bluuebunny/crossref_metadata_2025 huggingface dataset:
Modified DigiSound-16 becomes a precision digital oscillator
Authors: Hartmann W. M. , Edmunds D. L. , Atkinson T. V. , Chamberlain Hal
Published: October 2005

The Digisound-16 is a two-channel, 16-bit DAC-ADC system with a 32K word buffer. Because of its high performance and low cost it is an attractive signal source for psychoacoustic experiments. The device can be modified to serve as a precision digital oscillator by using the technique of fractional addressing [W. M. Hartmann, J. Acoust. Soc. Am. 82, 1883-1891 (1987)]. The frequency resolution is 0.003 Hz over the audible range and the total distortion approaches the theoretical limit of − 85 dB for single-channel operation. The modification requires that the 15-bit address register be replaced by a 24-bit increment-and-add circuit. Two-way communication between the modified device and the host computer allows the address increment to be changed continuously so that the oscillator can be manually tuned, as in the method of adjustment. The portion of the buffer that is recycled can be reduced by factors of 2, under program control, to as little as 2K words. The reduced buffer length promotes efficiency while compromising frequency resolution and distortion figures. [Work supported by the National Institutes of Health.]
Demonstration of auditory impedance/reflectance measurement technique, and MIDI enters the research lab
Authors: Keefe Douglas H.
Published: October 2005

A microcomputer-based implementation of a two-microphone wavetube technique for measuring acoustical impedance and energy reflectance will be demonstrated. Data can be fully analyzed with graphical output while the subject is present, thus giving the researcher important flexibility. The Musical Instrument Digital Interface (MIDI) is a standard in electronic music with potential research applications in the areas of acoustical stimulus generation, instrumentation control, and data collection. Selected applications will be demonstrated on the microcomputer with MIDI peripheral devices, and a MIDI software library designed for integration into the research setting will be described. [Work partially supported by NINCDS.]
A computer interface for psychophysical and speech research with the Nucleus cochlear implant
Authors: Shannon Robert V. , Adams Doug D. , Ferrel Roger L. , Palumbo Robert L. , Grandgenett Michael
Published: October 2005

A custom interface was designed and implemented that, under program control, allows presentation of any available pulse stimulus to a patient fitted with the Nucleus implant. The interface connects to a standard parallel output port of a PC or AT compatible computer. The host computer sends a stream of bytes to the parallel port that specify the configuration of the desired output pulses. Upon receipt of the data, the interface generates the appropriate pulse-coded sequence to deliver the specified pulses to the patient's external coil. This interface makes it possible to interleave pulses on two or more electrodes, to modulate the amplitude or timing of a pulse sequence, or to sweep a stimulus across the electrode array. This interface allows investigators to conduct psychophysical and speech experiments that cannot be performed with the standard speech processor interface (SPI) normally used to set the patient's device. [Work supported by NIH.]
IDENTIFY: A program for speech recognition, identification, and categorization experiments on PC compatible computers
Authors: Shannon Robert V. , Palumbo Robert L. , Grandgenett Michael
Published: October 2005

IDENTIFY is a program written in Turbo Pascal that accepts any combination of up to 100 speech waveform files as stimuli and up to 25 response categories. The waveform files can be from either ILS or C-Speech, or any compatible format. The program presents the stimuli in block-randomized order and waits for the subject to respond on the touch sensitive surface of a small digitizing tablet (Koala Pad). The tablet can be configured to reflect any number and pattern of desired responses. The most common case is when the number of stimuli and responses are the same, producing a standard confusion matrix. In this special case the program outputs an additional data file that conforms to the format for the Speech Information Transmission Analysis (SINFA) [Wang, Behav. Res. Methods Instrum. 8, 471–472 (1976)]. A categorization experiment can be run by specifying many stimuli that span a continuum, allowing only two responses. The program requires a PC or AT compatible computer, a Koala Pad connected to a game port, and a Data Translation DT2801-A D/A output board. [Work supported by NIH.]
Software for analyzing and plotting scientific data
Authors: Jesteadt Walt , Neely Stephen T. , Callaghan Brian P. , Ferrel Roger L.
Published: October 2005

Our PLT program, in use in a number of laboratories, translates a high-level data plotting command language into Tektronix or HP-compatible output. It is specifically designed for scientific applications, allowing maximum flexibility in the use of partial log axes, labels, and error bars, with rapid output on a graphics terminal or publication quality hardcopy on a pen plotter or laser printer. GREG, a spreadsheet data analysis program developed in parallel with PLT, allows data to be represented in terms of a five-dimensional factorial design. It provides convenient data entry from either keyboard or digitizer pad, automatic handling of missing values, data transformations, low-level statistical analyses, and flexible generation of tables and plots. Plots are generated by producing PLT input files. GREG can call user supplied programs to fit specific models or perform other functions, so the data analysis framework can be easily expanded. Both PLT and GREG run under RT-11, TSX, UNIX, and MS-DOS; they will be demonstrated on a portable AT clone. [Work supported by NIH.]
Application of digital signal processing techniques to research on sound localization
Authors: Wightman Frederic , Kistler Doris
Published: October 2005

