bluuebunny/crossref_metadata_2025 huggingface dataset:
Context-independent dynamic information for vowel identification
Authors: Strange Winifred , Jenkins James J.
Published: October 2005

Previous studies have demonstrated that listeners can identify vowels spoken rapidly in stop consonant-vowel-stop consonant syllables even when vocalic nuclei of the acoustic signals are attenuated to silence, leaving only initial and final transitions. In this study, mixed consonant “silent-center” (MC) conditions were generated in which the initial and final transitions of syllables containing the same vowel but different stop consonants were interchanged, thus altering context-dependent formant trajectories. In one condition, silent intervals were appropriate for the original vowel and the final consonant; in a neutral duration MC condition, silent intervals were adjusted so that intrinsic vowel duration cues were eliminated. Identification errors in the MC condition (8%) were not significantly higher than for original silent-center syllables; the neutral duration MC condition produced somewhat more errors (15%), but identification was more accurate than for initial transitions (35% errors) or final transitions (52% errors) presented alone. These results suggest that some perceptually relevant aspects of dynamic information are context independent. Representations of formant trajectories presented in F1-F0/F3-F2 “space” demonstrate context-dependent and -independent aspects of dynamic information. [Work supported by NINCDS.]
An audiovisual investigation of the loudness/effort effect for speech and nonspeech events
Authors: Rosenblum Lawrence D.
Published: October 2005

There is some evidence that loudness judgments of speech are more closely related to the degree of vocal effort induced in speech production than to the speech signal's surface acoustic properties, such as intensity [P. Ladefoged and N. P. McKinney, J. Acoust. Soc. Am. 35, 454–460 (1963)]. Other researchers have claimed that speech loudness can be rationalized by considering simply the acoustic complexity of the signal [R. D. Glave and A. C. M. Reitveld, J. Acoust. Soc. Am. 58, 875–879 (1975)]. Since vocal effort can be specified optically as well as acoustically, a study to test the effort-loudness hypothesis was conducted that used conflicting audiovisual presentations of a speaker producing CVs with different efforts. The prediction was made that if loudness judgments are based on effort perception rather than on simple acoustic parameters, then judgments should be affected by visual as well as auditory information. It is shown that loudness judgments are significantly affected by visual information even when subjects (1) are instructed to base their judgments on only what they hear and (2) cannot detect a discrepancy between the audio and visual components. Moreover, the same results are shown for a nonspeech “clapping” event attesting to the generality of the loudness-effort effect previously thought to be special to speech
An experimental study on the reed vibrations of the woodwind instruments
Authors: Idogawa Tohru , Iwaki Masakazu , Naoi Toshikatsu , Shimizu Michiko
Published: October 2005

Reed vibrations have been observed for the lowest notes of a bassoon, an oboe, a clarinet, and a soprano saxophone artificially blown. The blowing pressure in the artificial mouth used has been increased from 1.5 k Pa to about 10 kPa (20 kPa for the oboe) and decreased. The relative position and contact pressure between the lips and reeds have been fixed during a cycle of changes in the blowing pressure. Conclusions: (1) Many different modes of very stable periodic vibrations are found even under the same blowing pressure. (2) A slow continuous change in blowing pressure causes a sudden transition from one mode to the next without a change in the lip-reed relative position; this in turn gives rise to some change in the blowing pressure. (3) Which mode is excited under a given blowing pressure depends on the sequence of the preceding modes. (4) In the bassoon and oboe, not all the transitions are reversible; with blowing pressure being increased and decreased, the sequence of modes encountered exhibits hysteresis. (5) Wolf-tone-like waveforms have been found in all instruments used.
Wind instrument bores in the time domain
Authors: Ayers R. Dean
Published: October 2005

Recent interest in modeling nonlinear feedback control of the driver has led several researchers to examine Green's functions and reflection impulses for wind instrument bores. These are usually obtained as inverse Fourier transforms of experimental or computational results in the frequency domain. At CSULB, the focus instead has been on developing such waveforms stepwise, using multiple convolutions. This was motivated by the desire to obtain a better intuitive understanding than that provided by just the transmission line model in the frequency domain. Central to this work are such basic concepts as causality and round trip delay time. The intuitive power of this theoretical approach correlates nicely with the cleanness of experimental results obtained directly in the time domain, which allow us to isolate the effects of individual bore elements. The experimental technique employs Wiener filtering to generate a brief acoustic pulse from a piezoelectric element. Resultant waveforms are further processed by a Scientific Atlanta SD-380 spectrum analyzer, with data sent backward through an IBM-PC to average out low-frequency noise. [Work supported by the CSULB Foundation.]
Nonlinear theory of musical wind instruments
Authors: Fletcher Neville H.
Published: October 2005

