bluuebunny/crossref_metadata_2025 huggingface dataset:
Improved laser interferometry for ultrasonic vibration measurements on diffuse surfaces
Authors: Nagy Peter B. , Adler Laszlo
Published: October 2005

In spite of its obvious advantages over conventional contacts and immersion techniques, laser interferometry is not feasible in many acoustic measurements since its sensitivity is insufficient at ultrasonic frequencies. Part of the problem is that the maximum signal-to-noise ratio cannot be maintained on ordinary diffusely reflecting surfaces. Although these surfaces reflect a fair amount (5%–50%) of the incident laser light, this energy is randomly distributed among a large number of bright speckles. Unless the detector happens to see one of these bright speckles, the interferometer’s noise level will be much higher than the optimum. In most ultrasonic measurements, when the vibrations to be detected are periodically excited by a transmitter, this adverse speckle effect can be almost completely eliminated by a very simple procedure. The conventional interferometer is mechanically modulated to assure the random occurrence of a few very bright speckles and to move the whole speckle pattern around at an appropriate speed. Random but frequent bright flashes detected from the surface of the specimen result. The bright periods are sufficiently long, to trigger the ultrasonic pulser and detect the transmitted signals before the flash subsides and close to maximum signal-to-noise ratio can be maintained everywhere on the surface of a diffuse object. Numerous examples are shown to demonstrate the significant improvement of the detection sensitivity by this novel approach.
Acoustic calibrator for plethysmographic surface pressure sensors
Authors: Zuckerwar Allan J.
Published: October 2005

Plethysmographic surface pressure sensors measure heart sounds, breathing sounds, blood pressure, blood flow sounds, fetal movement, and other sounds indicative of physiological conditions. An acoustic calibrator for such sensors should provide a specific acoustic impedance and permit a mounting arrangement resembling in-service conditions. The calibrator described here consists of a water-filled balloon, on which the test sensor is mounted at the equator. A hydrophone positioned in the center of the balloon serves as a nearly isotropic source at low frequencies. Since the excited sound field is not free but reverberant, a free-field calibration of the hydrophone as projector cannot be used. Rather, a rigidly supported reference sensor of known calibration is tightly coupled to the balloon surface to calibrate the amplitude of the local surface pressure (which is sensitive to mounting conditions) as a function of frequency. The calibration of an acoustically based fetal heart tone monitor will be presented as an example.
How much random data is enough?
Authors: Burgess John C.
Published: October 2005

There is a relationship between the amount of data taken from a random signal and its statistical reliability. The usual assumptions are that the signal is stationary, white, and Gaussian. While procedures for analysis of a finite amount of random data after acquisition are well known, little information appears to be available about planning before data acquisition. With the usual assumptions, an upper limit can be found for the amount of random data required to meet a specified statistical reliability. This can be done for both time and frequency domain applications.
A universal shading method—successive approximation method
Authors: Lan J. , Simoneau M. J. , Jeffers R. K. , Boucher S. G.
Published: October 2005

A completely new shading method, called the successive approximation method, is developed. The foundation of the method is the energy extreme theorem with a constraint condition: Construct a function L, which is the integration of the square of the expression of sound pressure multiplied by a control function, letting L reach minimum will give a set of unknown numbers, therefore a sound pressure can be obtained for the specified control function, which is based on the expected function and the sound pressure obtained during the last iteration. After several iterations, the sound pressure will approach expected function. This method can be applied to any arrays, both in the far field and near field. A variety of patterns can be obtained by either amplitude shading, phase shading, or using unequally spaced arrays. Changing the constraint condition can alter the fitting way between the pattern to be shaded and the expected one. Several computed and experimental examples will be given, which includes: unequally spaced arrays; superdirectivity arrays; using unsymmetric amplitude shading to obtain fan-shaped patterns with any low sidelobe level; using amplitude shading only to steer mainbeam.
Identification of a single modulated component in a complex sound
Authors: Moore Brian C. J. , Bacon Sid P.
Published: October 2005

