bluuebunny/crossref_metadata_2025 huggingface dataset:
Helicopter aural detection as a function of reduced main rotor advancing blade tip Mach numbers
Authors: Mueller Arnold W. , Smith Charles D. , LeMasurier Philip
Published: October 2005

Acoustic tests were conducted to study far-field noise levels and aural detection ranges associated with a Sikorsky S-76A helicopter flying at various advancing blade tip Mach numbers spanning the range from 0.586 to 0.875. The responses of a sound jury of 12 persons, located along the flight path, were recorded at the moments in time when the aircraft was first heard. Simultaneously recorded with the subjective data were acoustic and aircraft flight operations data. Additionally, ground impedance characteristics were measured at numerous sites along the flight track. This paper presents a comparison between the normalized aural detection results obtained from the prediction code ICHIN (‘‘I Can Hear It Now’’) and the sound jury measurements. Predicted detection distances, which compare favorably to measured detection distances, are shown to improve by using measured ground impedances in the propagation module section of ICHIN.
Noniterative overlapping domain decomposition method for acoustic wave propagation
Authors: Priolo E. , Seriani G.
Published: October 2005

A noniterative overlapping domain decomposition technique for a linear acoustic wave equation is presented, based on the concept of regionofdependence of a point P. The field computed in P after a time interval Δt depends on the initial values of the points inside this region, whose extension depends mainly on Δt that in turn depends on the time integration scheme. In practice, the original domain is decomposed into overlapping subdomains containing an interface. The overlaps include the region of dependence of the interface. The numerical solution is then computed separately and concurrently on each subdomain. Finally, the fields on each overlap are updated with the values of the field in the underlying domain. The implementation for an explicit scheme is straightforward. For implicit schemes, great advantage can be gained by using a local spatial interpolation. In this work, the spectral element method and the implicit Newmark method are used, respectively, for spatial and time integration. The overlap region is reduced to a maximum of two elements. Numerical results are shown for 1-D and 2-D cases.
Advanced telerobotic interfaces: Background for the design of virtual environments
Authors: Aviles Walter A.
Published: October 2005

Experiences in the design of spatially oriented multisensory human/machine interfaces for telerobotic systems provide valuable background for the design of virtual environments. These interfaces may be viewed as virtual portals into world models with conventional Newtonian physics and minimal computational delay and distortion. As such, they help isolate the contribution and interaction of human/machine interface components in a system where a high-fidelity real-time world model is available. This paper will provide a survey of work performed in the field of advanced telerobotic interfaces and its import for the virtual environment research and development community.
Comparison of propagation models for source localization in a sloping shallow-water environment
Authors: Jackson P. , Wolf J. , Yang T. C.
Published: October 2005

Benchmark tests of propagation models have been conducted by comparing transmission loss of individual phones. While various propagation models often yield the same (range) averaged transmission loss, they differ in the calculation of phases. The difference in the calculated pressure field is examined here in terms of source localization in a sloping shallow-water environment. This study examines the impact on source localization due to boundary reflection loss, bottom bathymetry and profile, and mode coupling which are handled differently by different models. In particular, the coupled normal mode, adiabatic normal mode, and PE propagation codes are examined. The calculated pressure field from individual models are analyzed in terms of the normal mode amplitudes at the receiver site. The difference of mode amplitude surfaces are plotted as a function of source range to evaluate the range dependence of propagation model mismatch.
Effects of postvocalic voicing on the time course of vowels and diphthongs
Authors: van Santen Jan P. H. , Coleman John S. , Randolph Mark A.
Published: October 2005

Studies of the effects of contextual factors on segmental duration have been mostly restricted to overall duration and relatively few have paid attention to the internal time course of speech segments. However, fine-grained temporal analysis of speech segments is hindered by within-speaker variability and the unreliability of manual segmentation. In this paper, using new methods for reducing statistical ‘‘noise’’ in acoustic trajectories [mappings of the time domain into an appropriate acoustic parameter space, such as formants, cepstra, etc.], analyses are presented of the effects of postvocalic voicing on the internal time course of selected vowels and diphthongs preceding voiced and voiceless consonants in contrasting minimal word pairs [e.g., ‘‘mend’’ versus ‘‘meant’’] are presented. These methods include the computation of the centroid of a set of acoustic trajectories, the average of a set of time warps among minimal word pairs, and various smoothing techniques. The centroid trajectories of a minimal pair generally coincide closely. It was found that lengthening due to postvocalic voicing is nonuniform, with certain regions expanding more than others. The location of maximal expansion could be determined reliably in most vowels and diphthongs studied, and, except for /ay/, occurs towards the end of the vowel or diphthong. For /ay/, the maximal expansion occurs toward the beginning of /a/.
Perceptual centers in speech: An acoustic analysis
Authors: Scott Sophie , Howell Peter
Published: October 2005

