bluuebunny/crossref_metadata_2025 huggingface dataset:
Self-cohering conditionally constrained minimum variance matched-field processing in a random ocean environment
Authors: Krolik Jeffrey L.
Published: October 2005

In recent work, minimum variance (MV) beamforming with sound-speed perturbation constraints (MV–SPC) [J. L. Krolik, J. Acoust. Soc. Am. 92, 1408–1419 (1992)] was proposed as a means of improving the robustness of MV matched-field source localization methods to random sound-speed profile inhomogeneities in the ocean channel. In this paper, an extension of the approach is developed that uses wave-front measurements from a source-of-opportunity to substantially improve MV–SPC beamformer performance. The proposed method consists of MV–SPC beamforming with constraints derived from the statistics of hypothesized point-source wave-front replica vectors conditioned on the measurement of a source-of-opportunity wave front at a particular point in time and/or space. Using apriori knowledge of the space-time correlation of sound-speed inhomogeneities, optimal prediction of the conditional signal wave-front correlation matrix is used to compute more precise MV–SPC beamformer constraints. The method also includes an accurate procedure for estimating a source-of-opportunity wave front in a multiple source environment. [Work supported by ONR.]
Optimal linear array configuration and weighting through simulated annealing
Authors: Zakarauskas Pierre , Greening Michael V.
Published: October 2005

This paper reports on the application of the optimization technique called simulated annealing to the problem of finding the best array configuration for a conventional beamformer. The question the optimization process has to answer is: ‘‘For a given maximum array length and number of sensors, what is the placement and weighting of the elements that will give the narrowest mainlobe while keeping the sidelobes below a given level S.’’ It is emphasized that the technique is fairly general, and that different optimization criteria could easily be substituted. During the search, a cost C is calculated for each randomly generated array configuration and set of weights. The cost increases with the width of the mainlobe, and is proportional to how much the sidelobes are above S. When the search is complete, the array configuration may or may not occupy the whole length that was accorded to it, and the sidelobes are all at or below S. By keeping the part of the cost function that does not change and using table look-up for the transcendental functions, the search can be made quite efficient, taking only on the order of minutes of CPU time on a VAX 9000 computer for 14 receivers.a)Present address: Datavision Computing Services Ltd., 1545 Pandora Ave., Ste. 203, Victoria, BC V8R 6R1, Canada.
Changing the deployment of attention to phonetic structure
Authors: Baldwin Alison K. , Nusbaum Howard C.
Published: October 2005

Previous research on perceptual learning of synthetic speech and non-native phonetic contrasts suggests that listeners shift the focus of their attention from one set of acoustic cues to another, although this hypothesis has never been tested directly. The present study was carried out to determine whether training could alter perception of stop-consonant place of articulation of shifting attention from a transition cue to the spectral shape of the release burst. CV syllables were synthesized with transitions specifying one place of articulation and release burst shape specifying another. One group of subjects was given feedback consistent with the category specified by the transitions; the other group was given feedback consistent with the bursts. Although subjects initially tended to respond based on transition cues, feedback was effective for both groups. Furthermore, for subjects who showed learning, attentional changes also generalized to the same phonetic categories in new vowel contexts. The implications of these results for theories of speech perception and perceptual learning will be discussed.
Perceptual identification of phonologically flapped words
Authors: Charles-Luce Jan , Ragonese Elvira , Lewin Betina
Published: October 2005

Native English speakers originally produced bisyllabic words containing intervocalic alveolar stops, e.g., writer and rider. These words were embedded in both semantically neutral and semantically biasing passages. In addition, both neutral and biasing passages were produced in two conditions. In one condition, a listener was present in the recording room with the speaker. In the other condition, the speaker was alone in the recording room. The production results showed that flapping did not occur in the listener-present condition but that it did in the listener-absent condition. The present investigation examined whether these production results are functional for the listener. All production tokens were excised from the original passages and presented to subjects auditorily for identification. Subjects were presented with pairs of stimuli, e.g., writer and rider. On each trial subjects heard one member of the pair. They were to circle the word that they thought they heard in a response booklet. Percent correct identification results show that subjects were best at correctly identifying the tokens when they had been produced originally in the listener-present, neutral passage and worst at identification in the listener-absent, bias passage. [Work supported by NIH.]
Predicting intensity discrimination from power-law modifications of the loudness function
Authors: Hellman William S. , Hellman Rhona P.
Published: October 2005

Several modifications of Stevens’ power law have been introduced to account for the steepening of the loudness function below about 40 dB SPL. The relation between intensity discrimination (ΔI/I) and loudness is determined for three proposed modifications: (a) Ls=K(I−I0)n, (b) Ls=K(In−I0n), and (c) Ls=k([I+cI0]n−[cI0]n), where I0 is the sound intensity at threshold. Below 15 dB SPL Eqs. (a) and (b) generate a rapidly decreasing function for intensity discrimination, whereas the function derived from Eq. (c) rises with a slope of −1 in log–log coordinates. The slope of the measured intensity-discrimination function is compatible with results predicted for pure tones from Eq. (c) but not with those predicted from Eqs. (a) and (b). [Work supported in part by NIH.]
Loudness adaptation for high-frequency tones
Authors: Miskiewicz Andrzej , Hellman Rhona P. , Meiselman Carol H. , Scharf Bertram
Published: October 2005