The availability of fast, general-purpose laboratory computers has dramatically changed the way research is done in many areas of psychoacoustics. In studies of sound localization, for example, standard digital signal-processing techniques are used to synthesize stimuli for headphone presentation that contain all of the localization cues (e.g., interaural differences, pinna effects) that are available from sounds presented in free field. This brings to localization research a much needed degree of stimulus control and specificity, and allows us to address certain basic issues that were heretofore inaccessible. This presentation will survey the general principles involved in the application of digital signal processing to localization research, and will discuss the practical limitations of the specific techniques that are used (e.g., FFTs, FIR filters, inverse filtering). Empirical data will be shown in order to address questions relating to the classic tradeoffs among signal/noise ratio, bandwidth, word-length, and sampling frequency. The feasibility of real-time digital processing in currently available PC-based systems will also be discussed. [Work supported by NIH, NSF, and NASA.]
Classical respiratory conditioning used in auditory psychophysics of the goldfish: Methods and results in detection and discrimination studies
Authors: Fay R. , Coombs S. , Wheeles C.
Published: October 2005

Classical conditioning of respiration has been used to study many aspects of hearing in the goldfish, including absolute and masked thresholds, frequency, intensity, and time discrimination thresholds. A restrained fish is presented with a 7-s auditory signal that ends with a brief electric shock across the body. Shock causes an unconditioned suppression of respiration lasting several seconds. Several pairings of the auditory signal with the shock results in respiratory suppression during the signal. Initial conditioning is rapid (conditioned responses appear within the first 10 to 20 trials), several thresholds can be obtained in one day, and thresholds can be obtained for individual animals for several years. Critical factors for success with this method include the measurement of respiration (using a thermistor), method of animal restraint, levels of electric shock, intertrial intervals, false alarm estimation, overall respiratory rate, subject selection, and water conditioning. Details of the methods and procedures that have been found useful in conditioning and threshold definition will be given along with illustrative data on threshold values and stability in masking and intensity discrimination experiments.
Psychoacoustical measurements with delays in neonates' vocalizations
Authors: Gray Lincoln
Published: October 2005

Newborn chickens momentarily delay their usually incessant peeping when they hear either an onset or change in an auditory stimulus. This unconditioned response provides several ways to study the early development of hearing. (1) Receiver operating characteristics, drawn from pooled histograms of responses on stimulus and control trials, strongly suggest that this response is a measure of auditory detection. Areas under these curves can be used to quantify the auditory abilities of newborn subjects. (2) Adaptive procedures based on paired comparisons of stimulus and control trials provide rapid estimates of thresholds and difference limens. A five-frequency audiogram, for example, can be obtained from a newborn chick in several minutes. (3) Responses to all possible pairs of transitions between multiple stimuli can be used as proximity data for multidimensional scaling algorithms. This provides a “map” of how auditory perceptions change immediately after birth. (4) Neonates' responses to naturalistic stimuli may be different than those to more arbitrary puretone and noise signals. Delayed vocalizations also occur in other animals, allowing comparative studies of auditory development. [Work supported by NIH.]
The use of multidimensional scaling techniques for revealing perceptual categories for complex stimuli in animals
Authors: Dooling Robert J. , Okanoya Kazuo , Brown Susan D. , Park Thomas J.
Published: October 2005

A combination of operant conditioning and multidimensional scaling techniques for demonstrating natural perceptual categories for complex sounds in small birds are described. Birds are trained using operant conditioning procedures on either a same/different discrimination task or on a task requiring the detection of change in a repeating background. Response latencies are used to construct similarity matrices, and multidimensional scaling procedures are then used to produce spatial maps of complex sounds reflecting perceptual organization. Stimulus similarity is represented by spatial proximity. Stimulus groupings in multidimensional space indicate perceptual categories that can be confirmed by cluster analyses. These procedures have been used to study the perception of complex sounds such as bird calls and speech in small birds, but these techniques should also be useful in examining the perception of complex sounds in other animals. [Work supported by NIH.]
Small-sample statistical analysis of Levitt's psychophysical procedure
Authors: Edwards Brent W. , Wakefield Gregory H.
Published: October 2005

Levitt's 2IFC adaptive procedure is a widely used psychophysical method for estimating detection or discrimination threshold. The commonly accepted analysis shows that the average of the levels at which reversals occur approaches the 0.707 point on the psychometric function when the levels are updated according to the “2-up, 1-down” rule. This analysis is based on a continuous representation of the psychometric function and on the asymptotic behavior of the tracks. The more practical case was considered in which the psychometric function is represented by a set of discrete levels and in which the stopping criterion is a fixed number of reversals. A Markov model for the reversal levels is proposed as a method for determining the statistics of the estimator. Based on this approach, the estimator is significantly biased for small numbers of reversals, and decreases monotonically as this number becomes large. Convergence to the 0.707 point, however, is not guaranteed and depends on the sampling of the psychometric function. These results hold, in general, regardless of the form of the psychometric function or the method by which the psychometric function is sampled. More rapid convergence with less bias may be achieved by tailoring the sampling to the psychometric function. Several approaches to the design of such sampling will be discussed. [Research supported by AFOSR.]
A new set of auditory demonstrations
Authors: Houtsma A. J. M. , Rossing T. D. , Wagenaars W. M.
Published: October 2005