Acoustic signals have a perceptual moment of occurrence, called the ‘‘perceptual center’’ or ‘‘P-center’’ [Morton etal., Psychol. Rev. 83, 405–408 (1976)]. No simple acoustic correlates of P-center location, such as vowel onset, physical onset, and peak amplitude, have been found. What has not been accounted for, however, is distortion of the acoustic signal by the experimental equipment. This can lead to discrepancies between measured attributes of the signal, and the signal which reaches the subject’s ear. The degree of amplitude and phase distortion has been calibrated, and a comparison made of the standard TDH-39 headphones, against ER-2 insert earphones, for amount of signal distortion. The effect of this distortion upon the acoustic attributes that have been previously dismissed as determiners of P-center location, is discussed. A model for P-center location based on the distribution of energy over time in the signal is outlined.
Processor-controlled multichannel adaptive hearing aid
Authors: Gilman Samuel
Published: October 2005

The hearing aid is a 13-channel system, each channel controlled by its own amplifier to maintain both the sound level and spectrum at the eardrum at levels preset by the audiologist. A built-in microphone measured the sound at the eardrum and a digital processor controlled each channel from this feedback signal. Objective tests using speech-shaped noise and simulated speech signals showed a ± 3-dB control for each channel over an input dynamic range of 30 dB. Subject tests were performed using standard prescription procedures to preset the target levels for each channel. Test materials were speech-shaped noise, nonsense syllables, and continuous speech. Results of the subject tests qualitatively show improvement due to the control. However, statistical analysis of the data show that the results are inconclusive. An examination of our test procedures and materials have been made to determine possible reasons for the inconclusive results.
Basic study of estimation of auditory source width (ASW) (1) ASW of two adjacent 1/3-oct band noises
Authors: Ueda K. , Kiyama M. , Morimoto M.
Published: October 2005

First, the paired comparison tests show that auditory source width (ASW) cannot be uniquely determined by the degree of interaural cross correlation and SPL overall frequencies of the stimulus but is affected by the frequency characteristics of the degree of interaural cross correlation and the power spectrum. Second, the experiments, in which ASW of 2/3-oct band noises consisted of two adjacent 1/3-oct band noises with different degree of interaural cross correlation are quantitatively evaluated by subjects, indicate that ASW of 2/3-oct band noise can be estimated by summing weighted ASW of each 1/3-oct band noise.
How to measure the degree of interaural cross correlation as a physical factor for auditory source width
Authors: Morimoto M. , Iida K.
Published: October 2005

The degree of interaural cross correlation is well known as a physical factor for auditory source width (ASW). But, how to measure it is not yet described clearly. The degree of interaural cross correlation obtained by using a dummy head with an artificial ear simulator cannot explain the results by Barron and Marshall [J. Sound Vib. 77, 221–232 (1981)]. But, the degree obtained by using a dummy head without an artificial ear simulator can explain the results. Furthermore, it is impossible to explain ASW of stimuli that include the high-frequency components (higher than about 1.5 kHz) even if the latter degree is adopted. Finally, this paper proposes auditory processed interaural cross correlation (APICC) (the degree of interaural cross correlation obtained in consideration of the mechanism of binaural signal processing) as a physical factor for ASW.
On the nature of nonmonotonic notches in input/output functions of 2f1–f2 acoustic distortion products
Authors: He Ning-ji , Schmiedt Richard A.
Published: October 2005

To better understand the nature of nonmonotonic notches in input/output (I/O) functions of 2f1–f2 acoustic distortion products (ADP), I/O functions were examined in 10 young, normal-hearing human subjects in conjunction with measurements of the ADP fine frequency structure [Gaskill and Brown, J. Acoust. Soc. Am. 88, 821–839 (1990)]. Primary frequencies were incremented in 1/32-oct. steps, the f2/f1 ratio was 1.2, and the primary levels were equal. ADP fine structure was measured over a 1/6 to 1/3 octave surrounding either 2000 or 4000 Hz with primary levels between 45 to 65 dB SPL in 2.5-dB steps. As primary level increased, a downward frequency shift of the peaks and valleys of the ADP fine structure was observed. This shift is responsible for the nonmonotonic notch and the wide variety of shapes of the I/O functions. As a result, even small changes in the frequency or level ratio of the primaries can drastically alter the shape of the I/O function. [Work supported by NIH-NIDCD P50-DC00422.]
Uni- and multidimensional identification of rise time, spectral slope, and bandwidth
Authors: Campbell Shari L. , Burns Edward M.
Published: October 2005

A series of experiments was used to evaluate absolute identification resolution for rise time, bandwidth, and spectral slope. Stimuli were complex tones comprising the first 40 harmonics of 200 Hz. Ranges for each physical dimension were restricted to those with validity to natural music and speech signals, and the number of categories ranged from three to five. A paradigm similar to one designed to assess channel capacity, perceptual interaction, and the relationship between uni- and multidimensional resolution [Durlach etal., Percept. Psychophys. 46, 293–296 (1989)] was employed. Performance was quantified in terms of information transfer. Unidimensional identifications were made for both fixed and roved background conditions in order to determine the extent to which the dimensions interacted perceptually. Resolution in two- and three-dimensional identification tasks were compared to resolution in unidimensional tasks in order to assess the degree to which the summed unidimensional information transfers from both fixed and roved background conditions predict multidimensional information transfer. [Work supported by the Virginia Merrill Bloedel Hearing Research Center.]
Underwater acoustic facilities at AARC
Authors: Brown Neal A. , Africk Steven A.
Published: October 2005