Loudness adaptation was measured for pure tones at 4, 12, 14, 16, and 17 kHz. Loudness adaptation refers to the decline in the loudness of a continuous sound as a function of exposure time. More than 100 students judged—by the method of successive magnitude estimation—the loudness of the tones over a 6-min exposure time. Earlier investigations for frequencies up to 4 kHz had shown that loudness adapts only at sensation levels up to approximately 30 dB [B. Scharf, Hear. Res. Theory, 1–56 (1983)]. The current measurements show that at 12 kHz and above, loudness adapts at sensation levels above 30 dB as well as below. At high frequencies, loudness adaptation becomes greater as frequency increases up to 16 kHz; at 17 kHz adaptation is nearly the same as at 16 kHz. The marked loudness adaptation of steady tones at very high frequencies and relatively high sensation levels is ascribed to a restricted spread of excitation in the auditory system resulting from the steep rise of the threshold curve at the upper bound of hearing. [Work supported by NIH.]
Equal sensation functions in bilateral sensorineural-impaired hearing
Authors: Hellman Rhona P. , Meiselman Carol H.
Published: October 2005

Equal sensation functions were generated for individuals and groups with hearing losses ranging from 39 to 73 dB at the stimulus frequency. The functions were obtained directly from conventional loudness matching, and indirectly from magnitude estimation, magnitude production, and cross-modality matching. The results of the direct and indirect loudness-level determinations lie within a relatively narrow band. On the average, the intrasubject variability across procedures ranges from 4 to 6 dB. The corresponding intersubject variability has a mean value of 4.6 dB. Moreover, cross-modality matches obtained for 36 listeners exhibit temporal stability for intersession intervals as long as 5 years. Loudness functions constructed from the data are better described by Zwislocki’s [HandbookofMathematicalPsychology, 3–97 (1965)] proposed loudness equation than by either of the more widely applied power-law modifications. [Work supported by the Rehab R&D Service of the VA and by NIH.]
Changes in distributions of fodrin and F-actin in the cochlea after noise exposure as revealed by confocal microscopy
Authors: Spongr Vlasta , Attanasio Giuseppe , Henderson Donald
Published: October 2005

Changes in distributions of two cytoskeleton proteins, 240 kD alpha fodrin and filamentous actin, were studied in chinchillas exposed to 125 dB SPL impact noise and sacrificed immediately or 24 h after the exposure. To preserve the physiological integrity of the organ of Corti as much as possible, the surface preparations were studied using confocal fluorescence microscopy by serial, noninvasive optical sectioning. Immediately after noise exposure the damaged OHCs showed staining at the cuticular plates for both proteins, which varied in intensity. However, the lateral surfaces of the hair cells were not labeled. After 24 h, scars have replaced the missing cells and intense labeling of F-actin was seen in the Deiter’s cells bodies and phalangeal processes. The results will be discussed in terms of role of fodrin and F-actin in the pathology induced by noise exposure. [Work supported by NIDCD 1R01DC01237-O1A1.]
Power flow analysis for structural finite element models
Authors: Hambric Stephen A.
Published: October 2005

Research on calculating mechanical power flow quantities (steady-state structural intensities and power dissipation) in finite element models from dynamic forces and velocities output by the NASTRAN computer program is summarized. Mechanical power formulations for the axial, torsional, and bending motion of beams and the membrane and bending motion of plates are presented. The formulations have been implemented in the program McPOW (Mechanical POWer) [S. A. Hambric, J. Vib. Acoust. 112, 542–549 (1990)], which functions as a post-processor to NASTRAN. Results are shown from a study comparing experimental and finite element flexural power measurements for a point excited straight beam with compliant end mounts. The good agreement of the experimental and numerical power spectra serve to validate the finite element mechanical power method. Also, animations of the power flow response of a cantilever plate model over a frequency range encompassing several low- to mid-order flexural and membrane modes were performed on graphics computers and recorded on videotape, and will be presented to show the benefit of visualization techniques in understanding mechanical power analyses.
Laser vibrometer methods for measuring far-field structural intensity
Authors: McDevitt Timothy E.
Published: October 2005