A new series of auditory demonstrations for classroom use, modeled after the well-known “Harvard Tapes” [D. M. Green, Harvard Laboratory of Psychophysics (1978)] was developed and produced. The demonstrations are issued on a compact disc (CD), are accompanied by a 92-page explanatory booklet, and fit conveniently in a handy slide box. Almost all demonstrations were synthesized digitally on DEC Vax 11/780 and MicroVax II computers, converted to analog signals with DSC-200 16-bit two-channel D/A converters, and recorded on 16-bit PCM video tape. Brief spoken introductions to each demonstration were recorded on similar PCM video tape. Editing of the master tape was done digitally in 1630 format on 34-in. U-Matic tape, from which the CD master was made. The CD medium, allowing a S/N ratio of up to 90 dB, is very appropriate for sound demonstrations that require a clean acoustic background. The medium also provides random access to individual demonstrations or parts thereof, allows preprogramming of any desired combination of demonstrations, and is much more resistant against wear and tear than traditional tapes or records. The CD can be played on any CD player capable of handling selections from up to 80 tracks. The demo set is available through the ASA.
An auditory filterbank design and its hardware implementation with DSP
Authors: Komakine Takashi , Hirahara Tatsuya
Published: October 2005

As the first step in constructing a signal processor to simulate the human auditory function, a computational filterbank with cochlear frequency analysis function was designed [T. Komakine and T. Hirahara, Tech. Rep. IEICE Jpn. SP87-45, 65–72 (1987)(in Japanese)] and its hardware with DSP implementation completed. From the various cochlear filtering models proposed to date, a cascade/parallel type filterbank model [e.g., R. F. Lyon, ICASSP82, 1282–1285 (1982)] was chosen followed by design of a filterbank composed of 61-channel IIR filter blocks. Each filter block was spaced at 13 Bark intervals from 1 to 21 Bark. Parameters for each filter were decided by taking into account observed physiological data and filter stability. The parameter optimizing effects of the filter were discussed. Sound spectrograms produced by using the filterbank showed that this auditory filterbank model allows excellent frequency and time resolution. A hardware implementation of the filterbank, composed of floating-point DSP chips and a host controller, was studied. In addition to the computational filtering functions, some control functions, such as automatic addressing previously done in the host controller, are distributed to each DSP. Therefore, control of each DSP unit by the host is easier. More sophisticated functions are easily added. This is in part due to a program memory commonly accessible both from each DSP as well as the host. Finally, the auditory filterbank is capable of a frequency analysis of up to 3000 channels within 5 ms, which is equal to the time required for the wave to travel the basilar membrane. It is also expandable to allow hardware implementation with more than 30 000 cascaded DSP units.
Whistling with heavy gases or at elevated pressures
Authors: Mackay R. Stuart
Published: October 2005

Most divers can speak easily but cannot whistle at elevated pressure. Suggestively, sperm whales are not known to whistle. However, rigid flutes or bird calls remain easy to blow “at depth”; frequency changes little. The author can just whistle tunes at 4 atm (equivalent water depth of 30 m). At one atmosphere he could whistle with a breath of half air and half sulfur hexafluoride but not with pure SF6 (density relative to air of 5.1); alternate whistling in and out aids the observation. Possibly this is due to increasing gas density yielding damping vibrations in the soft tissues shedding vortices. Caution: Limiting duration prevents suffocation; possible impurities in SF6 are toxic and corrode metal. The SF6 lowers whisper and speech tones but not hum and whistle frequencies; reflex changes in the throat can confuse the latter. A colleague, Jon Pegg, can whistle tunes normally as well as through his nose with mouth closed and vocal cords constricted to a slit. His nose whistle quenches at 20 m and the normal one at 10 m, suggesting sites of different stiffness. It is said to be impossible for organ pipes to sound if the stream is denser than the surroundings. Several blew well (frequency lowered) with octafluoropropane (density 6.5) in air.
Digital synthesis of binaural auditory localization azimuth cues using headphones
Authors: McKinley Richard L. , Ericson Mark A.
Published: October 2005

A laboratory demonstration prototype of a digital auditory localization cue synthesizer has been developed. This synthesizer uses a single audio input that is separately processed in real-time for independent presentation to each ear using headphones. The headphone presented acoustic signals are easy to localize and appear to be out of head. The acoustic image is stabilized for head movement by use of a three-space head tracking device. The paper will describe the salient parameters of the design. A description of the psychoacoustic and electroacoustic measurements that led to the design will be presented. Human performance data on free field, simulated, and synthesized localization cues will be described and a real-time interactive demonstration will be available for interested listeners.
Fiber optic flexural disk microphone
Authors: Hofler T. , Brown D. A. , Garrett S. L.
Published: October 2005