While many important features of the design and playing of musical wind instruments can be appreciated from a linear theory of their acoustical operation, an understanding of tone quality, loudness, and transient behavior depends almost entirely upon the essential nonlinearity of the sound generation mechanism and its interaction with resonances of the horn. This has been appreciated in a general way since the time of Bouasse, early this century, while Benade has led the renaissance of interest in the topic over the past 25 years. Until recently, available theoretical methods allowed the calculation of behavior only for rather soft playing conditions and consequently for limited nonlinearity. The development of time-domain and mixed-domain “frequency-balance” techniques during the past few years, however, now allows all regimes of playing to be investigated. This paper reviews available theoretical methods and their application to typical instruments.
Kinematical and dynamical factors for deviation from Helmholtz motion
Authors: Weinreich Gabriel
Published: October 2005

In the ideal Helmholtz motion, the string velocity at the bow alternates between two constant values, the ratio of whose magnitudes is equal to the ratio of lengths of the two segments into which the bow divides the string. Deviations from this ideal can be of two types: (a) motions for which the string behavior at the bow maintains its “two-velocity” nature but with a different ratio; and (b) motions that deviate from the “two-velocity” pattern. Kinematically, two-velocity motions of arbitrary ratio are possible if the bowing point is not a rational subdivision of the string; otherwise, if it is defined as a fraction m:n of two integers in its lowest terms, the shortest possible duty cycle (that is, fraction of period for which “slipping” occurs) is 1/n. Small ratios may be favored by stability considerations. Deviations from the two-velocity pattern must occur when dissipation is present, and involve the response of the string to near-resonant periodic forces. Examples observed with the “digital bow” will be shown. [Work supported by NSF.]
Measurement of the bowing parameters in violin playing
Authors: Askenfelt Anders
Published: October 2005

A method is presented that allows the measurement of the bowing parameters in violin playing, without interfering with normal playing conditions. The measured parameters include the force between bow and string (“bow pressure”), the position of a contact point between bow and string along the bow (“bow position”), and the distance between the contact point and the bridge (“bow-bridge distance”). Typical registrations for a sample of bowing patterns will be presented, and the violinist's use of the bowing parameters in controlling the dynamic level will be discussed.
Cell manipulation by radiation forces
Authors: Coakley W. Terence , Bardsley David W. , Grundy Martin A.
Published: October 2005

Particles suspended in acoustic fields experience nonzero time averaged radiation forces that can cause particle movement, interparticle attraction or repulsion, and can exert a torque on suspended particles. In a standing wave, such field forces result in migration of the particles to positions separated by half an acoustic wavelength. The forces depend on some or all of the following properties: particle size and shape, sound frequency, the square of the sound-pressure amplitude, and the differences between the density and compressibility of the particles and those of the suspending phase. Threshold pressure for the vertical migration of particles to half-wavelength separations is derived. It is shown that appropriate choice of sound-pressure amplitude should lead either to the separation of particles of different densities and compressibilities or to the concentration of mixed samples essentially at the same location. Applications of radiation forces to bring cells together in suspension for cell electrofusion procedures and the use of ultrasound to support cells away from solid surfaces are described. Examples of cell concentration, cell sedimentation, cell alignment, and cell-cell interaction are also presented.
Dynamic characterization of a radio ear bone vibrator with laser vibrometry
Authors: Pechersky Martin J. , Stuart A.D. , Bufalini M. R.
Published: October 2005

The vibrational characteristics of a model B-71 bone vibrator were determined using a laser vibrometer. The bone vibrator is normally used for auditory evoked potential testing. The purpose of these measurements was to characterize the performance of this device between a frequency range of 500–5000 Hz. The dynamic characteristics of the bone vibrator were measured with the device mounted on a Beltone model M5B artificial mastoid. Frequency response functions were determined for the bone vibrator/mastoid system at several locations. Subsequently, transfer functions between the driving voltage and the seismic mass of the bone vibrator, the case of the bone vibrator, and the artificial mastoid were determined, as well as the transfer function between the seismic mass and the mastoid. The vibrations of the bone vibrator were obtained with a high degree of spatial resolution by focusing the vibrometer laser beam on the location of interest to obtain the instantaneous velocity at that point. The mastoid response was measured with a conventional accelerometer. Besides the resonance associated with the seismic mass, two structural modes of the bone vibrator case were identified in the frequency range of interest and the mode shapes of the case vibrations were investigated.
Effects of air gun energy releases on the northern anchovy
Authors: Greenlaw C. F. , Holliday D. V. , Pieper R. E. , Clark M. E.
Published: October 2005