Facilities at AARC include an acoustic wind tunnel (AWT), a pressurized pulse tube, and a reverberant water tank. The AWT may also be used as an anechoic room. The AWT was designed primarily for the acoustic evaluation of submersible propulsors. It is large enough to use prototype hardware for an important class of such vehicles. A model afterbody with control surfaces and propulsor is submerged in an open jet surrounded by a large anechoic room. In addition to the low background noise levels and turbulence required for this duty, the AWT has a unique nose-sting model mounting arrangement that avoids the noise and flow disturbances of strut or wire supports. The pulse tube provides measurements of reflection and transmission properties of materials and coating systems samples at submergence pressures to 1000 psi and with controllable temperature. The acoustic water tank provides a highly reverberant facility useful for measurement of radiated power from submerged sources and has been used to measure radiation transfer gains by the application of reciprocity.
Coastal Systems Station acoustic test facility
Authors: Todoroff Douglas G. , Kargl Steven G.
Published: October 2005

The Coastal Systems Station acoustic test facility has two facilities (pontoon barge and measurement pier) for providing calibrations and tests of underwater acoustic devices. Both facilities reside over a freshwater pond with a maximum depth of 25 ft under the measurement platforms. The facilities are digitally instrumented and routinely provide measurements of projector and hydrophone beam patterns, projector source levels, hydrophone sensitivities, transducer impedances, as well as target strength measurements and one of a kind customer defined tests. The physical attributes, instrumentation, and measurement capabilities of the facilities are presented.
Test and evaluation of the Seawolf class submarine bow sonar arrays at the Naval Undersea Warfare Center, Seneca Lake Detachment
Authors: Guigli John
Published: October 2005

The U.S. Navy’s newest and largest submarine bow sonar arrays recently underwent acoustic and vibration measurements at the Naval Undersea Warfare Center (NUWC) test facility at Seneca Lake, Dresden, New York. The Seawolf class bow assembly, a 150-ton full-scale front end for the Seawolf class submarine (SSN 21), was built by Newport News Shipbuilding (NNS) and delivered to NUWC Seneca Lake via the New York State Canal System in May 1990. Upon the assembly’s arrival at Seneca Lake, NUWC and NNS personnel made use of shore-based support facilities to convert the assembly into a test fixture. The test and evaluation of the Seawolf bow fixture (SBF) were performed at the NUWC systems measurement platform, which is a floating measurements laboratory stationed approximately at the middle of Seneca Lake. The SBF sonar arrays were tested at depths of up to 200 ft. The measurements included passive and active array acoustic calibrations, horizontal and vertical beam patterns, and vibration tests. A description of the NUWC Seneca Lake facilities that supported the test preparations, test configurations, and data collection is presented.
CAMAC-based, pulsed-mode, integrated system for underwater acoustical measurements
Authors: Citrin Arthur
Published: October 2005

An efficient and highly versatile measurement system is described as presently utilized in a number of naval and commercial test facilities. Salient features of the CAMAC platform are covered with special emphasis on the applicability of this packaging scheme to the mixed analog and digital needs of acoustical instrumentation. Included is a general overview of the system block diagram and expandability to multiple channel measurements. Basic design considerations yielding operation over wide frequency range with superior phase fidelity and stability are presented. Internal signal processing capabilities both in time and frequency are summarized along with schemes for internal data storage and transfer to host computer. A range of internal sinusoidal and complex signal generators is described including high-speed arbitrary waveform generators and linear or complex frequency sweeps. Special noise reduction features for random noise and power line frequency-related disturbances are discussed as well as the hands-on controls for facilitation of initial acquisition window setup. Approaches for optimization of signal dynamic range are outlined together with the hardware features that assist in this goal. The presentation in general supplements other invited facility papers that utilize the described system but treat it as a black box.
Model experiments on air-coupled surface waves
Authors: Hutchinson-Howorth C. , Attenborough Keith
Published: October 2005

Experiments carried out by Donato [J. Acoust. Soc. Am. 63, 700–703 (1978)] demonstrated a continuous wave field behavior consistent with the formation of an air-coupled surface wave above an impedance boundary consisting of a hard-backed rectangular rigid lattice. He found it necessary to adjust the impedance of the lattice, predicted according to a thin-air-layer model, to fit the measured impedance and attributed the required adjustment to clamping of the sample in the impedance tube. A repetition of Donato’s continuous wave experiment is reported. A rectangular pore model for the impedance of the lattice is found to give better agreement with measured impedance data than a thin-layer model. Pulse experiments are reported that show clear evidence of the air-coupled surface wave as a separate arrival and the observed dependence on the measurement geometry is in good accord with predictions.
Indirect acoustic impedance computations involving fixed free-space attenuation or fixed boundary loss factors: Experimental model validation studies
Authors: Oni A. A. , Kalb J. T. , Garinther G.
Published: October 2005