There is a need to develop measurement techniques that can quickly and accurately provide structural intensity data over an entire surface without interfering with the structural response. To this end, methods have been developed to measure the flexural intensity in a reverberant beam by using scanning laser vibrometry; kinetics and standing wave methods have been employed with a single scanning vibrometer while a finite difference method has been employed with a two-channel scanning vibrometer. Experimental results for the first two methods, which each provide a single value of intensity on the beam, are compared with experimental results for the finite difference method, which was used to provide values of intensity at various positions along the beam. Similarly, experimental data on intensity in longitudinal waves along a rod are also presented for the vibrometer(s) set in the differential Doppler mode. Results indicate that flexural intensity is measured very consistently by the three methods and that longitudinal intensity measurements are feasible if the longitudinal motions, induced by bending waves, are taken into account. [Work supported by ARL Penn State.]
Active vibration control using structural intensity
Authors: Sommerfeldt Scott D. , Hayek Sabih I. , Schwenk Allan E.
Published: October 2005

The subject of structural intensity has been investigated by a number of researchers over the last few years, as a means of gaining an increased understanding of the energy in the structural field. In the area of active control, structural intensity is also of interest for applications that involve energy propagation in the structure. For such cases, controlling the structural intensity, as opposed to the structural vibration, has the potential of providing more effective control of the energy in the structural field. Initial results obtained by implementing an active control system based on structural intensity are presented. The results indicate the potential of obtaining improved control of the structural field, particularly for the case where the sensors are positioned in the near field of the actuators. Some of the issues involved with implementing a control system based on structural intensity are also discussed.
The effects of speaking rate and stimulus variability on the perception of spoken words by young and elderly subjects
Authors: Sommers Mitchell S. , Humes Larry E.
Published: October 2005

This study compared spoken word recognition in young and elderly normal-hearing listeners as a function of speaking rate and stimulus variability. Monosyllabic words were presented in quiet at one of three speaking rates (fast, medium, or slow). In one condition, all stimulus items were presented at the same speaking rate while in a second condition speaking rate was varied from trial to trial. In control conditions, items were presented at either fixed or varying overall amplitudes. For young listeners, spoken word identification did not differ as a function of speaking rate. Older subjects, in contrast, demonstrated similar identification scores for slow- and medium-rate items but had significantly reduced identification performance for words presented at fast speaking rates. Identification accuracy was also reduced for older listeners in the mixed—as opposed to single—rate condition but only for those items presented at the fast speaking rate. Young listeners were not affected by variability due to speaking rate and neither group showed reduced identification as a result of variability in overall amplitude. Implications for speech perception in older listeners are discussed in terms of current views about perceptual normalization for different sources of variability. [Work supported by NIH.]
Drops in space: Spacelab experiments on surfactant-bearing drops
Authors: Holt R. Glynn , Tian Yuren , Zheng Xiaoyu , Apfel Robert E.
Published: October 2005

Utilizing a dedicated acoustic positioning apparatus called the Drop Physics Module (DPM) aboard the First United States Microgravity Laboratory (USML-1) mission, single surfactant-bearing drops were excited into their quadrupole resonance and observed during free decay. Surface shear and dilatational viscosity and Gibb’s elasticity are inferred from the frequency and damping measurements using a modification of a previous theory [H.-L. Lu and R. E. Apfel, J. Fluid Mech. 222, 587 (1991)]. Measurements were performed for various combinations of surfactant type, concentration, and drop size. Results from ground-based experiments in ultrasonic fields will also be presented and compared. [Work supported by NASA through JPL Contract No. 958722.]
The effect of compressibility on acoustical streaming near rigid boundaries
Authors: Qi Quan
Published: October 2005

The effect of compressibility on acoustic streaming near rigid boundaries is discussed for plane traveling, as well as, for plane standing waves. The investigation is motivated by the fact that neglecting compressibility in the study of acoustic streaming near a rigid boundary lead to an incomplete description of the streaming velocity field both inside and outside the boundary layer. It is shown that an improved description may be obtained by including the effect of compressibility in the analysis of flow field in the viscous-acoustic boundary layer. The resulting acoustical streaming velocity outside the boundary layer is shown to depend not only on the amplitude of the primary wave, but also on the Prandtl number and the ratio of the specific heat of the fluid medium. Calculations based on typical numerical data suggest that the streaming velocity outside the boundary layer is larger than the classical results obtained with the incompressible fluid assumption. Relevance to the on-going study of ultrasonic cleaning will also be briefly mentioned. [Work supported by the Hunt Fellowship.]
Comparison of adaptive procedures for obtaining psychophysical thresholds using computer simulation
Authors: Hawley Monica L. , Colburn H. Steven
Published: October 2005

In order to compare the efficacy and efficiency of various adaptive psychophysical procedures, a program that simulates these procedures has been developed and used to compare several adaptive methods for three experiments: binaural detection, interaural intensity discrimination, and interaural time discrimination. The simulation program takes as input a psychometric function, which specifies the probability of a correct response for any value of the adapting variable. A threshold is generated by choosing the next value of the variable according to the rules of the procedure being simulated, including the rules for stopping and calculating the threshold estimate. The software generates distributions of thresholds and statistical analyses are performed, such as bias and variance of the obtained distribution. The program has been applied to psychometric functions appropriate for normal-hearing and hearing-impaired listeners for each of the three experiments noted above, and has compared a number of adaptive procedures including the Transformed Up-Down (such as 2-down-1-up), PEST, BUDTIF, and Weighted Up-Down methods. [Work supported by NIDCD and Dept. of Education.]
A comparison of complex vibrotactile pattern perception on the OPTACON by young and old observers
Authors: Cholewiak Roger W. , Collins Amy A.
Published: October 2005