An interferometric fiber optic microphone consisting of a 10-m-long, 4-cm-diam, flat-wound (spiral) single mode optical fiber bonded to a simply supported, 8-cm-diam, 3-mm-thick aluminum disk will be described. In the presence of a pressure difference across the plate or an acceleration of the plate, the surface strain induced in the plate is transmitted to the optical fiber. The optical phase shift induced by the strain is detected in an all-fiber Michelson interferometer. The calculated strain [A. E. H. Love, A Treatise on the Mathematical Theory of Elasticity (Dover, New York, 1944), 4th ed., Sec. ♯309] and optical strain measured using static pressure, acoustic pressure, and a calibrated aecelerometer, are in good agreement (± 10%) and yield a sensitivity of 30 milliradians per Pascal per meter of optical fiber below the plate resonance frequency of 3 kHz. [Work supported by the Office of Naval Research and Office of Naval Technology.]
A magnetohydrodynamically driven Helmholtz resonant projector
Authors: Haney C. E. D. , Newmaster J. T. , Garrett S. L.
Published: October 2005

Unlike most conventional transducers that are noncompliant, a magnetohydrodynamic (MHD) transducer is ultracompliant since the volumetric velocity it generates is entirely determined by the acoustical impedance of the load it is driving [G. W. Swift and S. L. Garrett, “Resonance reciprocity calibration of an ultracompliant transducer,” J. Acoust. Soc. Am. 81, 1619 (1987)]. To exploit this property, a magnetohydrodynamically driven Helmholtz resonator has been designed and calibrated that consists of a 400-ml-PVC tube with a 6-cm-long Lucite neck of rectangular cross section that is placed in the 1.0-cm gap of a 0.25-Tesla permanent magnet and filled with salt water. The transducer was calibrated using a piezoelectric hydrophone within the resonator volume and by direct observation of the motion of the fluid in the neck using an MTI 1000-Fotonic (fiber optic) sensor. Operating at a resonance frequency of about 100 Hz with Q = 16, the transducer has a source strength of 5 cc/Amp-s. [Work supported by the Office of Naval Research.]
An optimal class of generalized time-frequency representations for nonstationary signal analysis
Authors: Zhao Yunxin , Atlas Les E. , Marks Robert J.
Published: October 2005

In this paper, a class of generalized time-frequency representations (GTFR) that have both good time and good frequency characteristics for nonstationary signal analysis is presented. The basis of the approach used is that a time-frequency representation can, in some cases, be improved by allowing negative value of the spectrum. The representation (and its associated kernel) can then be optimized to enhance the peaks in frequency while still maintaining finite time support. The finite time support property has the significance of preserving onset times of signals and also providing clear representations of fast-changing spectral peaks. Experiments were performed on simulated data and real speech for comparison of the GTFR with the spectrogram and the pseudo-Wigner distribution. The results show distinct advantages of the GTFR. For example, onsets of transitions are clearer and, in the case of speech, close formant peaks are easier to distinguish. [Research supported by NSF and Boeing.]
Transfer functions of piezoelectric transducers for underwater applications and their inverse problems
Authors: Ren Shuchu
Published: October 2005

Three related problems of the transfer functions of piezoelectric transducers are solved: (1) the theoretical calculation of the transfer function, (2) the reconstruction of the phase spectrum of the transfer function from the amplitude spectrum, and (3) the extrapolation of the transfer function to the full frequency range. The transfer functions of cylindrical and spherical piezoelectric transducers are obtained by solving the scattering problem of the transducer as a sound receiver. The calculations show that these transducers are typical minimum phase systems; consequently, the phase spectra of their transfer functions can be reconstructed from the amplitude spectra using the Hilbert transform. For an arbitrary piezoelectric transducer it is impossible to determine whether it is a minimum phase system. To solve this problem, an entirely different technique to reconstruct the phase spectrum is also developed. The phase spectrum of the transfer function of a F42-c spherical piezoelectric transducer is reconstructed from the measured values of its amplitude spectrum. The results are consistent with the measured values of its amplitude spectrum. The extrapolation of the transfer function of a piezoelectric transducer from the finite frequency range to the full range is done using the representation theorem of Hp functions. For a spherical piezoelectric transducer, the results obtained by extrapolation are in good agreement with the theoretical ones.
Characterization of length expander magnetostrictive rare earth-iron rods under normal use conditions in transducers
Authors: Claeyssen F. , Boucher D. , Faure S.
Published: October 2005