An experimental verification of a surface acoustic impedance estimation theory and method is hereby presented. Two distinct estimation methods are evaluated. Each method utilizes a search algorithm to estimate the impedance that will validate analytical expressions for ground attenuation of sound, using experimental measurements of continuous-wave sound attenuation changes between two above-ground sound source locations with respect to a single microphone close to the ground. With each of the two methods evaluated, reliable ground impedance estimates were computed with considerable reduction in computation and experimental complexity: The first method requires an experimental arrangement that allowed spherical spreading components of the excess attenuation terms to essentially cancel out. With the second method, the arrangement allowed the boundary loss factor to remain constant while the source sound location changes. Variations in meteorological parameters between the two sound source locations were assumed negligible in each case. These successfully verified analytical and experimental arrangements display the potential to provide a means of obtaining speedy and efficient estimates of acoustic impedance for various surfaces. [Work was supported by the US Army Research Office and the Human Engineering Laboratory, Aberdeen Proving Ground, MD.]
A neural networks approach to multiple passive sonar targets identification by source separation technique
Authors: Yang Hong-Tzer , Lee Tai , Tu Chu-Kuei , Lin Jyh-Maw
Published: October 2005

A source separation and neural network unsupervised learning procedure has been proposed and applied to the identification of multiple passive sonar targets. Acoustic noises radiated from 101 fishing boats were collected from two widely separated underwater hydrophones. A noise background whitening algorithm was applied to flatten the power density spectra (PDS) [W. A. Struzinski and E. D. Lowe, J. Acoust. Soc. Am. 76, 1738–1742 (1984)]. The system was trained by using the single-target spectrum shapes derived from one of the hydrophones and then used these to identify the sources from the other hydrophone for both single and multiple targets. Multitarget signals were preprocessed by a source separation technique to obtain the individual signals. Results of practical testing indicated that the system could correctly identify 90.1% of the recordings for a single sonar target. Identification rate of the multi-target signals can achieve 84% for 50 different combinations of single-target signals. This paper describes the system configuration, the experiment design, and experiences with the practical applications. [Work supported by CSIST.]
Talking to bats: what do arctiids say?
Authors: Fullard James H.
Published: October 2005

Certain tiger moths (Arctiidae) generate trains of high-frequency clicks from specialized structures (tymbals) on their thorax. Although social functions for the sounds have been demonstrated, debate continues on their anti-bat defensive role with aposematism, startle, and/or echolocation disruption being offered as to how the sounds operate. The controversy arises primarily from the near impossibility of observing natural, in-flight behavior of moths and hunting bats. As a result, most of the experiments attempting to explain the sounds have used indirect methods to infer their actions. These studies include examinations of the structure of the sounds, stimulation of stationary moths with artificial bats, and stimulation of trained, laboratory bats with artificial moths. Other laboratories have attempted to exploit the foraging behavior of wild bats in controlled conditions. These studies have not yet satisfactorily explained why arctiids talk but one observation that has become clear is that the defenses of tiger moths, as with most animals, are directed against a variety of predators and that single explanations do not exist for complex behaviors.
A novel hearing organ in an acoustic parasitoid fly
Authors: Robert D. , Miles R. N. , Hoy R. R.
Published: October 2005

Female parasitoid flies (Ormiaochracea: Tachinidae) are attracted to singing male crickets [W. Cade, Science 190, 1312 (1975)]. Gravid female flies orient to calling crickets to deposit larvae, which burrow into the cricket in order to feed and grow to adulthood. To detect and locate cricket songs over long distances, these flies evolved a tympanal hearing organ previously unkown in flies. Electrophysiological recordings reveal low auditory thresholds (20 dB SPL between 4–6 kHz) matching the peak of the power spectrum of the host’s song (Gryllusrubens, 4.5–5 kHz). Such sensitivity is consistent with long distance hearing. The ear, composed of two adjacent thin tympanal membranes, contains two mechanoreceptive sensory organs that are both located in a single air-filled chamber. Laser vibrometry suggests that this tympanal arrangement provides a novel mechanism for directional sound reception. Some aspects of its vibratory behavior in a sound field will be reported. [Work funded by NINDCD-DC00103 and Hatch NYC-191403.]
The acoustics of Hawaiian katydids
Authors: Strazanac John S. , Au Whitlow W. L.
Published: October 2005