In experiment 1, subjects were required to decide whether the frequency of a single-modulated carrier (with a suprathreshold modulation depth) in a complex sound was the same as or different from the frequency of a probe composed of a single-modulated carrier. They were able to perform well above chance, i.e., they could identify the modulated carrier. Performance was not greatly affected by whether the probe was presented before or after the complex or by the modulation depth used. Experiment 2 was similar to experiment 1, except that the level of each carrier in the complex was varied randomly from trial to trial. This produced only a very slight impairment of performance, indicating that short-term across-frequency differences in level were not used to identify the modulated component in experiment 1. In both experiments 1 and 2, the identification of a 1000-Hz carrier was best when that carrier was part of a six-carrier harmonic complex, less good when it was part of a six-carrier inharmonic complex, and worst when it was part of a two-carrier complex. The results are interpreted in terms of perceptual grouping. [Work supported by MRC, UK, and NIH.]
Across-channel processing of amplitude modulation in cochlear hearing-impaired listeners
Authors: Hall J. W. , Grose J. H.
Published: October 2005

Over the past few years, there have been several studies that have examined across-channel effects related to amplitude modulation (AM). Some of these effects, notably comodulation masking release (CMR), modulation detection interference (MDI), and monaural envelope correlation perception, have been hypothesized to be related to the analysis of signals in complex noise backgrounds, and to auditory grouping. Several studies of cochlear-impaired listeners using these paradigms have recently been completed. Cochlear-impaired listeners often report having difficulty processing signals in noisy backgrounds, and it is possible that the study of across-channel processes involving the coding of AM can shed light upon some components of this disability. Because physiological studies have indicated that neural/central abnormalities can occur secondary to cochlear lesions, it is worthwhile to consider the effect of cochlear hearing loss on across-channel auditory processes. At this point, the extent to which cochlear-based losses in humans are associated with some degree of concomitant neural deficit is largely unknown. One of the specific goals of the research was to determine the peripheral versus central contributions to the hearing loss related deficits in the processing of across-channel AM cues. The results of the studies indicated that relatively mild acquired cochlear losses in adults are associated with effects that can be accounted for in terms of peripheral processing deficits. [Research supported by NIH NIDCD.]
A simple sensor for direct measurement of particle velocity in water
Authors: Gardner David L. , Gabrielson Thomas B. , Garrett Steven L.
Published: October 2005

Normally, acoustic particle velocity is inferred from measurement of pressure or pressure gradient. In water, however, direct measurement of particle velocity is simple. A moving-coil sensor embedded in a neutrally buoyant package produces a voltage output directly proportional to the particle velocity in the surrounding fluid for frequencies above the resonance of the moving coil. Leslie etal. [J. Acoust. Soc. Am. 28, 711–715 (1956)] built such a sensor by mounting a moving-coil element inside a hollow brass sphere. They also presented the theory for its response with the assumption that the sphere was acoustically rigid. The device described in this work is considerably simpler to fabricate—the moving-coil element is cast into a small cylinder of syntactic foam to make the entire sensor neutrally buoyant—and all of the materials are readily available. In addition, the sensor is small and does not perturb the acoustic field as much as a rigid body would. This sensor can be used to measure acoustic particle velocity directly; it can be used in conjunction with a pressure hydrophone to distinguish between real and reactive power flows or to measure acoustic intensity; and, it can be used to demonstrate effectively the spatial variations of pressure and particle velocity in standing waves.
Acoustical holography with arbitrarily shaped sources in a cylindrical duct
Authors: Konzelman Charles J.
Published: October 2005

Radiated sound-pressure measurements of a device designed to operate in the presence of a fluid flow are frequently conducted in wind or water tunnels in order to control flow conditions. However, due to environmental reverberation, these sound-pressure measurements may bear little resemblance to those obtained when the device operates in a free-field environment. In particular, it may be impossible to infer the far-field radiation pattern from such measurements. This study examines the possibility of using acoustical holography to determine the far-field radiation pattern in such cases. Numerical simulations are used to synthesize the pressure field radiated by an arbitrarily shaped source that has a prescribed normal velocity distribution and which operates in a rigid-walled cylindrical duct. Acoustical holography is then used to reconstruct the velocity distribution. Assuming that the source velocity does not change when the device is operated in a free-field environment, the reconstructed source velocity is used to determine the far-field radiation pattern. This far-field radiation pattern is compared with that obtained using the exact source velocity in order to assess the performance of the technique.
A quiet, pleasant sounding environment—A goal for the ASA
Authors: Lyon Richard H.
Published: October 2005

A quiet and tranquil environment may seem an unreachable goal, but this profession has a responsibility to assist in the striving for it. The ASA has the disciplinary strengths in engineering, standards, perceptual science, environmental control, and physiological and psychological health studies that are required for the effort. As an international scientific and engineering organization with strong constituencies in academia, industry, and government, the ASA has the credibility to provide the balanced leadership in education, technical developments, and policy analysis that is needed. The ASA needs to assemble and organize these resources to provide assistance to society at large in this complex area.
Multiple speaker localization using envelope time delays
Authors: Zakarauskas Pierre , Cynader Max S.
Published: October 2005