Eggs, larvae of several ages, and adult specimens of the northern anchovy were exposed to acoustic energy from air gun seismic sources. Unlike earlier studies that exposed specimens to one or a few shots from the source, these exposures used a series of air gun shots using devices of various chamber sizes, simulating realistic exposures that might be experienced by wild fish in the vicinity of a seismic survey vessel. Exposure protocols ranged from the equivalent of single, distant passes of a typical seismic array to multiple passes at very short range. Air gun charging pressures were varied to simulate changing ranges. Histological examination of exposed specimens revealed no evidence of gross morphological damage caused by exposure. Comparison of survival with control groups showed subtle effects in the younger (2–4 days) larvae. Exposure of adults resulted in some damage to swimbladders, particularly for fish exposed at the surface where water particle motion effects are pronounced, but no significant effects on otoliths were found. Extrapolation of the survival and histology data suggests that noticeable impacts on eggs and larvae of this fish would result only from multiple, close exposures to seismic arrays. [Work supported by API and the California Egg and Larvae Committee.]
Graduate study in acoustics at the Georgia Institute of Technology
Authors: Berthelot Yves H. , Ginsberg Jerry H. , Jarzynski Jacek , Rentzepis George M. , Rogers Peter H.
Published: October 2005

The graduate studies at Georgia Tech span across three colleges, Engineering, Liberal Arts and Sciences, and Architecture and seven schools. Starting with a sequence of three courses on the foundations of physical acoustics, courses in underwater acoustics, atmosphere acoustics, nonlinear acoustics, aeroacoustics, jet noise, noise control architectural acoustics, waves in elastic and viscoelastic media, nonlinear wave propagation, bioacoustics, and vibrations, are available for the formal educational development and preparation for research in acoustics. Opportunities for research across campus are plentiful in such areas as combustion, aerodynamic and helicopter noise, machinery noise, low-noise environmental design, etc. More specifically, at the school of Mechanical Engineering, studies currently in progress are in variational methods in wave propagation and in particular acoustics, light-sound interaction and laser applications to sound detection, fluid-structure interaction and sound propagation, radiation mechanisms, active vibration control, scaling effects, characterization of materials by waves, auditory systems of fish, provide a wealth of graduate thesis topics with a faculty actively engaged in these areas. In addition to the graduate degrees, the institute offers the opportunity for specialized study in acoustics through its multidisciplinary certificate program. Under this program, those who are not as heavily committed to graduate studies in acoustics can be exposed to the field and receive an acoustical engineering or acoustics certificate.
Graduate study in acoustics at Northern Illinois University
Authors: Rossing Thomas D.
Published: October 2005

A master's degree is offered in basic or applied physics with specialization in acoustics. Research opportunities in the Acoustics Laboratory include vibrational physics, musical acoustics, psychoacoustics, and environmental noise control. Additional research opportunities exist at two national laboratories and several industrial research laboratories nearby. A doctoral program is in the final stages of development.
The graduate acoustics education program in the School of Mechanical Engineering at Purdue University
Authors: Bernhard R. J. , Bolton J. S.
Published: October 2005

The Acoustics and Noise Control Research Program of the School of Mechanical Engineering at Purdue offers educational and research opportunities at both the undergraduate and graduate level. In addition to undergraduate elective courses in noise control and vibration measurement and control, two graduate engineering acoustics classes are offered within the school. In addition, complementary graduate courses are offered in the areas of digital signal processing, using mechanical vibrations of discretized systems, vibrations of plates and shells, intermediate fluid mechanics, theory and design of control systems, digital control, numerical methods in mechanical engineering, and finite and boundary element methods. Graduate students also have access to a broad spectrum of courses taught in the School of Electrical Engineering and the Department of Physics. Eight faculty members and approximately 20 graduate students participate in the Acoustics and Noise Control Research Program. Most of the research is conducted at the Ray W. Herrick Laboratories. The facilities of the laboratories include an anechoic chamber with a useful working volume of 50 m3, a reverberation room with a volume of 200 m3, and a semi-anechoic chamber with a useful volume of 550 m3. Current research investigations consider: active noise control, active vibration control, structure-borne noise, nonlinear system modeling, numerical noise control design methods, acoustical imaging, modeling of acoustical foams, signature analysis and machinery monitoring, noise source identification, and application of noise control techniques to automotive powertrains and large refrigerant compressors.
Graduate education in underwater acoustics at Tokai University
Authors: Nishimura Minoru
Published: October 2005

Research and education in underwater acoustics and its application are indispensable for ocean exploration and development. In 1963, the faculty was established and several subjects on underwater acoustics were included in the undergraduate curriculum. In 1967, the Graduate School of Marine Science and Technology was opened and advanced subjects on underwater acoustics were introduced. Although few students study underwater acoustics, one or two students obtain a master's degree every year, and also five students have taken the degree of Dr. of Engineering in the past 6 years. Studies are mainly directed toward the ultrasonic characteristics of seabed sediment, sound reflection from fish, influence of sound on fish behavior, and nonlinear acoustic characteristics of a focused sound fields. One of the biggest problems is how to make underwater acoustics attractive to the students. The reason that students are less concerned with this field is because it appears unfashionable when compared with electronic studies such as computer science, etc. Another problem is that sea research is physically hard due to the conditions and because it requires much expense and specially designed devices.
AI and signal understanding systems
Authors: Reddy Raj
Published: October 2005