Nine species of katydids endemic to Hawaii were collected from seven of the main islands in the Hawaiian archipelago, and the acoustic signature of their calls were analyzed. Each species was enclosed in an anechoic sound booth and the calls generated by wing stroking were detected with a broadband miniature microphone and recorded with a broadband instrumentation tape recorder. Signature calls were digitized at an effective sample rate of 500 kHz and stored on PC floppy diskettes. Acoustic analysis of the digitized signals included a determination of the time waveform, frequency spectrogram, and time-amplitude envelope for each wing stroke. Duration of wing strokes varied from approximately 10 to 30 ms, with peak frequency of each click within a wing stroke varying between 18 to 35 kHz. Rate of wing strokes varied between 9 and 70 per second. Most of the species can be separated according to the acoustic features of the calls. Song variations among isolated populations from one of the islands gave indications of a new species that was later verified by cuticular hydrocarbon composition and morphological analyses.
The spatial amplitude mapping method for estimation of time delay using adaptive filtering
Authors: Nehal Mohammad K. , Henriquez Juan A. , Riemer Terry E. , Trahan Russell E.
Published: October 2005

Uniform and multiple delays/advances were estimated under heavy noisy conditions [signal-to-noise-ratio (SNR=signal energy/noise energy) below 0 dB]. The technique introduced, called the spatialamplitudemapping (SAM) method, isolates a data segment from each of the channels in a multichannel system by using a suitable window. When two matched segments are plotted on an x–y plane, the distribution of every pair of windowed coordinates will remain near a straight 45° line that passes through the origin. The distribution of two nonmatched segments or two matched segments containing noise will scatter around this line; hence, a pair of matching windowed segments can be found by searching for the distribution closest to the 45° line. This information is then used to estimate the segment delays. Under very noisy conditions, however, multiple delays can be detected. A recursive least-squares (RLS) filter is then used to adaptively estimate the correct delay. The technique was implemented in the delay estimation of synthetic (stationary) data and neurophysiological (nonstationary) data with satisfying results. Compared to the window correlation technique (WCT) [Callison etal., J. Acoust. Soc. Am. 81, 1000–1006 (1987)], SAM can estimate delays down to a SNR of −17.1 dB while the lower bound for the WCT is −2 dB.
Vowel-consonant-vowel transitions, analysis and modeling
Authors: Carré René , Chennouk Samir , Mrayati Mohamed
Published: October 2005

Male vowel–consonant–vowel formant transitions are analyzed using cepstral techniques. The first three formants are detected and represented as trajectories in the F1–F2 and F1–F3 planes. Distinctive formant trajectories are clearly revealed, exhibiting specific trajectory angles at the onset during the initial vowel, during the consonant production, and at the offset during the target vowel. Formant stability are also studied. These trajectories are interpreted using a distinctive region model of the vocal tract [Mrayati etal., Speech Commun. 7, 257–286 (1988)]. Such a model is controlled by means of a very simple command strategy: transversal constrictions at specific places. Trajectories predicted by the model correspond well with those observed on natural speech. Explanations on the articulatory timing control can be proposed. Simple synthesis-by-rule using a distinctive region model which inherently incorporates coarticulation effects is deduced.
A model of concurrent vowel segregation using an implementation of the subtraction strategy
Authors: Lea Andrew P.
Published: October 2005

The perceptual experiments performed by Lea [‘‘Auditory Modeling of Vowel Perception,’’ unpublished doctoral thesis, University of Nottingham (1992)] required listeners to identify the members of pairs of steady-state synthetic vowels presented monaurally, called concurrent vowels. Accuracy of identification was higher when one vowel was voiced and the other whispered compared to control conditions in which both were whispered or both were voiced with the same fundamental frequency (f0). Surprisingly, the improvement in accuracy was restricted to the whispered member of the voiced/whispered pair. This outcome is compatible with the idea that one strategy used by listeners to recover a target voice from a mixture of voices is to cancel an interfering voice by a process of spectrotemporal subtraction. Here, a computational model is described that implements this strategy. The model is based on a bank of bandpass filters and an array of autocorrelators. The period of the dominant pitch in a stimulus is estimated from the largest peak in the normalized sum of the autocorrelation functions. This estimate guides the synthesis of an array of autocorrelation functions that are subtracted from the array generated by the stimulus, hence, canceling an interfering voice. Following Meddis and Hewitt [J. Acoust. Soc. Am. 91, 233–245 (1992)], vowel identification is performed by matching the short-time part of the resulting summed autocorrelation functions to a set of stored templates. The model makes good qualitative predictions of the accuracy of listeners’ identification responses to the stimuli used in the perceptual experiments. In addition, it makes good quantitative predictions of the effects of introducing a difference in f0 between concurrent vowels reported previously by Assmann and Summerfield [J. Acoust. Soc. Am. 88, 680–697 (1990)]. [Work supported by the MRC of the UK.]
Spectral integration in vowel perception: Matching and discrimination studies
Authors: Johnson Keith , Fernandez Marisa , Henninger Michael , Sandstrum Jim
Published: October 2005