Magnetostrictive rare earth-iron rods are presently mainly used as active elements of emission transducers. In this case, rods are mechanically prestressed and magnetically polarized. They are excited at high level around the transducer resonance frequency, which is much lower than the rod natural frequency. In order to characterize the rods under these conditions, a two-rod device has been designed, on which the electrical impedance has been measured before being processed by a computer program. The results are the longitudinal piezomagnetic constants of the alloy. The program is based on a plane-wave model; its assumptions on eddy currents and leakage flux are discussed with respect to results provided by a finite element computation. Coupling coefficient k33, Young's modulus, permeability, and loss terms have been reckoned using this method for alloys of various origins and are presented. The characterization of materials started with L. Néel's laboratory alloy [J. Acoust. Soc. Am. Suppl. 1 81, S89 (1987)] and is carried on with grain-oriented alloys.
Composite structure vibration analysis using the finite element method
Authors: Hamonic B. , Debus J. C. , Tourneur H. , Boucher D.
Published: October 2005

The need of low-frequency sonar transducers led to the concept of flextensional structure and, currently, promotes the use of composite materials. To design these transducers, a finite element modeling is very useful, because this method can accurately handle the various constituting parts as well as the radiation damping. The finite element code ATILA [J. N. Decarpigny et al., J. Acoust. Soc. Am. 78, 1499 (1985)] has been used to analyze various types of metallic shell flextensional transducers and, more recently, a special three-dimensional composite finite element has been developed to take composite shells into account. This element is composed of unidirectionally reinforced lamina and various fiber orientations and materials can be used in the same element. The stiffness finite element matrix is calculated using equivalent elastic constants [S. W. Tsai, “Composite Design,” in Think Composites (Dayton, Ohio, 1985)] and a separate numerical integration in three directions for each lamina. This paper presents the element formulation and its tests. Comparison between numerical and experimental results for simple structures has allowed the determination of the physical model parameters. Then, the modeling of a flextensional shell has demonstrated the accuracy of this approach.
Complex environmental exposures and hearing functions
Authors: Manninen Olavi
Published: October 2005

This study deals with changes in TTS2 values, cardiovascular functions, haemodynamic activity, upright body sway, ratios of urinary catecholamines, and correlations between these changes in complex exposure situations. The study was based on a factorial experimental design with a total of 12 exposure combinations. Each individual experiment took 6 h with a pause of 1 h at noon. The subjects (n = 60) were exposed to noise and whole-body vibration at two different dry bulb temperatures. The changes were dependent on the combinations of noise, vibration, and temperature to which the subjects were exposed. The TTS2 values at 4 kHz were associated with the haemodynamic index (HDI) values when the subjects were exposed simultaneously to noise and stochastic vibration at 35°C. The TTS2 values at 6 kHz were associated most strongly with the HDI values after exposure to a combination of noise, stochastic, or sinusoidal vibration, and a temperature of 20°C. The TTS2 values at 4 and 6 kHz correlated positively with the noradrenaline/adrenaline (NA/A) ratio when subjects were exposed to noise at 35°C. The association between the TTS2 values and the 10A/NA ratio and especially the A/NA ratio was very strong when subjects had been simultaneously exposed to noise and sinusoidal or stochastic vibration at 35°C. Furthermore, the highest positive correlation coefficients were found between the TTS2 values at 4 kHz and the upright body sway values in the X direction when the subjects had been exposed to noise and sinusoidal vibration at 20°C.
The effects of noise and vibration on a complex task
Authors: Sandover J. , Porter C. S.
Published: October 2005

Investigations of the interactive effects of noise and whole-body vibration on task performance have been dogged by such problems as the direct effects of vibration on control and the resistance of simple tasks to the environmental stress. The paper describes an attempt to overcome some of these problems and at the same time to consider conditions of practical relevance. Subjects were exposed to heat, noise, and vibration singly and in combination. The environmental magnitudes were typical of some occupations and exposure duration was approximately 6 h. Subjects were asked to perform a visual vigilance and decision making task, a visual monitoring and decoding task, and a battery of simple tasks. The former tasks were designed to place a significant cognitive load on the subject and strategy was emphasized by performance of both tasks at the same time. The tasks had relevance to a real work situation that the subjects were used to. Subjective and physiological measurements were also taken. The paper will present the results of the series of experiments just completed.
Building noise criterion curves, BNC, for interior spaces
Authors: Beranek Leo L.
Published: October 2005

This paper presents an updated set of noise criterion (NC) curves that are renamed building noise criterion (BNC) curves. They are based, in part, on the latest definition of four-band speech interference level and on “spectrum balance,” that is to say, the premise that the loudnesses of all bands containing the same number of critical bands should be equal, as calculated by Stevens Mk. VII perceived loudness method. The curves are extended downward in frequency to include the two octave bands with mean frequencies at 16 and 31.5 Hz. Finally, the high-sound level, low-frequency region of the curves [Blazier, Noise Control Eng. 16, 64–73 (1981)] are demarked to show where human annoyance will probably result from vibrations caused by such noise levels in contemporary building construction. The paper details the use of the BNC curves in writing a specification for building construction and in determination of the conformance of the measured result to the specification, or in rating an existing noise. Particularly important are the procedures given for determining “spectrum imbalance.” The handling of two typical types of imbalance are discussed: (1) an acceptable speech interference level accompanied by high low-frequency band levels; (2) acceptable low-frequency band levels accompanied by a very low speech interference level.
Determination of natural frequency of bone: Study of bone abnormalities
Authors: Yadav Sanjay , Singh V. R.
Published: October 2005