College-aged and, more recently, senior observers are being surveyed for the ability to identify and recognize complex vibrotactile patterns. These patterns are presented to the left index fingertip on the 144-pin vibratory array from the OPTACON, a reading machine for the blind, in a ‘‘static’’ mode in which all elements that spatially define the pattern are turned on for its 26-ms duration. Pattern complexity ranges from simple line segments, identified only by number, to the 26 letters of the alphabet. The ability of these observers to discriminate between two simple patterns is first measured. They are then trained to identify these patterns, and are tested in a masking situation. Observers are then trained to identify a larger set of abstract patterns, and finally the alphabet. In addition, two measures of basic sensitivity are taken on both groups: threshold at ten different frequencies (10–400 Hz) of sinusoidal vibration, and the growth of magnitude for suprathreshold levels of four of these frequencies. Comparisons will be discussed. [Work supported by NIDCD.]
The dynamics of musical structure formation: Strange attractors, orbit formation, and the topology of musical structure
Authors: Casadonte Donald
Published: October 2005

Since the time of the ancient Greeks music theorists have dreamed of a ‘‘musical space’’ in which musical events reside. In this paper the complete phase space portrait of a piece of music, Bach’s TwoPartInventionNo. 4 in D Minor is reconstructed using techniques of dynamical systems theory. The space consists of a central strange attractor surrounded by wandering orbits that continually ‘‘revisit the attractor as the musical structure develops. This simple example is taken as a paradigm for more general types of music structure, in which stable melodic lines are topologically represented as orbitally stable subsets in phase space (most commonly as strange attractors), and the transitional or nonmelodic sections are topologically unstable or wandering subsets. Musical structure is generated as the musical ‘‘trajectories’’ contained in the different simultaneous musical lines of a composition interact via the stable and unstable regions of the musical phase space structure (ω-limit structure) defined by the composer. This ω-limit hypothesis is illustrated with several classical music examples. The implications of time-evolutionary pictures of the unfolding of musical structure and the topological process involved in the composer-shaping of melodic contours are discussed.
The effects of training and predictability on the discrimination of sequences of tones with various frequency ranges
Authors: Barsz Kathy , Jariwala Maya , Peter Jeffrey E. , Scism Melissa A. , Barnauskas Larissa A. , King Shannon M. , McCarthy Margaret M.
Published: October 2005

Listeners discriminated sequences: two different higher frequency tones alternated with two different lower frequency tones, each 200 ms. Sequences began at a random point and repeated without pause. Predictability was manipulated by holding the standard sequence constant (PREDICTABLE) or by varying it (UNPREDICTABLE). The frequencyrange between the higher and lower tones was two octaves (WIDE) or a minor second (NARROW). Medium range sequences (a minor third) were presented in two contexts: with the wide range sequences (MED/W) or with the narrow range sequences (MED/N). Sequences differed only in the relationship between the higher and lower tones. Fifty listeners each discriminated the sequences in one condition (MED/N or MED/W in either a PREDICTABLE or UNPREDICTABLE block); ten returned for two extra sessions (practice). Discrimination of the wide range sequences was at chance and only improved with practice when the block was predictable, although listeners still had a difficult time. Performance on this task was affected by frequency range; performance on the narrow range sequences benefited from practice and improved predictability There was a clear practice effect for the medium sequences, but only when the block was predictable and they were presented with the narrow range sequences.
Perceptual organization of temporal patterns in music
Authors: Takeuchi Annie H.
Published: October 2005

The temporal structure of music is based on meter, a hierarchical structure in which each level consists of periodic alternation of strong and weak pulses. This experiment examined the extent to which listeners have internalized metrical hierarchies and use them to perceptually organize temporal patterns. Musically experienced subjects rated how well probe tones that varied in temporal location and in loudness (accent level) fit into simple metrical contexts. Whereas Palmer and Krumhansl [J. Exp. Psychol: Human Perc. & Perf. 16, 728–741 (1990)] reported that listeners rated such probe tones on the basis of metrical location (whether they occurred on accented beats, unaccented beats, or between beats), in this experiment, the accent level of tones interacted with metrical location in determining probe tone ratings. Accented probe tones were judged better fitting when they occurred on accented beats than on unaccented beats or between beats. Unaccented probe tones showed the reverse pattern. Most subjects organized the patterns according to a three-level hierarchy for triple meter and a four-level hieracrchy for quadruple meter. [Work supported by NSF.]
Segmentation and stress in the rhythmic structure of music and speech: A wavelet model
Authors: Todd Neil P. McAngus
Published: October 2005