Signal interpretation in the presence of noise and uncertainty has always been a difficult problem. In the past 2 decades, a number of application areas have found it necessary to use symbolic knowledge about the situation, context, and language in the form of constraints to reduce exponential search and impove accuracy. This work will provide several examples from the areas of speech, image understanding and target identification, music analysis and recognition, and sensor-based diagnosis in expert systems where symbolic knowledge plays an important role in the interpretation of signals.
Machinery diagnostics
Authors: Lyon Richard H.
Published: October 2005

Machinery diagnostics means using vibration and/or acoustical signals to determine the performance parameters of the mechanisms and structure of a machine. The signals can be processed in a number of ways to extract the desired information, and commercial systems exist that are able to diagnose in limited but important situations. The purpose of this paper is to review some of the issues that need to be resolved in terms of the knowledge about mechanisms and structures, and signal processing procedures to deal effectively with situations that current systems do not address. These issues have to do with structural complexity and variability, the separation of received information into “source” and “path” components, and the association of faults with particular signatures. Some experiences with the development of customized diagnostic systems are included in the discussion to bring these issues down to earth.
An AI application in acoustic signal processing for target tracking
Authors: Dorrough Douglas C.
Published: October 2005

A spectrum of knowledge-based systems has been designed and applied to target acquisition, classification, and tracking. A survey of these leads to the conclusion that more profound techniques need to be developed and applied. Particularly necessary are techniques that match the performance of a highly trained sensor suite operator. Preferably, such a system would function autonomously. A knowledge-based system (called “PING”) for autonomously classifying and tracking underwater targets in terms of their acoustic signals as well as their context was designed. PING is based upon earlier work in the foundation of analogical reasoning. The performance of PING was tested and compared with the performance of a population of highly trained sonar operators; the system was also compared with one employing solely the Demster-Shafer evidential reasoning method. In both tests, PING exceeded the performance of the test controls.
A score transcription system using musical knowledge
Authors: Hirata Keiji , Aoyagi Tatsuya
Published: October 2005

Automatic score transcription systems have two problems. (1) How can a machine make a score from the results of an FFT? Particularly in the case of polyphonic sound, it is hard to decide whether peaks are the overtones or the fundamentals of the actually played notes. (2) How can a machine reduce the amount of FFT computation? Usually the FFT is performed while shifting the FFT window by a small amount of time. This results in a great deal of computing time consumed by the FFT. To overcome the above problems, musical knowledge is used. Assuming that an input signal is a musical sound, a system can eliminate unnecessary possibilities and wasteful computation. For instance, a system uses musical knowledge about reasonable notes on current scales. Further, the parameters for FFT computation can be optimized with musical knowledge. The musical knowledge decides the degree of the musical importance of data; according to this degree, this system can adjust the time and frequency of the FFT computation. For example, since the onset of a note is more important than its decaying part for the score transcription, the system focuses its computing power onto the onset part.
Spatialized computation in sound and music
Authors: Mont-Reynaud Bernard
Published: October 2005

The PRISM system (for Pattern Recognition In Sound and Music) is the acoustic front-end of an experimental research environment that attempts to imitate qualitative aspects of human hearing; examples include polyphonic music recordings. This paper focuses on the acoustic representation used in PRISM, an alternative to spectrograms well suited for perception work; the method used to compute the representation is described elsewhere. An ideal representation for analysis offers mathematical elegance, ease of computation, psychological plausibility, and perceptual immediacy. Linear time suits most purposes, but psychoacoustic sensitivity studies suggest a quasilogarithmic frequency scale; and psychological, musical, and mathematical reasons lead to using log (f) for the pitch scale. With the latter choice, pitch invariance maps to translation invariance; and musical intervals, scales, and harmonic series are patterns revealed by straightforward convolution, or by the naked eye. This leads to a promising family of algorithms for polyphonic pitch detection and other fundamental acoustic analysis tasks. Intuitive user interfaces and visualization tools are obtained from the representation. Last but not least, the algorithms map readily to most massively parallel architectures. [Work supported by NSF.]
Detection performance degradation due to signal uncertainties
Authors: Quazi Azizul H. , Nuttall Albert H.
Published: October 2005

Detection of a known signal in white noise using a matched filter depends on the signal-to-noise ratio (signal energy to noise spectral density). As the uncertainties of signal characteristics, such as phase and amplitude, increase, the required SNR must be increased in order to maintain the same detection performance. However, in both cases of known and unknown signals, it is usually tacitly assumed that the signal covers the total bandwidth of the detector and the total integration time. But, in real ocean environments, it is difficult to know the exact signal arrival time and the exact bandwidth required to cover the wide range of Doppler shifts. As a result, the actual detector may utilize a wider bandwidth, and the integration time may extend beyond the signal duration. This paper addresses the problem where the signal duration L is less than or equal to the actual integration time T, and the signal bandwidth B is equal to or less than the actual bandwidth W of the detector. The performance loss is shown as a function of TW/BL as it varies over a range of 1 to 1000, for various combinations of probability of detection and probability of false alarm. [Work supported by NUSC special initiative program.]
Acoustic source imaging using numerical models
Authors: Bernhard Robert J. , Gardner B. K.
Published: October 2005

