bluuebunny/crossref_metadata_2025 huggingface dataset:
A tool for the study of pitch perception
Authors: Teas D. C.
Published: October 2005

The equations describing the stimulus configuration for generating the Shepard pitch illusion [R. N. Shepard, J. Acoust. Soc. Am. 36, 2346 (1964)] have been implemented with the computer-assisted programming language, LabVIEW, using a MacIntosh personal computer equipped with digital signal processing capability and digital–analog conversion. Shepard’s demonstration used 10 octaves (c=10) and 12 increments (t=12). The intensity distribution of the components ranged from 22 to 56 dB, with the strongest components in the mid-range of frequencies. Qualitative observations of the effects of varying c and t suggest that Shepard’s values produce the richest perceptions of pitch. However, even for these original parameters, it is found that the essence of the illusion, i.e., the continuous ascent or descent of the pitch, is lost as the duration of exposure (not specified in the equations) for each t is decreased. The LabVIEW implementation allows the user to vary the number of octaves (c), the number of steps (t), the range of amplitudes for c, or the frequency range over which c is taken. In general, as the values for these parameters vary from optimum, the continuity of the pitch change is lost and the listener hears the cycles (through the t steps) repeat. A modification of the program permits sinusoids to be replaced by narrow bands of noise. At narrow bandwidths, e.g., 0.7(fti) and 1.3(fti), one can hear the pitch in a background of noise, and the illusion is preserved if the parameters are appropriate. As the bandwidth increases, the strength of the pitch perception decreases. [This research was supported by contract F33615-89-c-0603 to Krug Life Sciences.]
A procedure for the study of auditory stream segregation
Authors: Mendoza Lee , Collins M. Jane , Cullen John K.
Published: October 2005

An approach based on signal detection theory was used to measure the stream segregation of tone sequences. Previous research [Dannenbring and Bregman, J. Exp. Psych. Human Percept. Perform. 2, 544–555 (1976)] has shown that stream segregation is highly correlated with the elements in a sequence being perceived as ‘‘overlapped.’’ In the present experiment, two classes of sequences were employed. One half of the sequences was composed of tones that physically overlapped, and the remainder of the sequences in the experiment used tones which were physically separated. The sequences were parametrically varied in terms of duration of the sequences and frequency differences of the tones in the sequences. The experimental task required the subject to decide whether or not a sequence was overlapped. Responses to the two classes of sequences were used to calculate measures of discriminability, bias, and criterion. Results using this procedure primarily show significant main effects of frequency difference of the elements in sequences and duration of the sequences; however, no significant interaction of frequency difference and duration was found.
Intensity resolution performance and loudness growth in high-pass noise
Authors: Schlauch Robert S.
Published: October 2005

Several models predict tight coupling between loudness growth and intensity resolution performance. To examine further the link between these measures, intensity resolution performance and loudness growth for a 1000-Hz tone were measured in the presence of a unilateral, continuous, high-pass noise (cutoff: 1800 Hz). Thresholds for the three subjects were minimally effected at the test frequency; however, for frequencies in the noise passband, thresholds were shifted to roughly 60 dB SPL. Intensity resolution was measured for gated and continuous standards using a 3-AFC procedure. Loudness matches were obtained using the contralateral ear as a reference. Below roughly 40 dB SPL, loudness growth and intensity resolution were unaltered by the presence of the noise; above 40 dB SPL, intensity resolution performance was well described by Weber’s law (gated standard) or by a slight deterioration in performance (continuous standard) while loudness growth became slower. The results will be described in terms of current models for relating these measures.
Effects of standing waves on high-frequency thresholds
Authors: Zhou Bin , Green David M.
Published: October 2005

The effects of standing waves in the ear canal on high-frequency pure-tone thresholds were investigated. Sound was transmitted from a transducer to a high-impedance source by a long plastic tube 0.64 cm in diameter. The high-impedance source was coupled to the ear canal by a shorter plastic tube 7 cm in length. Thus a standing wave field was created between the high-impedance source and the eardrum. Using a broadband stimulus, standing wave patterns were recorded by a small microphone placed near the high-impedance source. Several maxima and minima of the standing wave pattern were located in the frequency range from 6000–12 500 Hz. Thresholds were measured at two successive minima, the maximum between them and two intermediate values. Thresholds were also obtained with the high-impedance source removed from the system, resulting in a source impedance approximately equal to the characteristic impedance of air. Thresholds with and without the high-impedance source were compared and standing wave ratios (the difference between thresholds at the standing wave maximum and minimum) of 5–10 dB were observed. [Research supported by NIH.]
Virtual instrument implementation of a one-third-octave-band analyzer
Authors: Hanks John , Koenig David , Yang Qiuhong
Published: October 2005