The rhythm of a piece of music or a poem can be described by hierarchical tree structure. A new application of wavelet analyses to rhythm in music and speech is described whose result is a hierarchical structure [N. P. McAngus Todd, J. Acoust. Soc. Am. 92, 2380 (A) (1992)]. This structure has two complementary components that are obtained from the projection of loci, in the frequency-time plane, of zero crossings of the slope of the energy flux surface in the energy-time plane: (a) a segmentationstructure, corresponding to positive second derivatives at the zero crossings; (b) a stressstructure, corresponding to negative second derivatives at the zero crossings. The value of the stress is given by the second derivative, which has a ‘‘Mexican hat’’ impulse response. These structures resemble the grouping/metrical structures of the theory of F. Lerdahl and R. Jackendoff [AGenerativeTheoryofTonalMusic (MIT, Cambridge, MA, 1983)]. Example analyses of musical and speech performance are demonstrated for this method and evidence is provided that the resultant structures have a reasonable perceptual basis. The possibility of using this technique for the synthesis of rhythm using a cascade structure is discussed.
Functional activation of right temporal and occipital cortex in processing tonal melodies
Authors: Zatorre Robert J. , Evans Alan C. , Meyer Ernst
Published: October 2005

Cerebral blood flow changes were measured to examine the cortical basis for perceptual analysis of musical information. Twelve normal right-handed volunteers underwent positron emission tomography (PET) scanning using the bolus water technique while listening to two types of stimuli: (1) unfamiliar tonal melodies, and (2) sequences of noise bursts, acoustically matched to the melodies for duration, intensity, and onset/offset characteristics. Subtracting the latter from the former PET image provides information regarding melodic processing beyond that associated with nonspecific auditory stimulation. Anatomical localization was provided by superimposing the average PET subtraction image on the average magnetic resonance image for the subjects tested. Results indicated significant increases in blood flow while listening to melodies in the right associative auditory cortex, suggesting that specialized neural systems in the right temporal lobe participate in higher-order perceptual analysis. In addition, regions of significant activity were observed in the right occipital lobe, within areas classically described as associative visual cortex. This result may implicate these regions in hitherto unexplored aspects of auditory processings, or may possibly reflect engagement of visuospatial processing mechanisms in perception of melodies.
Acoustics for space applications: A progress review
Authors: Wang Taylor G. , Anilkumar A. V.
Published: October 2005

A wide spectrum of containerless science experiments can be performed in a microgravity environment by employing the levitation and manipulation capabilities of acoustics. Acoustic radiation forces can be used to position, rotate, oscillate, and deform liquid samples; and thereby help perform controlled experiments. These aspects have been successfully demonstrated in space with the Drop Dynamics Module (DDM, May 1985), and the more recent Drop Physics Module (DPM, June 1992). The mastering of these noninvasive manipulation techniques opens the way for future drop physics and containerless materials processing experiments. In the latter context, the acoustic method is useful in the control of any molten material, regardless of it being conducting, or charged, or not.
Gap detection in young and old adults
Authors: Schneider Bruce A. , Pichora-Fuller Margaret Kathleen
Published: October 2005

Threshold for detecting a gap between two Gaussian-modulated, 2-kHz tones (σ=0.5 ms) were determined in young and old listeners. The gap detection thresholds of old subjects were more variable and about twice as large as those obtained from young subjects. Moreover, gap detection thresholds were not correlated with audiometric thresholds in either group. Estimates of the width of the temporal window of young subjects, based on the detection of a gap between two tone pips, were smaller than those typically obtained when a relatively long duration pure tone is interrupted [Moore etal., J. Acoust. Soc. Am. 85, 1266–1275 (1989)]. Because the amount of time it takes to recover from an adapting stimulus is likely to affect gap detection thresholds, smaller estimates of temporal window size would be expected in this paradigm if the amount of adaptation produced by the first tone pip was negligible. The larger gap detection thresholds of old subjects indicates that they may have larger temporal windows than young subjects.
Measurements and analysis with SYSid
Authors: Puria Sunil , Allen Jont B. , Elko Gary W. , Jeng Patricia S.
Published: October 2005

To make measurements a person sometimes has the need for many different test instruments on ones bench. The philosophy behind SYSid (SYStemidentification) is to integrate many commonly used test instruments into a single yet accurate and easy to use system. The SYSid software package runs on a DOS platform with the Ariel DSP-16+installed. The theory of operation in SYSid is simple. SYSid excites the system being measured with a stimulus and synchronously averages the measured response of the system. The stimulus can be a chirp, single tone, or multiple tones, MLS, or it can be user-defined (i.e., speech, pink noise, octave-band noise, etc.). It is important to synchronously average a system response in order to obtain accurate phase information. Synchronous averaging also allows one to measure a signal that is below the system noise floor. SYSid then uses FFT techniques to deconvolve the stimulus from the measured response and to further analyze the data. From this basic mode of operation SYSid can perform many types of analyses including phase responses, group delay, impulse response, Hilbert envelope, reverse energy time curve, RT60, waterfall displays, electrical impedance, etc. In addition to these linear measurements, SYSid also provides the capabilities to make distortion measurements due to nonlinearities in the system. These include harmonic distortion, intermodulation distortion, THD+N, input–output functions, and spectral contamination.
A new in
situ method to measure the directivity index
Authors: Elko Gary W.
Published: October 2005

