharp for the antisymmetric mode. But the behavior of the attenuation factor is reverse. In the adiabatic case, the phase velocity oscillates very less, up to frequency of 2 Hz; afterwards, it oscillates for both modes. In the attenuation factor, there is rapid oscillation throughout for the antisymmetric mode but it is less in the symmetric mode. Oscillation is almost absent between 2.5 and 4.5 Hz in the symmetric mode. The phase velocity is higher for the isothermal case than the adiabatic case whereas it is opposite for the attenuation factor.
A computational model of speech segmentation
Authors: McAngus Todd Neil P. , Brown Guy
Published: October 2005

Recently a computational model of prosody perception based on a multi-time-scale decomposition of the output from a cochlear model has been demonstrated [Todd and Brown, ‘‘A multi-scale auditory model of prosodic perception,’’ ProceedingsoftheInternationalConferenceonSpokenLanguageProcessing (1994)]. This model determines the temporal grouping and prominence of syllables from a speech signal. In this paper we present evidence to show that the model is able to carry out a complete segmentation of a speech signal, from the level of individual phonemes and phoneme clusters up to the phrase and utterance level. Implications for speech recognition are discussed.
Composite wavelet transform as an auditory model
Authors: Trappe Wade , Lakey Joseph D.
Published: October 2005

Motivated by the human auditory system, a new signal transform is presented which models the way humans hear. Cochlear processing acts like a constant bandwidth bank of filters in the low-frequency range but is of proportional bandwidth at higher frequencies. This new transform, which we call the composite wavelet transform, is better able to model this process than standard signal processing techniques such as the short-time fourier transform (STFT) and the continuous wavelet transform (CWT). The composite wavelet transform in fact provides a signal analysis tool that is able to examine signals with competing signal structures whereas the STFT and the CWT do not. Numerical results for this transform are presented along with a comparison to the STFT and the CWT.
Modeling the separation of acoustical signals based on the simulation of spike firing patterns
Authors: Fischer Kyrill A.
Published: October 2005

It is investigated to what extent the separation of overlapping acoustical sounds may be achieved by detecting and grouping comodulated frequency bands. Comodulation is defined by coherent envelope fluctuations which are detected by means of physiological motivated algorithms involving spike representation and coincidence detection. Groups of comodulated frequency bands are defined by cluster analysis and their temporal variations are traced. Every group detected can then be separated from its ‘‘acoustical background’’ by suppressing all channels not belonging to the given group. The paper addresses problems concerning the definition of suppression gain values and the problem of overlapping groups.
Environmental source tracking using measured and modeled replicas
Authors: Perkins John S. , Fialkowski Laurie T. , Collins Michael D. , Kuperman W. A. , Krout Timothy L. , Jannucci Jon , Barrett Ross F. , Kelly L. J. , Larsson Ashley , Hall Lindsay , Fawcett John A. , Hoffmann Anthony
Published: October 2005

Preliminary results will be presented for the second TTCP Environmental Signal Processing Exercise (TESPEX 2), which was conducted in June 1994 in shallow waters north of Darwin, Australia. There were two objectives: (1) demonstrate improvements in environmental source tracking [Collins etal., J. Acoust. Soc. Am. 94, 3335 (1993)] (a single-frequency, single-hydrophone synthetic-aperture tracking technique) by relying on a second, horizontally separated hydrophone, and (2) determine the horizontal resolution the environment provides to a vertical array through matched-field processing [Perkins and Kuperman, J. Acoust. Soc. Am. 87, 1553 (1990)]. The approach was to first characterize the region by towing a high-SNR source throughout the region and measure the replica fields using two vertical line arrays separated by several kilometers (one with 32 elements, the other 4 elements), and then to use this data to form the replicas needed to track the source as it traversed the region alone a variety of tracks. The measured replicas are supplemented by modeled replicas; the environment used in the modeling is determined by inverting the measured data. We present preliminary processing results relevant to both objectives, including environmental inversions and tracking with measured and modeled replicas. a)Present address: Scripps Institution of Oceanography, La Jolla, CA 92093.
Performance stability of broadband matched-field processors subjected to sound speed and array-element location error in a shallow-water environment
Authors: Chandler Howard A. , Akundi Murty , Smith G. B.
Published: October 2005