It is well known that an array of N sensors can discriminate between at most N-1 continuous sources. However, if the time series produced by the sources contain transients, this limit applies only to samples that are shorter than the duration of the transients. This principle has been applied to the problem of localizing two speakers by using only two microphones. Human speech contains a large number of more or less sharp transients. The time of arrival of the transients at each microphone is estimated by finding the local maxima in the envelope of the time series. The time series at each microphone is replaced with the value of the envelope at the local maxima location, and is set to zero everywhere else. The cross correlation of the two resulting time series presents peaks at the time delays corresponding to the correct location of speakers, whereas the cross correlation of the original time series had a single peak at the wrong time delay. It is shown that the difference is due to the nonlinearity of the operations performed on the time series.
Fourier descriptors for time domain labeling of multi-speaker voiced/unvoiced stop consonants
Authors: Green Jack , Pinkowski Ben
Published: October 2005

Fourier descriptors (FD’s), common shape features often used in computer image analysis, were used to describe time domain waveform shapes of speech signals. The signals examined were initial stop consonants contained in continuous speech samples of five speakers in the TIMIT corpus. The stops were extracted from a consonant/vowel (CV) combination, and information contained in the vowel portion of the utterances was not used. Data included with the corpus were used to segment the sounds. Sixty four FD’s were calculated for each of the 58 samples used in the study. These FD’s were used to label each stop as being either voiced or unvoiced. Preliminary results showed 61% of the stops were correctly labeled. Because time domain information requires less computational effort to obtain, a labeling could be obtained while frequency domain information is being calculated. The labeling could then be used to limit the search space in the frequency domain. Despite limitations of working in the time domain, it is possible that these results will approach 75%. Present work is examining the use of neural networks as a method of increasing correct classification.
Applications of a two-way, three-dimensional parabolic equation
Authors: Orris Gregory J. , Collins Michael D.
Published: October 2005

A method was previously reported for solving three-dimensional propagation and backscattering problems involving a range-dependent environment that is separable in one of the horizontal Cartesian coordinates. With this approach, the two-dimensional parabolic equation (PE) is modified by a separation constant that appears in the wave-number term. The self-starter has been modified to generate an accurate starting field for this problem. Large-scale problems can be solved efficiently when this approach is combined with the split-step Padé solution. This two-way PE is one of the few deterministic, three-dimensional reverberation models in existence and may be applied to problems involving a point source, a beamed array of sources, gradual range dependence, and scattering facets.
Three-dimensional modeling of narrow-band pulse spreading using normal modes
Authors: Wild Michael , Joyce Robert
Published: October 2005

Energy from a narrow-band pulse interacting with a time varying ocean surface will be spread into upper and lower sidebands. This can be modeled in a range-independent ocean environment using normal modes and a coupling scheme based on first-order perturbation theory. The structure of the scattered signal from the two-dimensional Pierson–Moskowitz surface wave spectrum is presented. The Kraken normal-mode model with environmental inputs from a shallow sea has been used, and a comparison is made with measured data.
Finite baffle diffraction effects on second-order differential microphones
Authors: Smith Michael A. , Elko Gary W. , West James E. , Kubli R. A.
Published: October 2005

Second-order differential microphones provide an excellent means for obtaining a superdirectional response with frequency-independent directivity and small size. By subtracting the output of two closely spaced first-order differential dipole [cos(θ)] microphones, a linear quadrupole is obtained whose directivity pattern is cos2(θ). This particular directivity pattern is realized only when the microphones are in an acoustic free-field. When differential systems are placed on or near a finite surface, the effects of diffraction and scattering can significantly modify the directional response relative to the free-field conditions. Since most differential microphones in teleconferencing and business environments are placed on a desktop surface, it is important to understand the effects of diffraction on these systems. A second-order differential microphone was constructed that has been tested for directivity fluctuation in the presence of a large reflecting surface. These experimental results match predicted theoretical patterns for diffraction on a hard disk. Further experimentation with varying surface treatment and different reflector geometries have given valuable insight into the effects of diffraction and scattering on differential microphones placed on a finite reflecting surface.
Proper time-frequency distributions for speech
Authors: Atlas Les E. , Pitton James W. , Loughlin Patrick J.
Published: October 2005