Chistovich [J. Acoust. Soc. Am. 77, 789–805 (1985)] reviewed evidence suggesting that two vowel formants within 3.5 Bark of each other are integrated into a single spectral center of gravity in vowel perception. This center of gravity hypothesis, which has been an important concept in theories of vowel perception, was tested in a series of experiments utilizing two-formant stimuli (a continuum from [a] to [æ]) and one-formant stimuli. Listener’s one-formant matches to the two-formant standards had formant frequencies between the frequencies of the two-formant standards (as predicted by the center of gravity hypothesis), however there were no abrupt changes in the matching function over a range of two-formant separations from 0.5 to 6 Bark. The best-matching one-formant frequencies were generally higher when the bandwidth of the single formant was narrower, but the results were not affected by manipulations of overall amplitude. A discrimination experiment showed that listeners could discriminate between a two-formant stimulus and its best-matching one-formant counterpart. A second discrimination experiment showed that when the stimulus durations were very small (50 ms) and the interstimulus interval was large (3 s) discrimination between one- and two-formant stimuli was poorer when the two formants were close to each other in frequency. However, contrary to the center of gravity hypothesis, there was no sudden change of performance around the hypothesized 3.5 Bark boundary. [Work supported by UCLA, Linguistics Dept.]
Perception of vowels synthesized from sinusoids that preserve either formant frequencies or global spectral shape
Authors: Zahorian Stephen A. , Zhang Zhong-Jiang
Published: October 2005

Vowel tokens were synthesized from sinusoids using two methods. For the first method (formant sinusoids), three variable frequency sinusoids were used, with frequencies adjusted to match the first three formants extracted from naturally spoken vowels. The amplitudes were filtered at −12 dB/oct, to approximate the roll-off of the glottal source. For the second method (spectral shape sinusoids), 16 fixed-frequency sinusoids, approximately equally spaced on a bark scale, were used. For this method, the amplitudes of the sinusoids were adjusted so that the overall spectral shape of the naturally spoken token was preserved. A forced-choice identification experiment was conducted using the five vowels /a,i,u,æ,schwa/. As a control, tokens were also generated using one period of the original speech, periodically extended to match the length (1 s) of the synthesized tokens. The average percentages of tokens correctly identified were 87%, 37%, and 67% for the original, formant sinusoids, and spectral shape sinusoids, respectively. These results clearly show that vowel stimuli which preserve formant frequencies, but which distort spectral shape, are perceptually impoverished. In contrast, vowel stimuli which preserve spectral shape, but which only approximately preserve formants, are identified with greater accuracy. However, neither set of stimuli are identified as accurately as are the original tokens.
A multidimensional mathematical model of vowel perception by users of pulsatile cochlear implants
Authors: Svirsky Mario A. , Svirsky Santiago H.
Published: October 2005

A mathematical model is proposed to account for vowel perception by users of pulsatile cochlear implants that encode one to three formant frequencies. Like the Durlach–Braida model of intensity resolution [N. I. Durlach and L. D. Braida, J. Acoust. Soc. Am. 46, 372–383 (1969)], this model incorporates an internal noise model and a decision model to account separately for errors due to poor sensitivity and response bias. The model was evaluated using published electrode identification data [Y. C. Tong and G. M. Clark, J. Acoust. Soc. Am. 77, 1881–1808 (1985)] and vowel confusion matrices obtained with stimulation schemes that employed one, two, and three formants, respectively [P. J. Blamey and G. M. Clarke, J. Acoust. Soc. Am. 88, 667–673 (1988)]. Vowel identification performance seems to be strongly influenced by the extent to which a given subject can adjust to electrical stimulation delivered to places in the cochlea that are more basal than the places that are stimulated by the same formant in a normal ear, and also by the subject’s basic electrode identification capability. Results suggest that cochlear implant users are able to partially adjust to this too-basal stimulation. [Work supported by NIH Grant Nos. DC01721 and DC00075.]
Effect of bearing errors and random sound-speed fluctuations on a holographic source localization method
Authors: Rouseff Daniel , Porter Robert P.
Published: October 2005

A holographic source localization method has recently been proposed by Mourad etal. [J. Acoust. Soc. Am. 92, 1031–1039 (1992)]. The signals from both a reference and an unknown source are measured and combined along a vertical receiving array. To the extent that the two signals travel through the same medium, fluctuations in the sound-speed profile along the common paths approximately cancel thereby reducing the knowledge of the medium required to locate the source. In the present work, the effects on the localization algorithm of bearing errors and random fluctuations such as those caused by internal waves are considered. A 21/2-dimensional formulation featuring adiabatic modes and weak scattering is used in the forward scattering problem. The statistical properties of the predicted source location are examined. The random fluctuations are shown to diminish the amplitude but not the location of the mean focus and also to modify the sidelobe structure. Numerical examples are given. [Work supported by ONR.]
An active noise control stethoscope
Authors: Harley T. R.
Published: October 2005

A prototype of an active noise control stethoscope has been developed. The device enables medical personnel to clearly hear sound from a stethoscope device while working in a noisy environment. Using a conventional stethoscope to detect lung noises and blood pressure is difficult in many emergency medical environments, such as helicopters. An electronic stethoscope device is to be designed that will make use of digital signal processing technology to remove unwanted noise. Acoustical noise and vibrations that are detected by the stethoscope are digitally subtracted out of the stethoscope signal. The stethoscope signal is then amplified and broadcast into the earcup of an an antinoise headset. Unwanted noise penetrating the shell of the earcup is eliminated by broadcasting counteracting ‘‘antinoise’’ in addition to the amplified stethoscope signal.
Detection of real propagated transient signals by linear transform based detectors
Authors: Maguer Alain
Published: October 2005