The directivity index is a considerably useful measure in quantifying the directional properties of electroacoustic transducers. For microphones used in reverberant environments, the directivity index gives a rough estimate of the gain in signal-to-noise ratio. The measurement of the directivity index of a transducer is complicated by the need to measure over a spherical surface that is in the far field of the transducer. This talk will present a new method for measuring the directivity index in a diffuse reverberant environment. The method uses the frequency response spectral variance to estimate the distance from a point source to a point receiver in a reverberant field. If the source-to-receiver distance, reverberation time, room surface area, and volume are known, the directivity factor can be calculated explicitly from the spectra variance. The method and its limitations will be discussed in the talk.
Finding the direction of a sound source using a vector sound-intensity probe
Authors: Hickling Robert , Wei Wei , Raspet Richard
Published: October 2005

Good spatial resolution of a sound source can be achieved with the four closely spaced transducers in a vector sound-intensity probe. In contrast, a receiving array, generally requires an extensive distribution of transducer elements. Experimental results are presented that demonstrate the ability of a vector probe to determine the direction of a sound source in water, both without and with noise interference. Measurements of the sound-intensity vector, or sound-power flux, were performed using the cross-spectral relation. Spectral subtraction and averaging over the frequency range of a source, were found to be effective tools for determining the direction of the source when there is interference and background noise.
DISC project: Auralization of virtual environments and the DIrectional
Scattering Coefficient
Authors: D’Antonio Peter
Published: October 2005

Accurate prediction of a room’s impulse response requires the appropriate directional scattering coefficients, as well as computational algorithms, which utilize these data in modeling programs. The directional scattering coefficients define how sound interacts with the various absorbing, reflecting, diffusing, and diffracting surfaces in a room, as a function of the incident direction and frequency, sample size, mounting, and observation direction. Unfortunately, these data and algorithms do not exist. The acoustics community is now preparing to meet the challenge of augmenting random-incidence absorption coefficients and establishing a database of DIrectionalScattering Coefficients (DISC) for all commonly used architectural materials, in a new basic research initiative called the DISC project. The goals of the DISC project are as follows. (1) Design and implement computer-controlled automatic testing instrumentation and facilities to experimentally map the full-spectrum backscattered hemisphere for all commonly used architectural materials for all angles of incidence and various sample sizes and mountings. (2) Develop a geometrical convolution algorithm that can be used with the image model to utilize directional information and provide n-bounce impulse responses, which agree with experimental measurements within accepted error limits, as judged by auralization test. (3) Conduct subjective perception auralization studies with headphones and in an acoustically optimized loudspeaker listening environment.
A rapid method for measuring tube attenuation
Authors: Chu W. T.
Published: October 2005

It has long been recognized that the acoustic impedance is the chief characteristic of the performance of acoustical materials. The impedance tube provides an easy way of measuring this characteristic at normal incidence. It is also known that the effect of tube attenuation must be included in the impedance tube analysis for accurate measurements. Although theoretical prediction can be deduced from the well-known Helmholtz–Kirchhoff formulation, disagreements with experimental results (up to 15%) have been reported. Previous results relied on locating and measuring the pressures as a number of maxima and minima along the standing wave pattern inside the tube. The method was tedious. A more rapid method for measuring the tube attenuation is possible based on the recently proposed three microphone method for impedance tube measurements [W. T. Chu, J. Acoust. Soc. Am. 92, 2315(A) (1992)]. The procedure will be presented together with results from both computer simulations and actual experiments.
Bioacoustic transient detection by image convolution
Authors: Mellinger David K. , Clark Christopher W.
Published: October 2005

A method is presented for detecting bioacoustic transient signals. The method detects a common feature of animal calls, frequency sweeps, or combinations of frequency sweeps over time, by convolving a spectrogram image with a two-dimensional kernel designed to find lines in images. This kernel has positive (excitatory) regions to achieve strong convolution operator response on desired frequency sweeps, and negative (inhibitory) regions to reject noise and interfering sounds. Design parameters control permissible variation in the sweeps. This technique is useful in screening long recordings for desired call types and perhaps for identifying individuals. For screening, it is compared with two other approaches on recordings of Bowhead Whale calls in a noisy Arctic environment: a hidden Markov model and a matched filter. This method performs better than both others on the Bowhead data, perhaps well enough for practical use. It also detects perfectly 15- to 20-Hz calls of unknown origin, possibly blue whales, in a moderately noisy environment. For identifying individuals, tests of spotted dolphin ‘‘signature whistles’’ are underway; it is hoped that similar bottlenose dolphin whistles will soon be tested.
Transmission beam pattern of a false killer whale
Authors: Au Whitlow W. L. , Pawloski Jeffrey L. , Cranford Ted W. , Gisner Robert C. , Nachtigall Paul E.
Published: October 2005