Direct and indirect methods have been used in the past [L. Nokes, W. J. Czyz, Mintow, I. Mackie, J. A. Fairclough, and J. Williams, J. Bio. Med. Eng. 6, 45–48 (1984)] for the determination of natural frequency of bone. Most of them are cumbersome and costly. A simple and quick experimental technique based on the stress wave propagation through bone is developed. An acoustic vibrator as a source of stress wave is used to send the stress wave through bone specimen under test, in this case, an in vitro sample, and a microphone pickup is used to receive the acoustic signal at the other end of the bone, which, in turn, gives an electrical output to be measured on a calibrated electrical detecting system. The natural frequency of bone is determined for an optimum output with respect to different stress wave frequencies. A comparative study of normal and fractured bone is made to diagnose the size of the fracture and its rate of healing.
The transient response of fluid-loaded structures using time-domain methods
Authors: Stephanishen Peter R.
Published: October 2005

Over the past several years, time-domain methods have been developed to address the transient radiation and scattering from structures in fluids with and without a mean flow. These methods are based on using in vacuo modal expansions with time-dependent coefficients that are coupled due to the fluid. To date these methods have been used to investigate the transient response of baffled membranes, plates, and cylindrical shells. The development of the basic computational method will be reviewed and compared to alternative methods, e.g., the doubly asymptotic approximation. Numerical results will then be presented to illustrate typical transient phenomena of interest. A particular example of interest is the destabilizing effect of a mean flow over a vibrating surface. [Work supported by the Office of Naval Research.]
Control of sound radiation from submerged plates
Authors: Meirovitch Leonard , Thangjitham Surot
Published: October 2005

The subject of sound radiation from submerged elastic bodies, and in particular flat plates, is our current research interest in the field of underwater acoustics. Because the plate is in contact with the surrounding fluid, it gives rise to a general class of coupled problems involving “radiation loading.” This radiation loading modifies the motion of the submerged plate via the excess acoustic pressure, which, in turn, depends on the velocity of the plate. As a result, an interaction between the fluid and the plate exists in the form of “feedback coupling” [M. C. Junger and D. Feit, Sound, Structures and Their Interaction (MIT, Cambridge, MA, 1986), 2nd ed.]. In many applications, the radiation pressure is an undesirable effect, so that the object is to suppress it. This paper is concerned with radiation pressure suppression through active control. The general idea of active control of a structure is to suppress the vibration by applying suitable feedback control forces [L. Meirovitch, Dynamics and Control of Structures (Wiley, New York, 1989)], which requires sensors capable of measuring the state of the system and actuators capable of applying control forces. A control technique capable of suppressing the farfield radiation pressure is developed.
Vibrational response of water-loaded flat plate to a turbulent boundary layer pressure field
Authors: Narayan Nilabh , Plunkett Robert
Published: October 2005

The vibrational response of a 1.2- × 40- × 200-mm flush-mounted flat plate, with heavy accelerometers attached to the backside, to a turbulent boundary layer (TBL) pressure field was measured at several flow velocities in a 190- × 190-mm low noise water tunnel. The measured response spectra of the water-loaded stainless steel flat plate was compared with the calculated response in the 400- to 1500-Hz frequency range using the plate modal properties and the pressure spectra. The phase indicated convection velocity varied with separation and frequency and was accounted for in the plate response calculations. The plate modal properties were obtained experimentally by in situ modal analysis and the plate was found to have a large number of heavily damped modes in the same frequency range. The turbulent boundary layer pressure field driving the plate was measured upstream and downstream from the plate and was found to be unaffected by the vibrating plate. The two-point pressure cross spectra at the wall for the TBL were found to vary with the axial and transverse separations in a way similar to that of the Corcos similarity model and followed the multiplication hypothesis but had a different form of explicit dependence upon the separation.
Noise radiation from transverse air flow over cylinders of constant and varying diameter: An empirical predictive model
Authors: Donavan Paul R. , Herdman Karen C.
Published: October 2005

A simplified model to estimate the amplitude and frequency content of the noise radiated by constant and varying (tapered) diameter cylinders has been developed. Using the measured intensity level of Aeolian tones for several constant diameter cylinders, the incremental source strength along the cylinder has been determined as a function of Reynolds numbers. After choosing an element size consistent with the correlation length of the vortex shedding, tapered cylinders are modeled as a series of small, discrete cylinders each of constant diameter. The total sound produced by the tapered cylinder is then determined by summing the contribution of each segment over the cylinder length, properly accounting for geometry using the assumption that the sources add incoherently. The results predicted by this approach have been found to be quantitatively consistent with the data measured for tapered cylinders at several orientations. The model can also be readily extended to examine the effect of nonuniform flow on radiated noise.
Applications of reactive intensity
Authors: Mann J. Adin , Tichy Jiri
Published: October 2005