The detection of transient signals by linear transform based detectors is investigated. The wavelet transform and the short-time Fourier transform are mainly considered. The wavelet transform is used here in its suboptimal time-frequency interpretation and is realized by a bank of nonorthogonal ‘‘constant-Q’’ filters. First, the performance of the wavelet detector is compared (by Monte Carlo simulations operating on synthetic data) to the performance of the short-time Fourier transform detector (STFT), the Wolcin detector, the matched filter detector, and finally the energy detector. The performance of the wavelet (‘‘constant-Q’’) detector and the STFT detector is shown to be almost comparable (however, the wavelet detector performs better with signals of wider bandwidths) but they are worse than the performance of the matched filter detector. The Wolcin detector, in this application, does not give the expected results and has poor performance, even for narrow-band signals. Then, the comparison of the performance of all the detectors is performed for real propagated transients (the same as those used in Monte Carlo simulations) transmitted by a sound source and received on a towed array in a deep water environment. The same conclusions as on synthetic data have been derived.
A twin study of spontaneous otoacoustic emissions
Authors: Russell Ann F. , Bilger Robert C.
Published: October 2005

Because spontaneous otoacoustic emissions (SOEs) are correlated with sex and because sex is determined genetically, Bilger etal. (1990) hypothesized that SOEs are inherited. A twin study was planned to pursue this idea. A total of 60 sets of twins [15 sets of monozygotic (MZ) females, 15 sets of MZ males, 15 sets of dizygotic (DZ) females, and 15 sets of DZ males] were measured for SOEs using an off-line procedure to determine the broad-sense heritability (H) of these emissions. Their ages ranged from 5–40 years. All subjects had essentially normal hearing. Zygosity was determined by means of a zygosity questionnaire (Lykken etal., 1988). Results indicate that SOEs are a highly heritable trait (H=1.0). Further, the SOE profiles of MZ twins were strikingly similar, whereas the profiles for DZ twins were not. [Work supported by DC00174.]
Optimization of actuator locations for active control of sound radiation from structures under multiple-frequency excitation
Authors: Song Tao , Fuller Chris R. , Burdisso Ricardo A.
Published: October 2005

The problem of active structural acoustic control (ASAC) of radiation from simply supported rectangular plates under multiple-frequency excitation is studied in this paper. An optimization procedure is presented for optimal location of piezoelectric actuator(s) to actively control the multiple-frequency sound radiation. Expressions for the multiple-frequency radiated sound power by the plate are derived and chosen as the objective function in the optimization approach. Control performance for optimally located actuator(s) are compared with that of actuator locations based on some physical understanding of the system response. In general, it is demonstrated that it is possible to optimally locate actuator(s) in order to effectively control the sound radiation at the frequencies considered. In addition, it is shown that the optimization approach provides a method to design a robust controller with respect to the frequency uncertainty for single-tone excitation cases. [Work supported by ONR.]
A nonreflecting boundary condition for use in PE calculations of sound propagation in air
Authors: Mayfield M. E. , Thomson D. J.
Published: October 2005

Although the parabolic equation (PE) method has long been used to model underwater sound propagation, its applicability to outdoor sound propagation has been realized only recently [e.g., K. E. Gilbert and M. J. White, J. Acoust. Soc. Am. 85, 630–637 (1989)]. In ocean acoustics, an absorbing layer is usually appended to the base of the computational grid to suppress unwanted reflections that arise from a Dirichlet boundary condition imposed at a finite distance below the ocean-bottom interface. For sound propagation in air, however, this local boundary condition must be applied at the top (z=H) of an absorbing layer that occupies the region h≤z≤H above the air–Earth interface. Herein, this absorbing layer is replaced with an exact boundary condition applied along the plane z=h itself. Although this boundary condition is nonlocal and is thus more computationally demanding than a local one, the extra computing effort is offset by requiring fewer grid points in air. To illustrate the numerical effectiveness of the new boundary condition, PE calculations are compared to reference solutions for several examples of propagation in air.
An inexpensive air-compensated parabolic dish for underwater acoustics
Authors: Santos Gerald M. , Menoche Richard K. , Stein Peter J.
Published: October 2005