Virtual instrumentation, using software application tools and off the shelf plug-in cards to turn your computer into an instrument, is a very economical, attractive, and flexible method to perform audio analysis. Plug-in digital signal processing (DSP) boards provide personal computers with a level of processing power that was once restricted to all but the most powerful mainframe computers. These boards transform the PC into a parallel processing environment, delivering millions of floating point operations per second (MFLOPS) or more to numerically intensive tasks. Couple this with high-quality A/D converters and virtual instrumentation software tools and you have a powerful environment for building applications ranging from digital audio to underwater acoustics to vibration and modal analysis. This paper details a real-time acquisition and analysis DSP application using a 16-bit delta-sigma A/D converter and an AT&T WE DSP32C-based plug-in board to acquire and process restricted band audio frequency (dc to 5 kHz) information. A real-time restricted band one-third-octave analyzer was implemented using a AT computer and a plug-in board equipped with a digital signal processor and an audio frequency A/D converter. The restricted band one-third-octave analyzer provides the AT computer with enough power to meet or replace several stand-alone one-third-octave analyzers, and virtual instrumentation tools allow ease of configuration for new applications.
Issues in the development of virtual haptic environments
Authors: Srinivasan Mandayam A.
Published: October 2005

The term haptics refers to manual interactions with the environment. The development of virtual haptic environments requires a good match between the haptic capabilities of the human user and the performance of the interface devices. A haptic interface can be viewed as having two basic functions: (1) to measure the positions and forces (and time derivatives) of the user’s hand and other body parts, and (2) to display forces and positions (and/or their spatial and temporal distributions) to the user. The biomechanical, sensorimotor, and cognitive abilities of the human set the specifications for the haptic interfaces. How well these specifications can be achieved by the interfaces depends on the mechanical design, sensorimotor control, and computational capabilities of the devices. In this talk, an overview of the issues in the development of haptic environments will be presented from the perspectives of human studies, interface device development, and computations.
Neuronal processing in cochlear nucleus of vowel combinations: Effect of intensity and intervowel time
Authors: Mandava Pitchaiah , Rupert Allen L. , Moushegian George
Published: October 2005

This study presents evidence on how three cochlear nucleus (CN) neuronal types, primary, chopper, and pauser/buildup, process complex sounds. The stimuli were vowels, /i/, /a/, and /u/, from three corners of the F1–F2 vowel space. Having first gathered neuronal data to vowels presented separately, pairs of these vowels were then delivered with intervowel time intervals from 0 to 100 ms in 20-ms steps. The intensities of both vowels were systematically varied. Vowel sound combinations almost invariably provoked changes in neuronal responsiveness, e.g., discharge rates and interval distributions. In some instances, a given vowel sound altered the discharge pattern to a succeeding one, even after time intervals extending to 60–100 ms. In others, the effect was maximal when the vowels had no time separation, and did not persist beyond 20 ms. Finally, various combinations in vowel intensities and intervowel times modified the temporal response characteristics of each neuronal type; but in particular the choppers, for which alterations were considerable.
Full-spectrum industrial-base hydroacoustic support: A survey of the NSWC Crane Division underwater test complex
Authors: Moody George T. , Koeper Wayne R.
Published: October 2005

The Crane Division of the Naval Surface Warfare Center operates a full-spectrum underwater acoustic test complex for engineering and production testing of air launched and hull mounted sonar transducers and related products. Primary emphasis is on accuracy and efficiency, gained predominately through extensive automation, standardization, and documentation. Facilities include various pressure/temperature-controlled tanks and lake and ocean sites. The presentation describes the attributes, limitations, data acquisition systems, and typical applications of the individual test facilities.
Current and projected evolutions in the Navy’s sonobuoy acceptance test facilities
Authors: Scherschel Michael R.
Published: October 2005

For 25 years the Navy has conducted sonobuoy functional acceptance tests at a dedicated contractor operated facility in St. Croix, U.S. Virgin Islands. This program is supported by P-3 aircraft and ocean test vessels also operated by the contractor. Other critical sonobuoy design parameters are evaluated at a tow test facility in the waters of southeast Alaska. To accommodate program downsizing, it is necessary to utilize Navy owned and operated assets to fulfill the acceptance test function (considerable savings result from this approach). As a result the acceptance test function will transfer from St. Croix to San Clemente Island so that existing Navy assets are more fully utilized. This function transfer must be properly monitored to assure no reduction in reliability of sonobuoys entering Fleet stockpiles. Currently the Navy is exploring a possible alternative to ocean airdrop and tow tests by developing a lake facility near NSWC, Crane Division in Indiana.
Design of a low-cost ambient pressure acoustic test facility
Authors: Broding Mark S. , Knight J. Scott
Published: October 2005

The setup and operation of a low-cost acoustic test facility currently in use by Syntech Materials, Inc. is described. Transmission and reflection characteristics of bulk materials (in the form of test panels) are measured at ambient pressure and temperature at normal incidence from 10 through 100 kHz. Data collection is accomplished through standard low-cost devices controlled by a personal computer via GPIB link. Results of sample acoustic tests are presented and discussed. Few problems have been encountered during the system’s first year of operation.
Maximum-likelihood harmonic matching for fundamental frequency estimation
Authors: Rodet Xavier , Doval Boris
Published: October 2005