An acoustic imaging method is formulated based on boundary element numerical approximations to the Helmholtz integral equation. A geometric model of the source and any important environmental features such as scatterers or enclosures are used. The numerical model quantifies the interrelationship of various parts of the source and the interaction of the source and its environment. Known boundary condition data and field pressure data are used to specify the problem. The unknown boundary conditions of the acoustic source are solved. The boundary solution describes the acoustic source. Using the boundary solution, other desired information can be numerically computed to further describe the interaction of the source with its environment. The method is intended to complement nearfield acoustic holography for sources of general shape and for applications in reactive acoustic fields or where scatterers are present. The method is more computationally intensive than holographic methods but is more versatile and requires less measured data. Results will be shown for the method applied to sources in enclosures. In addition, the sensitivity of the method to random and bias errors will be illustrated. [Work supported by NASA Grant NAG1-58.]
Backward reconstruction in acoustical holography based on iterative inversion techniques
Authors: Bai M. R. , Pate A. L.
Published: October 2005

Reconstruction of a source surface motion from acoustic holographic data presents computational difficulties because the problem is ill-posed as is well known in the literature. In order to deal with these difficulties, a method based on a recursive algorithm was developed. In this method, the inverse problem is convened to a well-posed forward propagation problem. An initial guess regarding the source images is required to activate the iterative inversion method. Then, the tentative image is forward propagated to the hologram plane and the residue is determined. Next, a feedback operator is used to process the residue by which the image is updated. Two types of feedback operators were investigated: (1) Wiener type, suboptimal operator, and (2) dynamic, optimal operator (designed subject to minimum mean-square error optimization criteria). Both iteration methods provided satisfactory convergence. In addition, these methods were found to be relatively insensitive to the choice of the initial guess and the noise parameters used in feedback operators. The efficiency of the iteration process is greatly enhanced by 2-D FFT. Numerical and experimental results obtained by using iterative inversion techniques will be presented. The iterative inversion techniques will be compared with conventional inverse filtering methods. In addition, performance of the iterative inversion methods when used in connection with the nearfield acoustic holography technique will be discussed. [Work supported by NSF.]
The use of nearfield acoustic holography in noise control applications
Authors: Dietschi U. D. , Bolton J. S.
Published: October 2005

Nearfield acoustical holography (NAH) allows a radiated sound field to be reconstructed from complex pressure measurements made on a plane in the nearfield of a source. In this way, source normal surface velocities and surface normal intensities may be determined. The latter quantity may be used to locate radiating acoustical sources and so guide noise control treatments. However, the availability of NAH systems has been limited by their cost and complexity. In addition, all current systems (with the exception of that developed by Hald and Ginn) require explicit or implicit knowledge of the forcing function that results in the sound radiation in order to derive the complex pressure field; this information is not normally available to the noise control engineer. At the Herrick Laboratories, a system addressing both of these concerns has been built using instrumentation commonly available in acoustics laboratories. It may be considered to be a single reference implementation of the cross-spectral method developed by Hald and Ginn. Measurements have been made of both ideal sources and a small appliance. The results have been shown to compare well with theoretical results and with conventional intensity measurements.
A normal-mode model for calculating bistatic reverberation in shallow water
Authors: Ellis Dale D. , Crawford Anna M. , Wolf Stephen N.
Published: October 2005

A model for calculating shallow-water reverberation in bistatic geometries has been developed using normal modes and ray-mode analogies. The acoustic propagation from the source to the scattering area, and from the scattering area to the receiver is described using normal modes. The scattering at the sea-surface or sea-bottom interface is described by a ray-type scattering function, assumed to be known. Ray-mode analogies are used to relate the modal wavenumbers to vertical angles, and to obtain an expression for the scattering area. Travel times are determined from the modal group velocities. The theoretical formulation applies to fairly general environments; a model for a Pekeris-type environment has been implemented for bottom reverberation due to short pulses. Results will be compared with measured reverberation from an explosive source received on a horizontal line array.
High-resolution remote sensing of geoacoustic properties of the seabed by a buried OBS
Authors: Yamamoto Tokuo , Badiey Mohsen , Trevorrow Mark , Turgut Altan
Published: October 2005