The reactive intensity vector is a very useful analysis tool for the acoustic nearfield. Vibration sources, active and passive absorbers, and diffraction can be differentiated by measuring the reactive intensity. The real and imaginary parts of the specific acoustic impedance should also be considered. Nearfield measurements will be discussed where the reactive intensity vector is used to determine the location and distinguish between piston sources and a resonator that are acoustically coupled. The reactive intensity must be used because the time-averaged intensity vector does not distinguish between a resonator absorbing power and an active source absorbing power. Diffraction will also be identified. These results will further show that all quantities measured with the acoustic intensity technique should be analyzed.
Voice onset time and F perturbation under conditions of mixed airstream mechanism
Authors: MacKay Ian R. A. , Zwierzynski Dariusz A.
Published: October 2005

Pulmonic and velaric airstream mechanisms are employed together in the production of clicks and accompanying vowels. Questions thus emerge as to the temporal and microprosodic coordination of these two mechanisms, particularly with respect to the phonatory characteristics of clicks. This report is on the acoustic measurement of the post-alveolar click in Sesotho, with particular reference to VOT and F0 perturbations. There are two phonemically distinct varieties of click in Sesotho, which might be characterized as aspirated and unaspirated in accordance with orthographic conventions. The unaspirated click has a mean VOT of approximately 40 ms on the order of voiceless aspirated plosives in English. The aspirated click has a mean VOT of approximately 120 ms, greater than any plosive type reported by Lisker and Abramson [Language 47, 767–785 (1971)]. The F0 values were measured pre- and post-click in both word-initial and word-medial clicks. Our data indicate a consistent fall in F0 immediately prior to a click, regardless of lexical tone, presence of word boundary, and the distinctive prosody of interrogatives. This perturbation of fundamental frequency may be indicative of an intrinsic property of click production. By contrast, measures of post-click F0 reveal a less regular pattern of microchange both across the silent interval and during the initial glottal cycles at post-click vowel onset; these phenomena are apparently influenced by lexical tone.
An acoustic study of tone sandhi in Taiwanese
Authors: Lin Hwei-Bing
Published: October 2005

In Taiwanese, there are five long tones: high level, high falling, mid falling, low rising, and mid level. These tones undergo sandhi changes when they do not precede a major syntatic boundary: mid level→mid falling→high falling→high level→mid level←low rising. The aim of this study was to validate these phonological observations by investigating the acoustic properties of tones in sequence. Another issue of interest was whether tonal coarticulation occurs in Taiwanese. Six native speakers produced several repetitions of a sequence of two syllables /si/ and /do/ in sentence-final position with all combinations of tones. These utterances were analyzed to determine the average fundamental frequency (F0) contours of the five tones on each syllable. The tonal contours on /do/ syllables were highly similar to those observed previously in isolated syllables. Some slight perserveratory effects of the preceding /si/ tones on the F0 onset of /do/ tones were found. The tones on the penultimate syllable /si/ underwent sandhi changes, as expected. No coarticulatory effects were found on /si/ tones. However, the sandhi tones differed in their F0contours from those on the final /do/ syllable. [Work supported by NICHD.]
Perceptual switching in Spanish/English bilinguals
Authors: Bohn O. S. , Flege J. E.
Published: October 2005

Elman et al. [J. Acoust. Soc. Am. 62, 971–974 (1971)] found that, whereas English monolinguals heard short-lag stops as /b/, Spanish monolinguals heard them as /p/. Spanish/English bilinguals who were proficient in English gave more “English” (i.e., /b/) responses in an English than Spanish perceptual “set.” This study further tested the hypothesis that bilinguals may use two criteria in identifying short-lag stops. In experiment 1, /to/ tokens spoken by Spanish monolinguals with short-lag VOT were presented along with English long-lag /to/ or English short-lag /do/ tokens. English monolinguals and Spanish subjects who learned English as adults labeled the Spanish /to/ stimuli as /to/ (rather than /do/) more often in the context of English /do/ (73%) than /to/ (50%). Since both groups showed the same pattern, this likely reflected a context effect rather than a criterion shift. In experiment 2, the English monolinguals did not label the Spanish /to/ stimuli as /to/ more often in a Spanish than English “set.” Whereas the bilinguals who participated in the Spanish set first showed a 12% shift, those who participated in the English set first showed no evidence of a switch in criteria. It has been hypothesized [Flege and Eefting, J. Acoust. Soc. Am. 83, 729–740 (1988)] that Spanish speakers who learn English in early childhood establish separate categories for Spanish and English /t/. Work is underway to determine if such subjects will show a larger language set effect than the adult L2 learners already examined. [Work supported by NIH.]
Voicing in French and English labial stops: A cross-language perceptual study
Authors: Munro Murray J. , Nearey Terrance M.
Published: October 2005