This presentation deals with the estimation of fundamental frequency (f0) of pseudoperiodic sound signals with important results for polyphonic frequency tracking, and voice separation. Given a set of candidate partials in the signal, the estimation of f0 is taken in the sense of finding the optimal period duration(s) according to a criterion of maximum-likelihood harmonic matching. Excellent results have been obtained on large databases of speech (40 mn) and music [B. Doval and X. Rodet, Proc. IEEE-ICASSP, Toronto, May (1991)]. The algorithm has been implemented at IRCAM to run in real time for live performance frequency tracking. Developments are in several directions. A combined estimation of f0 and of a spectral envelope improves both estimations. Most important is the estimation of the ‘‘apriori’’ distributions of the different random variables on a learning set. Finally, a hidden Markov model tracks f0 trajectories between adjacent frames. The first experiments of polyphonic frequency tracking and voice separation are very promising. The model can be transposed directly to the maximum-likelihood estimation of several harmonic sounds since it already considers more than one f0 value. a)Presently on sabbatical at Ctr. for New Music and Audio Technol. (CNMAT), Univ. of California at Berkeley, 1750 Arch St., Berkeley, CA 94709.
Frequency tracking of solo and duet passages using a harmonic two-way mismatch procedure
Authors: Maher Robert C. , Beauchamp James W.
Published: October 2005

Estimation of the fundamental frequency (F0) of quasiharmonic signals is an important task in musical acoustics. Many existing techniques have unsatisfactory performance due to octave errors, noise perturbations, and other artifacts encountered in practice. Our work in this area, begun in 1986, uses the quasiharmonic assumption to guide a search for an estimated F0, based on the McAulay–Quatieri short-time spectrum measurement procedure. Estimation of F0 involves a two-way mismatch calculation where the discrepancy between the measured partial frequencies and a harmonic search grid is used to refine the F0 estimate. Inaccuracies present in the measured partial frequencies are averaged over all the available partials, thereby improving the F0 prediction. A heuristic weighting scheme is used to help reduce the susceptibility of the procedure to noise or partials missing in the spectral data. The frequency search strategy has also been extended to fundamental frequency estimation of duet (two voice) polyphony. Results for F0 tracking of solo and duet recorded musical instrument passages will be demonstrated.
Live-performance frequency and feature extraction for synthesis control
Authors: McMillen Keith , Wessel David
Published: October 2005

Musical frequency and feature extraction techniques for guitars and other string instruments are described. These methods obtain rapid and accurate estimates of frequency, intensity, and related spectral features to be used for synthesis control in live musical performance. Live-performance control imposes severe constraints in that the string interface and the synthesis system to which it is attached must respond with the control intimacy characteristic of an acoustic instrument. This requires extremely rapid frequency determination, which is accomplished by using fingerboard information in combination with the acoustic signal from the string. For the guitar, a fret scanning system has been developed to provide an estimate prior to the sounding of the note. Specialized sensor technologies are used to isolate the individual strings. Waveform features are timed to obtain period estimates used to continuously update the initial information from the fret scan. This combination of fret-scan information and period estimation from the signal provides for very accurate tracking and synthesizer notes can be started within 2 ms of the string excitation. It will be demonstrated that the MIDI protocol imposes severe constraints on control intimacy. A more suitable communications protocol for synthesis control and a new generation of instruments that provide an intimate link between instrument interface and synthesis are presented.
New formulas for spherical wave reflection from an impedance surface
Authors: Li Y. L. , White Michael J. , Hwang M. H.
Published: October 2005

The problem of sound reflection of a spherical wave from a locally reacting surface is studied by a new method, which is based on the solution of the heat equation. The new method maps the wave equation formulation with its associated boundary condition to a heat equation problem. In the heat equation domain, a Green’s function solution is relatively easy to obtain via the inverse Laplace transform. Finally, a compact approximation of the Green’s function for a point source above an impedance ground has been obtained using the method of stationary phase. Numerical comparisons show that the new expression is often more accurate than those obtained by more complicated techniques, such as the method of steepest descent. Following the same procedures, a new expression of the Green’s function for a horizontal line acoustic source above an impedance ground is also given.
LS multi-line curved array signal localization
Authors: Bucker Homer
Published: October 2005

Let Fj be the analytic signal received at sensor j of a hydrophone array of unknown shape and let gjk =< FjF*k≳ be an element of the covariance matrix. Above, the * means complex conjugate and < ≳ indicates a time average. If there are N plane wave signals incident upon the array and ĝjk is the expected covariance element after ‘‘sufficient’’ time averaging then ĝjk is equal to ∑nAn2 exp{i(2π/λ)[(xj−xk)cos φn+(yj−yk)sin φn]}. In the above equation, An and φn are the amplitude and bearing of signal n, xj, and yj are the horizontal coordinates of sensor j, and λ is the wavelength. An LS (least-squares) iteration takes place to reduce the error function E=∑j∑k‖gjk−ĝjk‖2 to a minimum. Values of {An}, {φn}, and several harmonic coefficients that define the sensor coordinates relative to straight lines are adjusted to minimize the error function. Several examples will be presented to illustrate the method.
A Bayesian approach to passive acoustic signal processing
Authors: Pitre Richard , Davis Nolan R.
Published: October 2005