The effects of array-element location errors on the performance of the three broadband matched-field processing algorithms (Bartlett, maximum likelihood, and sector focusing) have been investigated. The KRAKEN propagation model was used to generate the replica acoustic pressure field at multiple frequencies for a shallow-water channel with a depth variable sound-speed profile typical of a midlatitude summer environment. It was also used to simulate a ‘‘detected’’ field due to an acoustic source in the presence of both uncorrelated and correlated (modal) noise. These fields were then correlated using the three algorithms for selected degrees of sound speed and array deformation error. Results will be presented. [Work sponsored by the Office of Naval Research, Program Element 61153N, with technical management provided by NRL-SSC.]
Matched beam processing for geoacoustic inversion of bottom sound-speed profiles
Authors: Yang T. C. , Yates T.
Published: October 2005

A broadband signal (shot) received on a vertical array was processed for geoacoustic inversion of bottom sound-speed profile. The data were beamformed to show the distribution of waterborne and bottom interacting signals. By correlating the (complex) beams containing the bottom interacting signals with the equivalently formed beams for the replica field (matched beam processing), one evaluates the ambiguity function maximum at the source location and searches for the bottom sound-speed profile which yields the highest correlation for the matched beam processing. Different frequency bands of the signal are investigated to evaluate the sensitivity to bottom sound-speed variations. The inverted sound-speed profiles are found consistent (within the resolution of 50–100 m/s) with the sound-speed profiles estimated from (1) core samples and other geoacoustic data at the site, and (2) the arrival time of the Head waves [J. Wolf, J. Acoust. Soc. Am. 94, 1769 (1993)].
A novel mechanism for directional hearing in a parasitoid fly
Authors: Robert D. , Hoy R. R. , Miles R. N.
Published: October 2005

Sound localization is a basic behavioral task of the auditory system. Incident sound waves arrive at the ears and generate interaural differences in time of arrival and in amplitude that are key cues for the computation of sound direction. In small animals, both cues can become vanishingly small, posing a challenge for directional hearing. Yet nearly all animals that hear can localize sound. In the fly Ormiaochracea, the two acoustic sensors are separated by only about 520 μm, and are contained within an undivided air-filled chamber, an arrangement that results in minimal differences in interaural time (<2 μs) and no intensity cues from an incident sound wave. Using laser vibrometry, it is shown that the mechanical response of the tympanal membranes has a pronounced directional sensitivity. Using probe microphones and neurophysiological recording techniques, it is demonstrated that this fly utilizes a novel mechanism for the detection of an incident sound wave. This mechanism relies on the mechanical coupling between the two tympanal membranes. In effect, the fly’s ears operate by mechanical preprocessing, converting interaural acoustic time differences of 2 ms or less into neural time differences large enough to encode in the central nervous system.
Seismic communication in amphibians
Authors: Narins Peter M.
Published: October 2005

The white-lipped frog, Leptodactylusalbilabris, exhibits the greatest sensitivity to substrate-borne vibrations (seismic stimuli) reported to date for any terrestrial animal. Nerve fibers from the sacculus, the source of this extraordinary sensitivity, show clear responses to sinusoidal seismic stimuli with peak accelerations less than 0.001 cm/s2. In addition, this animal generates substrate-borne vibrational signals during calling. As the male’s vocal sac expands, it strikes the substrate impulsively, generating a vertically polarized surface (Rayleigh) wave that is detected by neighboring males. These Rayleigh waves are used as intraspecific communication signals to coordinate chorus behavior in this species. Recently, a particularly unusual behavior has been described for one species of Malaysian treefrog (Polypedates). During nocturnal courtship, females living in dense mats of floating vegetation perch conspicuously on a reed or blade of grass and tap their rear toes rhythmically. Males on neighboring reeds were observed to quickly locate and mate with the tapping female. Thus it is likely that toe tapping functions as a vibrational signal indicating the female’s presence to neighboring males. It is becoming clear that seismic communication and sensitivity to whole-body vibrations are more ubiquitous among the vertebrates than had been previously imagined. [Work supported by NIH.]
Studying the evolutionary history of communication: The ghost of signals past
Authors: Ryan Michael J.
Published: October 2005