Measurements of European French and Canadian English labial stops indicate that the French /p/ and English /b/ categories are very similar, often exhibiting only small differences in voice onset time. Two groups of English-speaking listeners (one with some knowledge of French, the other without) were asked to categorize a set of modified natural tokens from these categories as either /p/ or /b/. The tokens were taken from word-initial stops produced in a sentence context. Components preceding the release bursts were removed, and the signals were truncated at 68 ms. Despite the small difference in VOT between the categories, most listeners were able to reliably separate them at levels above chance. Analysis revealed that the listeners may have relied primarily on VOT in making their judgments and that overall amplitude [C. J. Darwin and M. Pearson, Speech Commun. 1, 29–44 (1982)] may have played a secondary role. These findings indicate that listeners may be sensitive to small differences between categories in their native language and analogous categories in a foreign language.
Differences in relative temporal positioning between initial consonant and vowel in syllables: Evidence from Chinese
Authors: Ren Hongmo
Published: October 2005

Junctures and coarticulations between syllable initial consonant and vowel have usually been described in terms of transitions and target variations. This study proposes that we also need to specify differences in relative temporal positioning between these two major syllabic constituents. The C + /iau/ syllables in Chinese were examined. The /iau/ is a triphthong rhyme. Because rhyme internal formant pattern remains constant in different syllables, the turning point in formant trajectories can be used as reference for determining the temporal position of the rhyme as a whole. In fact, we have observed the whole rhyme shifting along the time axis relative to the initial consonant boundary. Data from six speakers shows that syllable initial consonant and rhyme exhibit different degrees of overlapping and “repelling,” with a distance ranging from − 15 to 83 ms. The relation between these findings and the P center or stress beat measured in the same tokens is also discussed. [Work supported by NSF.]
Pitch neutralization in Japanese one mora homonyms
Authors: Mogami Yuko
Published: October 2005

Japanese one mora homonyms in phrases are distinguished by pitch. The present experiment was designed to discover if this distinction between the homonyms was also retained in an isolated environment; i.e., if speakers distinctively pronounced the homonyms in isolation and if listeners could correctly identify them. Following Fourakis and Iverson (1984), the influence of the speakers' attention to the purpose of the experiment was considered in the analysis of the experiment. Four pairs of homonyms, of presumably equal frequency in use, were recorded in two ways: first, subjects unaware of the homonyms orally responded to questions whose answers contained the homonyms as well as other decoy words; second, subjects orally read a list of the same homonyms. The obtained homonyms were compared with respect to F0 value, duration, and pitch contour and then used for listeners' identification task. Both the acoustical analysis and the results of the listeners' identification task proved that one mora homonyms were completely neutralized in an isolated environment regardless of the speakers' attention to the homonyms. [Special thanks to John Ohala, Masayoshi Shibatani, Yoko Hasegawa, and Mariscela Amador.]
Delayed pitch fall in Japanese
Authors: Hasegawa Yoko , Hata Kazue
Published: October 2005

The Tokyo dialect of Japanese is regarded as a prototypical pitch accent language where the location of F0 fall is the only acoustic correlate of the accent. Neustupný [Onsei-gakkai Kaihoo 121 (1966)], however, reported that F0 fall does not always synchronize with an accented mora but may be carried over to the following mora. Sugito [Shooin Joshi Daigaku Ronshuu 10 (1972)] found that native speakers perceived a pitch accent on a mora when it was followed by a falling F0 contour and also that the F0 peak value in the accented mora need not be higher than that in the following mora. Not well investigated though is the correlation between the peak location and the steepness of falling contour. The working hypothesis is that the later the F0 fall occurs, the faster F0 drops. This delayed pitch fall is characterized in terms of (1) the F0 peak location and (2) the steepness of F0 contour computed in Hz/cs. The results from 560 tokens uttered by seven speakers of the Tokyo dialect confirmed our hypothesis. [Work partially supported by Sloan grant to Berkeley Cognitive Science Program.]
Computer generation of singing by manipulation of speech parameters
Authors: Olive Joseph P.
Published: October 2005

This paper describes a computer system for simulating a singing voice by using parameters derived from natural speech. The system was constructed to manipulate parametrized speech to study prosodic and segmental properties of speech. The system enabled a user to manipulate fundamental frequency, amplitude, timing, and spectral characteristics of speech sounds. While manipulations of the parameters of natural speech could easily produce “unreal” or “weird” sounds, simulating a singing voice with qualities approaching those of a natural singing voice was more difficult. The latter required careful manipulations of the fundamental frequency and timing to imitate human singers. Examples of both types of sounds will be demonstrated. In an attempt to imitate natural singing, the simulation of solo and choral singing, as well as contrasting singing styles of “classical” and “pop” singers is described.