This work considers a systematic approach to data inversion and data fusion for stationary passive sonar in low signal-to-noise situations. Bayesian inversion is applied to the probability distributions that are implicit in conventional signal processing methods. The resulting source location probability distributions are multimodal, reflecting the sidelobe structure of conventional ambiguity functions. Using the probability interpretation of these distributions the secondary sidelobe peaks can be compared quantitatively with the mainlobe. Results of model calculations are presented for an ocean waveguide in order to demonstrate the method, provide a comparison with conventional approaches, and assess the performance under low signal-to-noise conditions. A preliminary discussion of data fusion is given for probability distributions derived from inversion of independent data sets. An application to frequency fusion is made, and a performance improvement is demonstrated with the computational model.
The equivalent sources method−Cremer’s last contribution to structure-borne sound
Authors: Heckl M.
Published: October 2005

In his late years Professor Cremer was very interested in violins. After having worked a lot on the vibrations of the strings and the violin body, he wanted to get a better understanding of the radiation properties of violins. As was typical for him, he did not like to use a general numerical method such as the boundary element method; he wanted to find a calculation scheme that does not hide the physics of the problem. So he replaced the radiator by equivalent sources and adjusted their amplitudes so that the radiated sound is approximated to any degree of accuracy. He also showed the close relationship of some aspects of this method to the law of conjugate energies. After Cremer’s original publication it turned out that the calculation scheme he proposed can be applied to many other radiators apart from violins. The method and its extension, as well as some examples (boxes of different shape, railway rails, etc.) will be presented in the paper.
L. Cremer’s contributions to sound transmission through and within structures
Authors: Holmer C. I.
Published: October 2005

Professor L. Cremer was avidly interested in the transmission of sound through walls as part of his practice in architectural acoustics. The frequent use of masonry constructions for partitions and exterior facades in Europe, and their documented departure from ‘‘mass law’’ sound transmission performance led him to study this phenomenon in detail. He first identified what we now call the wave-coincidence effect (i.e., the matching of sound wavelengths in the surrounding fluid with the flexural wavelengths in the panel) as the mechanism which produced the non-mass-law performance in the 1940s. This work influenced London, Beranek, Watters, and Feshback in this country, leading to our present understanding of this phenomenon. His later work in impact sound transmission for masonry floors, and the transmission of flexural waves through wall junctions for flanking transmission also had a pioneering quality. Cremer systematically and significantly altered our understanding of sound propagation in buildings and other structures. This paper will review these several pioneering investigations to help us appreciate this giant of our field, on whose shoulders we all stand.
Responses of green lacewings (Insecta: Neuroptera: Chrysopidae) to modified features in computer-synthesized mating songs
Authors: Henry Charles S. , Wells Marta M.
Published: October 2005

Vibrational tremulation songs of Chrysoperla green lacewings have been shown to have a strong reproductive isolating effect on otherwise identical and interfertile populations, indicating the presence of cryptic, sibling species within the genus. The substrate-borne songs consist of pulses or volleys of abdominal vibration and are characterized by a modulated carrier frequency. To determine which features are essential for species recognition and how much change will be tolerated, synthetic songs were produced on a computer and played back to individual females of the common North American lacewing species, Chrysoperlaplorabunda (Fitch). Responses of those females to selectively modified songs were then tabulated and compared to responses to unmodified songs. Song features tested included the first harmonic and modulation characteristics of the carrier frequency, as well as the length and spacing of the song volleys. It was found that females pay little attention to modest changes in volley duration, volley repetition rate, or fundamental frequency, but discriminate strongly against songs that lack smooth frequency modulation or that consist of random low-frequency noise. The significance of these findings to the evolution and speciation of members of the Chrysoperlaplorabunda complex in North American and Europe is discussed.
Behavioral study of zooplankton response to high-frequency acoustics
Authors: Sullivan-Silva Kilian B. , Forbes Michael J.
Published: October 2005

An experimental investigation of behavioral patterns and responses of zooplankton was performed in Narragansett Bay, RI by the Naval Undersea Warfare Center Division, Newport, with scientists from Woods Hole Oceanographic Institution (WHOI). Normal swim behaviors of zooplankton (unstimulated) were compared to behaviors of zooplankton subjected to high-frequency acoustic pulses (stimulated). Of particular interest was the effect of acoustics on the jump or ‘‘burst’’ behavior of the zooplankton. The zooplankton were videotaped with the WHOI Video Plankton Recorder (VPR), resulting in approximately 30 h of video data from each of the VPR’s four cameras. The cameras were concentrically focused on the ensonified volume of water, with fields of view ranging from 6.3 to 0.5 cm. The dominant species observed was the copepod Acartiatonsa, ranging in size from approximately 0.2 to 1.5 cm. Data analyzed yield an average particle density of 117 000/m3, and overall burst rates per particle of 0.62 bursts/s with acoustics and 0.80 bursts/s with no acoustic stimulation. However, real-time observations and a statistical data assessment reflect no quantitative difference in swim behaviors due to the acoustic stimulation.
Sound spatial transverse decorrelation due to surface wind waves in shallow water
Authors: Wang Qin
Published: October 2005