The echolocation transmission beam pattern of a false killer whale (Pseudorcacrassidens) was measured in the vertical and horizontal planes. A vertical array of seven broadband miniature hydrophones was used to measure the beam pattern in the vertical plane and a horizontal array of the same hydrophones was used in the horizontal plane. The measurements were performed in the open waters of Kaneohe Bay, Oahu, Hawaii, with the whale performing a discrimination task. The width of the beams in both planes were similar to those of the Atlantic bottlenose dolphin (Tursiopstruncatus), and broader than those of the beluga whale (Delphinapterusleucas). The major axis of the vertical beam was directed slightly downwards between 0° and −5°, in contrast to the +5 to 10° for Tursiops and Delphinapterus. The beam in the horizontal plane was directed forward. Differences in the fatty structure of the melon of Pseudorca, Tursiops, and Delphinapterus could explain differences in the elevation angle of the vertical beam axis. a)At SAIC, Kailua, HI 96734.
Living sounds from the sea
Authors: Walkinshaw Helen M.
Published: October 2005

In mid-1951, with initial support by ONR, Bell Telephone Laboratories began an exploratory program of undersea acoustics research that lasted some 40 years. As a result, a diverse collection of insitu recordings of marine life slowly accrued from chance encounters during other planned experiments. Most have been identified as marine mammals—various species of whales and porpoise—a few remain unknown even now. The precise mechanism for producing certain very low frequency sounds (20 Hz and below) also remains unclear. The original recordings were made on several bottomed and off-bottom suspensions of hydrophones with effective bandwidths of from 5–400 Hz and 10–1000 Hz. Their installation depths ranged from 220 to below 2000 fathoms, cable-connected to shore, enabling continuous monitoring of the undersea environment. Four samples are presented. Locale and circumstances of their acquisition are described, as are geographic scope and observed seasonal occurences. Copies of the original recordings are available for study.
Measurement and analysis of the two-peak resonance of fish swimbladders
Authors: Zhou Wen-xu , Rogers Peter H.
Published: October 2005

The object of this research is to study the frequency response of the two-chambered goldfish swimbladder. An ultrasonic system was devised to measure swimbladder frequency response invivo in a noninvasive, nonintrusive manner. Measurements were also made for oscars which have a single-chambered swimbladder. Unlike the oscar, two peaks often appear in the response curves of goldfish. To explain this two-peak resonance, an experimental model and an analytical model have been developed. The experimental model employs two gas bubbles to simulate the swimbladder. The experimental results show that the interaction between the two chambers of the goldfish swimbladder causes the two-peak phenomenon. The analytical model treats an oscillating system with two degrees of freedom. The parameters for the analytical model are related to the physical parameters of the system by theory. From comparison of the experimental and theoretical models, it is found that the damped-spring-mass system with two degrees of freedom reproduces what has been observed in the two-bubble experiment and does a reasonable job of modeling the two-chambered goldfish swimbladder.
Automatic classification of labial and alveolar stop consonants
Authors: Zierten Stephanie , Espy-Wilson Carol Y.
Published: October 2005

This study investigates the automatic classification of labial and alveolar stop consonants. The algorithm implemented was proposed by Lahiri etal. [J. Acoust. Soc. Am. 76, 391–404 (1984)], and uses relative and dynamic measures relating the spectral shape at the burst with that of the onset of the following vowel. The purpose of the measure is to capture that (1) for labials, the difference in energy between the burst release spectrum and the vowel spectrum is either less at low frequencies than at high frequencies or the difference is compatible in both frequency ranges and (2) for alveolars, the difference in energy between the burst release spectrum and the vowel onset spectrum is less at high frequencies than at low frequencies. To make these distinctions, the algorithm draws slope lines on the burst and the vowel onset spectra based on F2 and F4 and computes a ratio of the difference (in dB) of the slope lines at 3500 and 1500 Hz. In the automatic procedure, F2 and F4 are picked from LPC spectra based on the covariance method. Preliminary results obtained from word-initial stops consonants occurring in monosyllabic words show that the algorithm does classify correctly many of the stop consonants. However, an analysis of the errors suggests the need for refinements. In particular, the alveolars classified as labials had less energy than the vowel between 0 and 4000 Hz (falling slope line), but more energy than the vowels at frequencies higher than 4000 Hz. In the case of labials, the energy difference at low frequencies was not less than that at high frequencies, but, the labials did have comparable or less energy to that of the vowel at frequencies above 4000 Hz. The refinements suggested by these findings will be made and results will be presented.
Strident-feature extraction in English fricatives
Authors: Bitar Nabil N. , Balian Armen Y. , Chandra Vinay
Published: October 2005