An entirely new geophysical inverse theory has been developed to remotely measure the shear modulus versus depth profile of the seabed sediments using the gravity wave induced seabed motion [Yamamoto and Torii, Geophys. J. R. Astron. Soc. 85, 413–431 (1986)]. From several experiments conducted in shallow water (4–135 m), shear modulus profiles were extracted down to 200-m sediment depths, and at a resolution of 1 m [e.g., Trevorrow et al., Geophys. J. R. Astron. Soc. (to be published)]. There is a unique relation between the shear modulus and the porosity of sediment at a given overburden pressure. The porosity versus sediment depth profile is uniquely determined from the shear modulus profile using this relation. Finally, Vp, Vs, Qp−1, Qs−1 versus sediment depth profiles are obtained through the Biot-Yamamoto theory [Yamamoto and Turgut, J. Acoust. Soc. Am. 83, 1744–1751 (1988)]. Comparisons between the BSMP remote sensing results and borehole data at five AMCOR and other geophysical borehole sites show excellent agreement. Prediction of transmission loss using BSMP geoacoustic data agrees very well with experiments conducted at the New Jersey Shelf. [Work supported by ONR Code 11250A.]
Seismoacoustic measurements within the sedimentary subbottom of the Ligurian Shelf
Authors: Muir T. G. , Akal T. , Guidi G. , Michelozzi E.
Published: October 2005

A three-axis seismometer was deployed at a depth of 6 m in a mud sediment by first making a 20-cm-diam hole in the bottom, using a bottom-mounted “drilling” machine, inserting the seismometer, and then removing all machine components, allowing sediment to cave in around the sensor. An identical seismometer was deployed on the sediment/water interface. Data were acquired with both ambient noise and explosive shot sources detonated on the bottom as a function of range to 3 km in a water column 16 m deep. While the bottomed seismometer data showed the existence of Sholte-type interface waves present on both horizontal (radial) and vertical geophones, the subbottom seismometer data show signals predominantly on the vertical geophone. The results are analyzed and modeled with the SAFARI code.
Development of a compact apparatus to measure both the complex bulk modulus and rise complex mass density of a fluid contained in a rigid porous solid
Authors: Grant Steven D. , Baker Steven R. , Wilson Oscar B.
Published: October 2005

Two methods for determining the complex bulk modulus B and the complex fluid mass density ρ of a fluid contained in a rigid porous solid are being investigated. The fluid-filled solid is contained within a small (≪λ) cylindrical cavity capped on each end by identical transducers. One method, suitable for acoustically “hard” transducers, is based upon a direct application of Newton's second law. Here, B and ρ are extracted from the pressure at each transducer which results from its known velocity when the transducers are caused to oscillate, “in-phase” and “out-of-phase,” respectively. The other method, suitable for acoustically “soft” transducers, is based upon electroacoustic network theory. The B and ρ are obtained from the input electrical impedance of each transducer when they are wired in parallel “in-phase” and “out-of-phase,” respectively. A description of the apparatus and preliminary experimental results will be presented. [Work sponsored by NRL-USRD.]
Modal scales of the Dan Tranh
Authors: Keefe Douglas H. , Nguyen Phong , Burns Edward M.
Published: October 2005

The Dan Tranh is a Vietnamese 17-string zither tuned to pentatonic modal scales. Each mode encompasses a given tuning and a specific mental expression or modal nuance, and ornamentation using pitch bends is an important embellishment. The Dan Tranh was tuned entirely by ear, and the repetition rate of the lowest-pitched string varied from G3 to E4 in data obtained over a 6-month period from a single artist (P.N.). The octave stretch was approximately 13 cents over 3 octaves with most of the increase in the upper octave, and was subtracted out to define a mean modal tuning. The intervallic distances and standard deviations in cents relative to the tonic of each mode are: Bac = [0, 188 ± 7, 500 ± 9, 698 ± 7, 891 ± 4], Ho Mai Nhi = [0, 166 ± 10, 513 ± 5, 695 ± 9, 859 ± 9], Oan = [0, 338 ± 9, 503 ± 6, 699 ± 6, 1026 ± 15], and Vong Co = [0, 375 ± 13, 519 ± 8, 704 ± 10, 866 ± 6]. The octave stretch and tuning variability are smaller than comparable data obtained in Western and other music cultures. The Vong Co mode strongly and the Ho Mai Nhi mode weakly violate coherehcy as defined by Balzano (1980), but two additional pitches in the Vong Co mode obtained through ornamentation restore coherency. The artist's ability to categorize isolated melodic intervals in a laboratory context will be discussed. [Work partially supported by NIH.]
Birdsong—A quantitative acoustic model
Authors: Fletcher Neville H.
Published: October 2005