Transverse decorrelation of sound pressures signals received at horizontally separated points is investigated in shallow water in the presence of surface wind waves. By applying the adiabatic approximation and smooth-average method, the relationship between the transverse correlation and surface-wave frequency-direction spectrum is developed. Both surface scattering and multipath interference are involved. Since surface scattering degrades the correlation but multipath interference enhances the correlation as range increases, three range regions or zones are suggested: (1) near-field region in which normal modes or paths with large grazing angle are important, (2) multipath interference region in which the effective propagating modes or paths are degraded due to bottom reflection loss, and (3) surface scattering region in which the waveguide modes become dominant. The correlation has also been shown to be strongly dependent upon wind speed, bottom reflection loss, and frequency.
Active noise control applied to extended sources in a half-space
Authors: Shepard Steve , Cunefare Kenneth A.
Published: October 2005

Much of the theoretical and computational work to date on the active noise control of extended structures generally considers the subject body to be in free-space. In practice, most structures of interest are located in close proximity to a bounding plane, such as the ground surface. The research presented in this paper extends the existing body of work related to the active control of noise radiated by extended three-dimensional structures to include the presence of a reflective infinite plane, that is, within a half-space. As with point sources, when the structure to be controlled is within one wavelength of the reflective plane, be it rigid or pressure release, the plane can impact the control versus control under free-field conditions. The research exploits the method of images to adapt the existing boundary element techniques for active noise control analysis of extended structures.
In-flight validation of an active noise control system for propeller aircraft
Authors: Lefebvre S. , Billoud G. , Ribet H. , Paillard J.
Published: October 2005

This paper describes an active noise control experimental implementation in propeller aircraft. It relies on synchronous excitation of acoustic sources in order to minimize the sound field measured at passenger head height by a set of microphones. System performance predictions coupled with optimization algorithms showed that 32 loudspeakers and 48 monitoring microphones were needed for achieving satisfactory noise reduction. Therefore, a multivariable (48 inputs–32 outputs) digital adaptive controller, based on the time domain filtered-X LMS algorithm, has been developed. It comprises 20 boards plugged in a VMEBus adding up to more than 200 Mflops. The controller has been tested in a Dornier 228, two 4-blade propeller aircraft, under cruise conditions. An averaged reduction of 11 dB for the 1×BPF, 10 dB for the 2×BPF, and 6 dB for the 3×BPF was achieved, which was in accordance with predictions (12 dB at 1×BPF). The reduction was the most significant in the front of the cabin, where the initial noise level was the highest. The optimum noise reduction was reached within 30 s. [European Brite-Euram Aero project n°0028 ‘‘ASANCA,’’ supported by the Commission of European Community, DG-XII.]
Cross-linguistic preferences in onset-nucleus sequences
Authors: Maddieson Ian , Flemming Edward
Published: October 2005

For the purpose of testing hypotheses concerning cross-linguistic preferences in sound sequences, Maddieson and associates have compiled a database of lexical syllable frequencies in a geographically and genetically diverse sample of the world’s languages. The present study is an analysis of the frequencies of onset-nucleus sequences to assess the prediction that there should be a preference for sequences which minimize articulatory effort, i.e., in which the movement between onset and nucleus is minimized. This hypothesis is one component of Lindblom’s Theory of Adaptive Dispersion. Preferences in a language are determined by deviations from the expected frequency of a sequence determined on the assumption that the individual frequencies of the onset and nucleus are independent. Three classes of models of the preferences based on different approximations to the articulatory distance between onset and nucleus are evaluated. The first measure is in terms of differences in binary feature specifications, the second in terms of categorical articulatory scales, and the third utilizes quantified estimates of actual articulatory distances.
Perceptual boundaries of diphthong-like stimuli
Authors: Czyzewski Zbigniew , Nabelek Anna K. , Crowley Hilary
Published: October 2005