The usual representations that are used for speech involve sliding windows across the signal in time and doing Fourier or other analysis on this windowed signal. However, this approach is inconsistent with the notion of a proper distribution. A proper distribution is defined to have the usual properties of a joint probability density function and it has been shown that proper time-frequency distributions exist for speech [Loughlin etal., Proc. ICASSP, V-125–V-128 (1992)]. The most obvious difference of these new distributions is the improved simultaneous resolution in time and frequency. However, this advantage may not be useful for speech recognition and some other advantages could be utilized. For example, the signal-dependent kernel of a proper distribution may be insensitive to changes in room acoustics and could also effectively normalize out differences in vocal tract length. Also, a proper distribution’s representation of a periodically excited, time-varying resonator is quite different than that of a spectrogram’s, and new features with greater pertinence for auditory modeling may be apparent. [Work supported by Boeing and the Washington Technology Center.]
Multi-microphone cross-correlation based processing for robust speech recognition
Authors: Sullivan Thomas M. , Stern Richard M.
Published: October 2005

A new algorithm of signal processing for robust speech recognition using multiple microphones is described. The algorithm, loosely based on human binaural perception, consists of imposing time-aligning delays on the speech signals from each microphone and passing the delayed speech through a bank of bandpass filters and nonlinear rectifiers. The outputs of the nonlinear rectifiers within each frequency band are cross-correlated, providing an estimate of the spectral profile of short-term energy in the speech signal that is resilient to the presence of off-axis noise sources. A cepstral representation of these energy estimates is used as the feature set for automatic speech recognition using the CMU SPHINX system. The multichannel cross-correlation-based algorithm was found to preserve the shape of vowel spectra in additive noise, and it provides better recognition accuracy than is obtained using equivalent single-channel processing with nonclosetalking microphones. The performance of this system was compared to that obtained using delay-and-sum beamforming and conventional adaptive filtering approaches. Finally, some implications of these results for human binaural hearing will be commented on. [Work supported by Motorola and DARPA.]
Time-frequency analyses of coherent frequency fluctuations among spontaneous otoacoustic emissions
Authors: Burns Edward M. , Pitton James W.
Published: October 2005

Previously it has been shown [e.g., Keefe etal., J. Acoust. Soc. Am. Suppl. 1 88, S17 (1990)] that in ears with multiple spontaneous otoacoustic emissions (SOAEs) there are often short-term amplitude fluctuations that are highly correlated (either positively or negatively) among SOAEs widely separated in frequency. In this report it is shown that in many such ears there are also coherent frequency fluctuations among widely spaced SOAEs. However, the demonstration of these small (a few tenths of a percent), short-term frequency fluctuations is difficult using the usual spectrogram-based analysis because of the inability of the spectrogram to accurately resolve frequency-modulated signals. To circumvent this drawback, advanced time-frequency representations (TFRs) have also been employed: e.g., the cone kernel TFR and the Wigner distribution. These time-frequency representations provide additional insights into the nature of the short-term frequency and amplitude fluctuations in SOAEs. [Work supported by NIDCD.]
Comparison of suppression across frequencies
Authors: Thibodeau Linda M. , Fagelson Marc A.
Published: October 2005

Although much research has focused on the temporal, spectral, and intensity relationships between a masker and a suppressor, there has been little attention directed towards relative amounts of suppression in different frequency regions. The purpose of this experiment was to compare the magnitude of suppression at 500 and 2000 Hz in two forward-masking experiments with seven normal-hearing persons. In the first experiment, the threshold for a 10-ms probe was determined as the bandwidth of a 400-ms masker increased from 0.05 to 1.6 times the probe frequency. Suppression was determined by observing a decrease in threshold when the masker bandwidth exceeded the estimated critical band. In the second experiment, the threshold for a 10-ms probe was obtained with a 400-ms tonal masker in two conditions: masker-alone and masker+suppressor. Suppressor frequencies ranged from 1.1 to 1.5 times the probe frequency. Suppression was indicated by a lower threshold in the masker+suppressor condition relative to that in the masker-alone condition. In both experiments, the magnitude of suppression was smaller for the 500-Hz probe relative to the 2000-Hz probe.
Cross-modal enhancement of repetition detection for very long period recycling frozen noise
Authors: Bashford James A. , Brubaker Bradley S. , Warren Richard M.
Published: October 2005