Most discussions of birdsong are at the descriptive level or, at best, involve only a qualitative discussion of the acoustical mechanisms involved in sound production. Qualitatively, two different types of song can be distinguished and these may be described as “voiced” or “whistled,” respectively, though these terms do not necessarily refer to the sound-production mechanism. There is general agreement among biologists that the “voiced” song is produced by vibration of the membranes of the vocal organ (the syrinx) under the influence of aerodynamic and pressure forces, though hitherto no quantitative acoustical analysis has been attempted. The present paper puts forward such an analysis using the large body of knowledge that has been built up on the study of reed-blown musical instruments. It is shown that this approach allows a quantitative description to be formulated which yields pressure and flow waveforms and thus the form of the radiated spectrum of the song and a value for the total radiated power. Results are in quite good agreement with observations. The inability of the model to yield the nearly pure-tone sounds characteristic of “whistled” birdsong suggests that this song may be produced by an entirely different mechanism and may be truly whistled.
Quantitative explanation for the “flattening effect” on bowed string instruments
Authors: Boutilion Xavier
Published: October 2005

McIntyre et al. [J. Acoust. Soc. Am. 74, 1325–1345 (1983)] have shown through computer simulations that the “flattening effect” which occurs above a certain bow pressure level is a consequence of the hysteretical character of the stick-slip process but their time-domain analysis only indicates qualitatively why an overall increase in the period should occur. Using a frequency-domain analysis, it will be shown that the area delimited by the frictional characteristic and the string characteristic impedance is related to the spectral composition of both force and velocity at the bowing point. Because these spectral contents are also related by the string admittance at the bowing point, one can predict the flattening effect, provided that the admittance peaks are approximately harmonic. The dependence of the effect on bow velocity, termination impedances, and other parameters will be exhibited. [Work supported by NSF, CNRS, and French Ministry of Culture.]
Transposing device(s) for musical instruments
Authors: Small Terence S.
Published: October 2005

The agonies of a concertizing musician performing upon acoustically imperfect instruments are legend. Traditional instruments (nonelectronic) continue to be manufactured largely based on the dimensions and other data derived from early times, typically through trial and error procedures. Contending with severe intonation flaws, transpositions of various sorts, and U.S. versus European pitch standards are among a performing artist's major frustrations. Efforts to achieve greater control of tone quality, pitch levels, and the like include G. Oliveri's extension core (1897), Alberti, Bender, Gemeinhardt, and a host of other inventors all striving to improve upon existing instruments. All inventions, including the writer's, deal with means by which the performer may selectively modulate the pitch and tonal quality of the instrument. In every case, the bore of the instrument is seen as the major area for manipulation, whether by altered air flow pattern or by actual real-time changes in bore dimensions. In the latter case, implications for radical instrument redesign are seen. Live musical demonstrations of a variety of these devices, with redesign commentary, form an integral portion of the presentation.
Comparison of double and triple cross correlation for arrival time identification of amplitude- and frequency-modulated acoustic transient signals
Authors: Ioup Juliette W. , Ioup George E. , Field Robert L. , Leclere James H.
Published: October 2005

The triple cross correlation of three signals is a simultaneous function of two lags. It is an alternative to cross correlations taken two at a time for determining the lags for a given source at three distributed sensors. It should offer improvement in arrival time identification only when the statistics of the signal have significant third moment components. In this study, amplitude- and frequency-modulated snythetic transient signals are propagated over several possible paths to three sensors, and the triple correlation of the received pulses computed, as well as the cross correlations of the same three signals two at a time. The efficacy of these two approaches is compared for a variety of amplitude- and frequency-modulated transient signals and multipath interference conditions
An experimental comparison of the tonal qualities of a Chinese flute and a Western flute
Authors: Jie Pan
Published: October 2005

The Chinese flute has an oval, membrane-covered hole between the embouchure hole and the first tone hole. This paper presents spectrum measurements of individual notes from a Chinese flute and from a Western flute. The results indicate that the flute membrane enhances the harmonies of the flute notes and results in richer tones.
An adaptive noise control system in air-conditioning ducts
Authors: Hamada Hareo , Miura Tanetoshi , Takahashi Minoru , Oguri Yoshitaka
Published: October 2005

This paper discusses an adaptive control system for the active cancellation of acoustic noise in air-conditioning ducts. Recently, in several systems for active noise control, efforts have been focused on the use of adaptive digital filters to implement the system controller used to produce the artificial sound. The approach used in the newly proposed model assists in canceling broadband noise and also helps in improving the adaptation speed for colored noise. This model consists of a system identification (IDT) process and an adaptive noise cancellation (ANC) process. Two stand-alone types of adaptive controllers using the fast-least-mean-squares (FLMS) algorithm and the variable-step least-mean-squares (VS-LMS) algorithm were used in various experiments. Results of experiments in actual air-conditioning ducts are presented.
Compression system stability enhancement using active control
Authors: Epstein A. H. , Greitzer E. M. , Guenette G. R.
Published: October 2005