Fourteen-step /ai-a/ continua were generated to investigate effects of: (1) listening conditions (quiet, noise, reverberation), (2) subjects’ hearing (normal and impaired), and (3) position of the end point of F1 and F2 trajectories (target frequencies). Two models of formant trajectories were tested: (1) the model proposed by Gay [J. Acoust. Soc. Am. 44, 1570–1573 (1968)] in which a characteristic and invariant rate of change in formant trajectories is required for diphthong perception and (2) the model proposed by Bladon [Speech Commun. 4, 145–154 (1984)], in which a certain change between the steady state and the target frequency is required for diphthong identification. The variable in this experiment was the length of the steady state of F1 and F2. Stimuli were generated with the Klatt synthesizer. Speech-spectrum noise was mixed at S/N=0 dB. Reverberation was generated by a computer program (T=1.1 s). Subjects, ten normal hearing and ten hearing impaired, were tested individually. Listening condition, especially reverberation, and type of formant trajectories had a significant effect on location of the boundary for both groups of subjects. Diphthongization was preserved better in the stimuli generated according to the Bladon’s model than according to the Gay’s model in all listening conditions. [Work supported by NIDCD.]
Hearing threshold shifts from repeated, 6-h daily exposures to impact noise
Authors: Hamernik Roger P. , Ahroon William A. , Davis Robert I. , Lei Sheau-Fang
Published: October 2005

Recent results of research on the bidirectional transduction properties of the cochlea have opened new possibilities for understanding and explaining various phenomena associated with noise-induced threshold shifts. Several studies using animal models have shown that the cochlea is capable of developing a resistance to noise-induced threshold shift (TS) that is dependent upon previous exposure history and on the timing schedule of a repeated exposure. These results were obtained using continuous octave band stimuli. This paper will describe the developing resistance to TS from intermittent daily exposures to a broadband impact noise. The chinchilla was used as an animal model; the evoked auditory response as a measure of threshold; and the cochleogram used to quantify the sensory cell population. The animals (N=12) were exposed over a 20-day period to a 6-h daily exposure of 113- or 119-dB peak SPL impacts presented 1/s. Thresholds measured at the beginning and end of each exposure period showed that up to 30-dB resistance to TS could be developed over the first 5 days of exposure. The magnitude and time course of the resistance effect was frequency and intensity dependent. The audiometric and histological results of these intermittent exposures will be contrasted with equal-energy exposures using the same impacts but an exposure schedule without interruption. [Research sponsored by NIOSH Grant No. 2R010H02317.]
Functional and structural changes in nucleus magnocellularis (NM) of the neonatal chick exposed to intense sound
Authors: Saunders James C. , Cohen Yale E. , Kazahaya Ken
Published: October 2005

Young chicks were exposed to a 0.9-kHz tone for 48 h at 120 dB SPL. Experimental groups were formed at 0 and 12 days post exposure along with age-matched control groups. Single-cell responses in the NM, cochlear pathology along the basilar papilla, and brainstem neuroanatomy in NM, were evaluated in each animal. At 0 days of recovery, single-cell tuning properties, spontaneous activity, rate-intensity functions, phase locking, BF sensitivity, etc., were all severely altered by the exposure. The tonotopic organization of the NM was unaffected. The cochlea exhibited significant damage to the short hair cells and the tectorial membrane. Further, a significant reduction in the surface area of NM cells was noted. All physiologic dimensions recovered completely within 12 days, and the short hair cells were nearly completely replaced. However, in spite of a normal compliment of physiologic behaviors at the cellular level, there remained a substantial disruption to the tectorial membrane. Moreover, the surface area of the NM cells remained significantly reduced. The relation between abnormal cochlear and neural structures, with normal physiologic function in the peripheral auditory system, is discussed. [Work supported by the NIDCD and the Pennsylvania Lions Hearing Research Foundation.]
Optimizing actuator locations in feedforward active control systems using stepwise regression
Authors: Ruckman C. E. , Fuller Chris R.
Published: October 2005

Using a new technique, the physical locations of control actuators in a feedforward active noise or vibration control system may be optimized for the best possible system performance. The proper choice of actuator locations can have a profound impact on controller performance, but in a complex system the analyst’s intuition might not always pick the most effective arrangement. The most common approach used in the literature, based on numerical minimization software, suffers from various practical drawbacks that make it unsuitable for complex structures or systems with large numbers of actuators. By formulating the feedforward control approach as a multiple regression problem, it is possible to instead use stepwise regression to find which actuator locations will provide the best system performance. This new technique is general enough for use on complex structures that cannot be modeled analytically, and is efficient enough to allow comprehensive studies involving large numbers of actuators. The technique will be demonstrated by numerically simulating a simple system in which radiation from a cylindrical shell is controlled by oscillating forces applied on the shell surface. [Work supported by David Taylor Model Basin and ONR.]
Active structural acoustic control of noise transmission through double panel systems
Authors: Carneal James P. , Fuller Chris R.
Published: October 2005