The efficiency of communication can be maximized by a tight match between signal and receiver. This view underlies the field of animal communication and has led biologists to propose functional and evolutionary models that are firmly grounded in such linkage. Recent empirical studies have shown, however, that tight linkage between sender and receiver need not be the case. Using the phonotactic response of female frogs to variation in mating calls, it has been shown that some aspects of the receiver are quite broad relative to the conspecific signal. For example, females show preferences for signals of their own species to which are added components from the signals of other species. Also, using algorithms for reconstruction of ancestral traits, it is shown that key stimuli needed for species-specific pattern recognition have not been tightly linked with receiver evolution. These studies suggest a more liberal interpretation for the evolution of animal communication systems.
Using FFT spectrum analyzers to measure animal sounds
Authors: Awbrey Frank T.
Published: October 2005

Sound-pressure levels of animal sounds have been measured in so many different ways that comparison between studies is very nearly impossible. The transient, repetitive nature of most vocalizations means that standard sound level meters may yield very different numbers, depending on meter time constant and frequency weighting. Integrating sound level meters, which average sound over various periods and measure sound exposure level directly, are an improvement, but are hard to use for measuring single, transient sounds such as a frog call or an echolocation pulse. A better way, with brief signals, is to record them through calibrated transducers, along with calibration tones of known amplitude, onto tape recorders with linear response to varying levels. These recordings can then be measured with digital spectrum analyzers in a way that is reliable, accurate, and repeatable. Possible measures include peak, octave, one-third octave, maximum, time-average, sound exposure, and spectrum levels. Comparability requires attention to and reporting of important details, such as averaging times, filter bandwidth and weighting functions, and adherence to standards for measuring and reporting sound.
Hot topics in architectural acoustics
Authors: Siebein Gary W.
Published: October 2005

This lecture will review some of the current topics of interest in the architectural acoustics community. There has been much recent work in trying to determine what physical factors contribute to subtle perceptions of spaciousness and source width in rooms. Efforts to standardize measurement techniques for recently proposed acoustical measures derived from the impulse response of rooms is underway as well as experiments to determine the utility of these measures in applied situations. These measures are also predicted in rooms by tests in physical scale models and computer models. Aural simulation of the effects of the architectural features of rooms on the qualities of music and speech is also being pursued. Measurements of the effects of flanking transmission, plumbing noise, and impact noise are also being refined as are recommended methods for assessing background noise levels in rooms. A significant area of activity in the architectural community that differentiates it from other technical committees occurs through the design and construction of real buildings to meet design goals based on current understandings of sound perception and propagation. Several case studies of actual buildings that have been designed to test some of the emerging knowledge gained in the laboratory will be presented.
Increasing the Acoustical Society’s role in noise control and noise effects
Authors: Brooks B. M. , Dubois T. J. , Hoover R. M. , Maling G. C. , Sutherland L. C.
Published: October 2005

At the meeting of the Technical Committee on Noise at the fall 1991 meeting of ASA in Houston, a discussion was held on how, or if, the Acoustical Society should develop more concrete policies or action concerning noise control and noise effects. As a result of that discussion, a Noise Task Group was formed by the authors at the direction of the Chair of the Technical Committee to explore the issues involved. Since that time, several special sessions have been held to help establish a direction for this activity. This talk will briefly review some of the more important elements of that activity which, properly, are beginning to involve members of other technical committees within the Society in such areas as hearing screening tests at ASA meetings, development of lecture materials for use in presenting talks to youth on acoustics and noise, and potential development of room noise criteria and/or rating schemes for meeting spaces. Other such action areas that have evolved from these special sessions will be mentioned. These offer other opportunities for ASA members to help achieve a goal worthy of the Society—to preserve quiet and tranquility in concert with advances in technology and population growth.