An automatic method was developed to classify fricatives as strident/nonstrident. The method is based on relative energy measures in the frequency bands (2–4 kHz, 4–6 kHz, and 6–8 kHz). Preliminary results based on 1176 fricative sounds extracted from sentences in the TIMIT database showed that most of the fricatives /š ž s z/ were classified as strident and most of the fricative /f v θ (hooked schwa) were classified as nonstrident. An analysis of the remaining fricatives showed that their acoustic realizations differed substantially from their canonical form. The typically strident fricatives did appear to be ‘‘weak’’ and the typically nonstrident fricatives did appear to be ‘‘strong.’’ Factors influencing the change in the manifestation of these fricatives are phonetic context and stress. A detailed analysis of the observed variability will be presented.
Methods for separating adjacent sounds with the same manner of articulation
Authors: Balian Armen Y. , Bitar Nabil N.
Published: October 2005

A difficult problem in automated time alignment is the separation of neighboring sounds with the same manner of articulation, such as two adjacent vowels or two adjacent fricatives. Usually, a broad classifier will find only one region that contains both sounds. In this study, the automatic detection of acoustic events that separate such sequences is investigated. For example, an algorithm based on abrupt changes in the energy between 2000 and 4000 Hz separates /v/’s and /š/’s, which are adjacent to /(hooked schwa)/’s or /s/’s Preliminary results based on 38 cases of adjacent fricatives taken from the TIMIT database show that 68% of the automatically placed boundaries agreed with the labeled boundaries. Of the remaining 12 cases, 67% were considered to be as good as, or even better than the TIMIT labeling. The fricative separation algorithm is presently being refined and tested across a larger database containing other similar sounds. Methods for separating adjacent stops and adjacent vowels will also be presented.
Measurement and in
situ calculation of room acoustic parameters
Authors: Jacobsen Torben
Published: October 2005

Room acoustic parameters should describe the subjective evaluation of a room. The most important parameter for evaluation of acoustic comfort in a room is the reverberation time, but additional parameters are necessary for evaluation and design of concert halls and auditoria. Beside the reverberation time and EDT (early decay time) it is possible from the integrated impulse response to calculate acoustical parameters such as ‘‘early to late sound index’’ (clarity), ‘‘early energy fraction’’ (deutlichkeit), ‘‘center time,’’ and ‘‘total sound level.’’ These measurements and calculations are today possible to perform insitu by a sound level meter with application modules connected to a sound source and a lap-top computer; traditionally these measurements required a much bigger measurement system. Therefore, by means of this easy portable instrumentation, the results may be inspected in the field and additional measurements can be done if necessary. During the presentation, the above acoustical parameters are mentioned and measurement results, obtained by instrumentation from Bruel & Kjaer, from a European concert hall and from a hall in the USA are described.
A novel acoustical approach to biomedical thermodynamics
Authors: Sarvazyan Armen , Chalikian Tigran , Breslauer Kenneth J. , Funck Theodor , Belonenko Vladimir
Published: October 2005

The principal thermodynamic quantities that describe biomolecular structures and interactions (ΔG°, ΔH°, ΔS°, ΔCp) traditionally have been obtained indirectly by measuring the temperature dependence of some equilibrium property. A more direct approach for obtaining the relevant thermodynamic data is based on calorimetric techniques. Significantly, these thermodynamic parameters also can be derived by measuring the speed of sound U over a range of pressures and temperatures, since U is a function of the second derivative of the free energy over pressure. This talk will describe a novel approach that employs acoustical measurements for determining energetics of biologically relevant molecules and their interactions with small ligands (e.g., drug–biopolymer interactions). Preliminary results will be presented that illustrate how acoustic measurements allow for a thermodynamic characterization of model biomolecular systems. Advantages and complementary features of the acoustical versus calorimetric approaches for obtaining the requisite thermodynamic data will be discussed. a)Permanent address: Inst. of Biophys., Russian Acad. Sci.
Lithotripter shock wave propagation through layered media
Authors: Jankovsky Joseph , Christopher P. Ted , Everbach E. Carr
Published: October 2005

Recent advances in the computation of shock wave front propagation [P. T. Christopher and K. J. Parker, J. Acoust. Soc. Am. 90, 488–499 (1991)] have allowed the prediction of spark-gap lithotripter waveforms both on and off-axis for a variety of geometries and conditions. Since in both urinary and biliary lithotripsy, shock pulses must propagate through layered human tissues (e.g., fat, muscle, and other soft tissues) it is especially important to understand how waveforms evolve in layered media. Comparisons of computational predictions and experimental results are presented for a laboratory lithotripter of our own construction whose axis of symmetry is vertical. Measurements of waveforms were made using a novel PVDF membrane hydrophone [E. C. Everbach, J. Acoust. Soc. Am. Suppl. 1 87, S128 (1990)] immersed in oil–water layers horizontally stratified by buoyancy forces. The comparisons show sufficient agreement to allow the possibility of optimizing lithotripter reflector geometries so that waveforms could be tailored to accommodate tissue configurations of individual patients.