This work is concerned with an experimental investigation of active control of sound transmission through double panel systems. The technique used is the active structural acoustic control (ASAC) approach where control inputs, in the form of piezoelectric actuators, were applied to the structure while the radiated pressure field was minimized. The double panel system was mounted in a common wall between two reverberant chambers comprising a transmission loss test facility at VPI and SU. Two different panel systems were tested; the first in which both panels were of similar dynamic response and a second in which the radiating panel (of sandwich construction) was much stiffer and had higher resonant frequencies. Excitation of the source panel was induced by reverberant and plane-wave harmonic sound fields. Configurations in which the piezoelectric actuators were bonded to either the source or radiating side panel were tested. A multichannel filtered-X LMS algorithm was used as the controller. Error signals were taken from microphones in the radiated field. Results indicate that a 12-dB attenuation of acoustic radiated power could be typically achieved. The influence of excitation frequency and noise field, radiating plate stiffness, and control system configuration was studied. In particular, the results demonstrate that it is possible to take advantage of the double panel behavior to enhance control performance. In general, this investigation demonstrates the validity of ASAC of noise transmission through double panel systems. [Work supported by NASA LaRC.]
A phase screen approach to sound propagation through small-scale turbulence
Authors: Di Xiao , Gilbert Kenneth E.
Published: October 2005

A Green’s function method has recently been used to derive a fast parabolic (PE) method for atmospheric sound propagation over a locally reacting ground surface [K. E. Gilbert, X. Di, and C. You, J. Acoust. Soc. Am. 90, 2307(A) (1991)]. Because the algorithm can take range steps many wavelengths long, it is approximately 100 times faster than existing PE algorithms that use a Crank–Nicolson range step. In applying the fast PE method to propagation in a turbulent atmosphere, however, one must somehow account for inhomogeneities that are considerably smaller than the optimum range step of the PE. To avoid losing the speed of the method by taking short-range steps, the use of ‘‘integrated-turbulence’’ phase screens has been investigated. On a given range step, the phase screen method incorporates the integrated phase change due to turbulence into a random phase factor that is applied at the end of the range step. The objective is to use long range steps and still accurately account for the effects of small-scale turbulence. Results from the phase screen approach are compared to calculations based on a Crank–Nicolson method which accurately accounts for small-scale turbulence by taking very short-range steps. A theoretical and numerical analysis is given for the maximum range step one can use with realistic models for atmospheric turbulence.
Sound field in a horizontally stratified, moving medium
Authors: Wilson D. Keith
Published: October 2005

A numerical method for computing the sound field in a stratified, moving medium is described. The method also applies to moving sources. As suggested by Raspet etal. [J. Acoust. Soc. Am. 91, 498–500 (1992)], the wave equation is Fourier transformed in both horizontal coordinates, and the solution in the transformed space then is found using vertical finite elements. It is more convenient to formulate the solution in terms of the velocity potential rather than the acoustic pressure or velocity. The finite elements are determined using a global matrix method similar to the well known SAFARI program. Computational resources limit the maximal range of the solutions.
Modeling the mechanical aspects of the cochlea to determine acoustic phonetics
Authors: McCray Talia , Rutledge Janet C.
Published: October 2005

Using the sinusoidal model as developed by Quatieri and McAulay [IEEE Trans. Acoust. Speech Signal Process. ASSP-34, August (1986)] as an analysis tool that operates on both a time varying and frequency-dependent basis, a software design was developed to show how the inner ear resembles a series of bandpass filters. Each speech frame was divided into half-octave bands ranging from 125–4000 Hz. This was done by partitioning sinusoidal parameters based on frequency for the purpose of providing energy profiles. Speech analysis using the energy profiles of the bands was a ready method to determine formant locations for the purpose of identifying vowels, voiced/unvoiced regions, and an approximation of consonants. The software program runs on the TMS320C30 processing chip. Isolated phonemes were spoken into a microphone. For analysis purposes, the data obtained were compared to the corresponding band location of average formant frequencies as found in research conducted by Peterson and Barney [J. Acoust. Soc. Am. 24, 175–184 (1952)]. A more accurate analysis was given by comparing the software output of individual phonemes to the magnitude of the phoneme’s actual Fourier transform. The output of the software program compared almost perfectly.
Acoustic test facilities at Lake Pend Oreille in Idaho
Authors: Guedel George A.
Published: October 2005

The primary mission of the Acoustic Research Detachment is to support the development of advanced submarine stealth technology. The environmentally attractive characteristics of Lake Pend Oreille in Northern Idaho provide near ideal testing conditions in which to accomplish this mission. The lake has a maximum depth of 1158 ft with more than 26 mi2 deeper than 1000 ft. The water temperature below a depth of 200 ft is constant at 39.5 °F. During periods of minimal boat interference, the acoustic ambient noise level is below sea state zero greater than 43% of the time. Six large scale models are available, along with a number of other special test fixtures, to perform experiments in the areas of ship silencing, underwater acoustics, structural acoustics, target strength, hydroacoustics, and hydrodynamics. There are three permanent test sites on the lake, with a fourth site currently under development. Mobile test platforms are also available. Each site provides a separate and unique test capability. All of the facilities are highly adaptable and can be reconfigured to support most test requirements.
Resonance tubes and Choirchimes
Authors: Rossing Thomas D.
Published: October 2005