The benefits of using highly directive parabolic reflectors for acoustics and electromagnetics work is well known. The primary difficulty for underwater acoustics work is making a highly reflective surface, one that is acoustically very hard or very soft, in the shape of a parabola. Very hard would require a great deal of machined metal and would most likely be cost and weight prohibitive. Therefore, for underwater acoustics work, parabolas are usually made to be acoustically soft. Although soft foam is a good choice, it will crush at even shallow depths. Other solutions include compliant tube designs with and without pressurized air compensation. These tend to be very expensive. Here, an inexpensive and rugged parabolic receiver/transmitter with virtually no depth limitation is described. It consists of a fiberglass parabola backed by a fiberglass dome. The air cavity between them results in a soft reflector. Vibration isolation is provided by steel rings and rubber gaskets. To prevent crushing, the air chamber can be pressurized either by compressed air fed by hose from the surface or an attached tank and regulator. Both transmit and receive directivity patterns will be shown. The device is currently being used for target strength measurements at the NUWC Lake Seneca Test Facility.
Time-delayed nonlinear dynamical systems as musical instruments
Authors: Rodet Xavier , Doval Boris
Published: October 2005

In many physicalmodels of musical instruments, the instrument itself is represented by a passive linear system coupled to a nonlinear dynamical system (NDS) [M. E. McIntyre etal., ‘‘On the Oscillations of Musical Instruments,’’ J. Acoust. Soc. Am. 74, 1325–1345 (1983)]. Modeling the excitation process is one of the key points for music synthesis [Proc. Colloquium on Phys. Model., ACROE, Grenoble, France (Oct. 1990)]. It would allow the design of synthetic instruments with as much flexibility and expressivity as natural ones and hopefully with high sound quality. Some classical NDS and the various forms they have in musical instruments such as voice, trumpet, clarinet, or strings are presented. A clarinet-like basic model is designed taking as a starting point the sound requirements. Conditions of periodic oscillation are studied. Properties of the sound are related to the characteristic of the nonlinearity. Other typical behaviors such as period doubling or chaotic regime result in interesting sound effects. A real-time implementation allows for easy experiments and tests of the parameter values. Sound results are demonstrated and discussed. a)Presently on sabbatical at Ctr. for New Music and Audio Technol. (CNMAT), Univ. of California at Berkeley, 1750 Arch St., Berkeley, CA 94709.
Fractal waveforms and cellular dynamata
Authors: Radunskaya Ami
Published: October 2005

Chaotic dynamical systems have intrigued artists since the computer produced images of their complex trajectories. For the composer, the structures inherent in these systems and their time evolution provide a framework for musical expression that reflects a contemporary interpretation of reality; for the sound-designer, the same structures can be used to synthesize rich, organic timbres. These two applications are explored in this paper. The map Rα(x):x→x+α (mod 1) is iterated to produce a fractal waveform for each irrational number α. These quasiperiodic waveforms have complex overtones, and exhibit a quasi-self-similarity reflecting the number-theoretic properties of α. This audification of a system is often a biproduct of the musician’s explorations of mathematical structures, and on the compositional level methods of hearing and performing a dynamical system become a primary focus. Performance gestures can be mapped to the space of initial conditions and the parameter space of the system. The time series from the resulting system, along with averaged data (Lyapunov exponents, for example), are mapped to musical parameter space. Several such mappings are illustrated using a toral lattice of two-dimensional twisted logistic maps.
Status of numerical methods and the experimental database
Authors: Daigle Gilles A. , Bass Henry E. , Raspet Richard
Published: October 2005

Propagation of sound in the atmosphere is governed by a number of interacting physical mechanisms including geometrical spreading, molecular absorption, reflection from a complex impedance boundary, refraction, diffraction, and scattering. Accurate predictions of sound transmission loss from a source to a receiver must somehow account for all of these phenomena simultaneously. Although this goal is beyond current capabilities, the application of numerical techniques to solve the full wave equation has led to significant advances. Current implementations agree well with experiments for low turbulence and flat ground. At moderate ranges, even the effects of turbulence can be accounted for. Members of NATO Panel 3 Research Study Group 11 began an effort in 1990 to compare predictions of the various models and different implementations of the same model with the goal of identifying any differences and sources of those differences and to provide benchmark cases which future researchers can use to check new models. Further, the amount of well-documented experimental data to which the numerical solutions can be compared is very limited. During the summer of 1991, a Joint Acoustic Propagation Experiment (JAPE) was organized in order to extend the database. This paper will summarize the benchmark effort, review the available experimental data and comparison to theory prior to the JAPE, and finally, summarize the JAPE.
Phase and amplitude fluctuations in a refractive shadow
Authors: Havelock David I. , Stinson Michael R. , Daigle Gilles A.
Published: October 2005

For a point source in an upwardly refracting atmosphere, an acoustic shadow region exists beyond the limiting sound ray that grazes the ground. A receiver located within the shadow will have no direct ray path for sound propagation and sound energy arrives at the receiver through other mechanisms. Experiments using an array of microphones at a range of about 1 km and source tones of about 1 kHz indicate that the character of the phase and amplitude fluctuations of the received signal within an acoustic shadow region can vary dramatically. Relationships between the characteristics of the fluctuations and physical propagation mechanisms are explored. In particular, the possibility is investigated that the mechanisms of scattering, diffraction, and ‘‘line-of-sight’’ propagation, each intermittently, dominate the received sound field. Application to adaptive beamforming in the atmosphere is discussed.