All compression systems can suffer from severe dynamic instabilities, which stand as absolute limits to the system performance. The zeroth-order instability is a planar mass flow disturbance known as surge, which involves the entire pumping system [compressor or pump, ducting, plenum(s), throttle]. The high-order instabilities are called rotating stall and are local to the turbomachinery blading. Nonlinearities couple the modes together and in many practical machines rotating stall triggers surge, resulting in a large loss in performance and possible damage to the system. A large amount of effort has been expended over the last 35 years to understand these phenomena and engineer around them. More recently, work has begun on the use of active feedback control to artificially enhance compression system stability and prevent (or delay) rotating surge and stall, with particular application to aircraft engine compressors. Surge has been successfully controlled in small centrifugal compressors yielding a 20–50% increase in operating range. Work on the stabilization of rotating stall in axial compressors is ongoing. These are very complex systems, involving hundreds or thousands of individual airfoils. Much of the work, therefore, necessarily involves signal identification in a noisy environment using sparse sampling techniques and wavelaunch with relatively few actuators. Since aircraft compressors can consume tens of megawatts, control power requirements are of concern. Analysis has shown that, in fact, the control power requirements are not tied to the size of the machine controlled but rather to the power of the destabilizing perturbations introduced to the system. Typically, the control power required is only 103 105 of that of the machine power and, in some cases, may be zero (the instantaneous controller power input can be either positive or negative). This paper reviews the state of the art in active compression system stabilization and is a progress report on ongoing efforts.
Application of a digital filter to active noise control in a duct
Authors: Kuraya Satoshi , Nishimura Yoshitaka , Usagawa Tsuyoshi , Ebata Masanao , Okda Josuke
Published: October 2005

The possibility of active noise control and its frequency range are discussed using sound-source impedance. When a digital filter is used in an active noise control system, the delay time required in the filter must be longer than those in AD-DA converters and low-pass filters. In order to make a two-microphone active control system possible and to lower its frequency range, it is required that (1) the distance between the microphone set near the original source and the loudspeaker used for an additional source should be far; and that (2) the sampling frequency should be high. However, as the number of digital filter taps is fixed, there is a limitation on the sampling frequency in order to realize the characteristics in the low-frequency range. In the proposed system with motional feedback, the active control is realized in a frequency range lower than the fundamental resonance frequency of the loudspeaker, f0. Thus f0 must be raised in order to realize a system at a higher frequency range.
Adaptive digital filter configurations for active control of lightly damped systems
Authors: Waters D. E. , Bernhard R. J.
Published: October 2005

The application of system identification techniques using adaptive digital filters to achieve active noise control in one-dimensional duct systems has been previously investigated [J. C. Burgess, J. Acoust. Soc. Am. 70, 715–726 (1981), L. J. Eriksson, Ph.D. thesis, University of Wisconsin, Madison (1985)]. In this investigation, efficient system identification techniques using adaptive digital filters for active control of lightly damped systems at low modal density are sought. Three cases are considered: (1) where feedback to the detector sensor exists but where the error path is negligible, (2) where the error path exists but feedback is negligible, and (3) where both feedback and error paths are important. In each case, the active controller uses two system identification algorithms in a manner similar to that developed by Eriksson. The first algorithm models the error path by driving the secondary source with its own random noise generator and sensing the resulting response at the error transducer. The error path model is shared with the second system identification algorithm which uses a detection transducer signal, the error signal, and the estimated error path to model the plant (and feedback, if necessary). In each case, both FIR and IIR filters are considered. [Work supported by Nelson Industries, Stoughton, WI.]
An active noise canceling microphone
Authors: Warnaka Glenn E.
Published: October 2005

This paper describes a noise canceling microphone that uses the principles of active noise control. Both physical (acoustic) cancellation, by means of a loudspeaker, and electronic cancellation have been applied to the problem. The results show that a high level of performance may be obtained while removing the requirement to have the microphone extremely close to the lips. Excellent cancellation was achieved at a distance of 24 in. from the signal source using tones, harmonic noise, pink noise, and a variety of cockpit noises from a helicopter, a turbo-prop, and a jet fighter. Conventional microphones may be used to implement the active noise canceling microphone, or a conventional noise canceling microphone may also be used as a basis for implementation of this concept. The active noise control microphone is superior to conventional noise-canceling microphones in rejecting background noise.
Active noise control on systems with time-varying sources and acoustical elements
Authors: Eriksson L. J. , Allie M. C. , Bremigan C. D. , Gilbert J. A.
Published: October 2005

An active noise control system has been developed based on a system identification concept using a recursive adaptive filter to model the direct and feedback elements of an acoustic system. An independent random noise source is used with a second adaptive filter to provide a model of the output transducer and error path transfer functions. This system has the ability to continuously cancel narrow band and broadband noise. The system identification configuration combined with adaptive filter techniques enables the system to respond quickly and accurately to changes in time-varying sources, such as frequency or amplitude, and changes in the acoustical system, such as temperature. Noise reduction results will be presented. The system capabilities will be demonstrated on a variety of time-varying sources and time-varying acoustical elements. The ability of the system to track these changes in real-time as well as implications for actual applications will be discussed.