For some time, the so-called Ames tube, consisting of a tuning fork combined with an open-end resonance tube, has been used in acoustics demonstration experiments. A few years ago, Malmark, Inc. designed a closed-end version with a handbell-type clapper. The ChoirchimeTM has become a very popular musical instrument, especially in elementary schools and churches. Some acoustical features of Ames tubes and Choirchimes are demonstrated, and the ways they can be used in acoustics education are shown.
Chaotic systems as generators of musical raw materials
Authors: Bidlack Rick
Published: October 2005

Many of the features of nonlinear dynamical systems that are intriguing from a mathematical and scientific point of view, especially the affinity such systems have with the behavior of natural phenomena, are likewise suggestive of musical qualities. A variety of musical textures may be elicited from chaotic systems, some quite novel, while others are evocative of historically established practices. Typical is a kind of continuous variation form, in which motivic materials are spun out within a constantly evolving context. Four well-known chaotic systems are discussed in terms of the quality of the musical raw materials they generate, as well as basic techniques for mapping their orbits into musical parameter space.
The geometry of melodic structure: Is there a strange attractor at the heart of Bach’s, Two Part Invention No. IV?
Authors: Casadonte Donald
Published: October 2005

Although certain works of music show melody lines that contain highly developed pitch organization, no study to date has demonstrated that for certain works this pitch organization may be chaotic in nature. In this paper, the principle melodic line for the TwoPartInvention, No. IV (for clavier) of Johannes Sebastian Bach has been analyzed. Taking a decidedly geometric attack, the pitch organization of the melodic material has been treated as a time series. In this particular example of counterpoint, the melodic line weaves continuously back and forth between the two lines of the clavier part. The melodic information has been converted into its associated pitch class numbers, mod 12. By concatenating the resulting numerical information of the two melody lines into one continuous line and treating it as a time series in pitch, it was possible to plot the three-dimensional phase space of the derived time series by use of time-delay plots. The attractor generated by this means, in this special case of musical counterpoint, indeed showed the requisite properties of a strange attractor. In addition to calculating the Lyopanov exponent, dimensional constants, and Fourier spectrum of this attractor, the following fascinating question was discussed: why is this data chaotic and how does it relate to the geometric structure of musical form in general?
A neural network model of recoding for musical stimuli
Authors: Large Edward W.
Published: October 2005

According to music psychologists, listeners internalize characteristic patterns of rhythm, pitch, harmony, and so forth, which are then used to recode and organize sequentially presented musical stimuli. In a previous paper, a connectionist model was proposed for performing the recoding task in music [Large etal., Proc. of the 13th Conf. of the Cog. Sci. Soc. (1991)]. A compressor network recursively recodes musical patterns into fixed-width neural vectors. A reconstructor network decodes the vectors produced by the compressor into facsimiles of the original musical patterns. These mechanisms are trained simultaneously by linking their training sets together using an auto-associative form of backpropagation. It has been shown that the reconstructions produced by the network agree both with empirical data from studies of improvisational music performance, and with reductionist theories of mental representation for music [F. Lerdahl and R. Jackendoff, AGenerativeTheoryofTonalMusic (MIT, Cambridge, MA, 1983)]. Here, the behavior of the neural network is analyzed from a dynamical systems perspective. The analysis helps explain how the model computes reductions of the musical surface, while minimizing loss of information. It also provides insight into the generalization capabilities of this architecture.
Modifications to a parabolic equation algorithm for the prediction of atmospheric sound fields which extend its range and improve its accuracy
Authors: West M. , Sack R. A.
Published: October 2005

A two-dimensional algorithm based on Gilbert and White’s method [West etal., Appl. Acoust. 36, 1–19 (1992)] coded for a 486 PC, has served as a test bed for the following modifications: (i) a new second-order ground boundary condition, (ii) a ‘‘tuned’’ upper absorbing layer, (iii) a novel upper boundary condition, which replaces the absorbing layer, and (iv) a new procedure for calculation of the starter field. The algorithm with the above modifications has also been adapted to operate with range-dependent sound-speed profiles obtained from an isoturbulent atmospheric model. [Work supported by MOD and DRA.]
Underwater acoustics test facility (ATF) at Naval Underwater Warfare Center (NUWC), Newport, RI
Authors: Ledoux Claude A.
Published: October 2005

The acoustics test facility at NUWC Newport, RI is one of the US Navy’s primary facilities for acoustic evaluation of underwater vehicles. The ATF can measure the acoustic characteristics of complete underwater weapon systems as well as individual system components. Typical tests are transducer and structure transfer functions, target strength measurements, self-noise evaluations, and acoustic evaluation versus pressure and temperature. Principal features of the ATF are; a 700 000 gal test tank with nonparallel walls, multiple test setup flexibility, 3-ton weight handling with two X–Y traveling cranes, an automated state-of-the-art test system for data acquisition and reduction, large X–Y–Z and tilt probe positioner, an acoustically transparent pressure vessel, an underwater TV camera, mechanical laser alignment, and exceptional rotation equipment.