Guttman and Julesz [J. Acoust. Soc. Am. 35, 610 (1963)] reported that recycling of frozen noise segments can be readily detected for periods up to 1 s and can be detected with difficulty for periods of 2 s. These experiments demonstrate that this limit can be exceeded with ease. It was found that a cross-modal cue (e.g., a light flash) presented synchronously with a randomly selected point in a 10-s recycled frozen noise allows listeners to tap accurately at the repetition rate when the recycled noise is subsequently presented without the cue—even following delays of 24 h or more. Two mechanisms for the effect of cueing are suggested: (1) it provides an accurate time base for determining when repetition of salient events will occur; (2) cross-modality attentional effects highlight auditory events concurrent with the cue. Experiments distinguishing between these two mechanisms will be discussed along with other aspects of the effect. [Work supported by NIH.]
An illusion based on dichotic fusion of harmonically related tones
Authors: Huang Wenyi , Hall Michael D. , Pastore Richard E.
Published: October 2005

In the octave illusion, dichotic pairs of octave-related tones are physically alternated across ears, with perception of only a single fused tone shifting both in pitch and from side-to-side in synchrony with the physical alternation. Huang etal. [J. Acoust. Soc. Am. 91, 2374 (A) (1992)] demonstrated that dichotic fusion of octave stimuli is a primary factor in the occurrence of the illusion. Since the work of Helmholtz, it has been known that fusion occurs most often for stimuli in an octave relationship and decreases in frequency for other specified relationships. Therefore it is conjectured that similar illusion patterns should be observed with specific frequency intervals other than octaves. Experiment 1 evaluated the incidence of dichotic fusion for different frequency integer ratio. Experiment 2 replicated parts of experiment 1 and determined the incidence of the illusion for some noninteger frequency ratios. Incidence of the illusion tended to be greater for stimuli that are known to more readily fuse. [Research supported by NSF and AFOSR.]
Buildup of echo suppression: Effects of echo delay and source position
Authors: Yang Xuefeng , Grantham D. Wesley
Published: October 2005

When a subject in free field is presented with a test click and a simulated echo (originating from different positions on the horizontal plane), the degree of suppression of the echo increases with the number of clicks that precedes the test click [Freyman etal., J. Acoust. Soc. Am. 90, 874–884 (1991)]. Two experiments were conducted to investigate aspects of this buildup of echo suppression. Echo suppression for a test click was quantified by measuring echo threshold (i.e., the echo delay for which subjects could just identify the direction of the echo) in various conditions in which the test click was preceded by a click train. In experiment 1, echo threshold was largest (the maximum buildup occurred) when the echo delay in the preceding click train was 10–20 ms. Experiment 2 investigated the effect of the spatial positions of the preceding train and its echo. For subjects who showed echo suppression buildup, it was maximal in those cases when the train’s echo was in the same position as the test click’s echo. Results will be discussed in terms of the spatial and temporal characteristics underlying the buildup of echo suppression. [Work supported by NIH.]
Shifting auditory attention through space
Authors: Mondor Todd A. , Zatorre Robert J.
Published: October 2005

Auditory spatial attention was investigated by measuring the relation between azimuthal distance and the time necessary to shift attention. On each trial, listeners determined whether a pure or a complex tone had been presented. Either of these target tones could sound at any of five free-field locations arranged in a semi-circle in front of the listener. This arrangement required that subjects respond to targets presented up to 180 deg from an initial attentional focus. Prior to the target tone, an attentional cue, consisting of a brief noise burst, was presented to the same position as the target. The time interval between the onset of this cue and the onset of the target tone was varied from 50 to 1400 ms to control the amount of time available to orient attention to the cued location. Results indicated that response time decreased as the opportunity to shift attention to the cued position increased. Thus, the cue was successful in effecting an attentional shift. However, performance was unaffected by the spatial extent of the attentional shift. Under these experimental conditions, therefore, the time required to move auditory spatial attention from one location to another appears to be independent of distance.
Localization and recall of the spatial position of sounds
Authors: Loy Ingelise J. , Zatorre Robert J.
Published: October 2005

Memory for the location of auditory stimuli was tested after delays of 1 min, 30 min, or 24 h. Eleven environmental sounds were presented to 22 subjects in each of 11 different free-field positions, spaced evenly in a semicircle around the subject in the horizontal plane. Initial localization acuity was recorded upon presentation of each stimulus. To test memory for location, target stimuli were presented over headphones, and subjects attempted to recall the original position. Memory for location was significantly less accurate than initial localization, even after 1 min. Male subjects’ accuracy decreased over time, but there was no effect of delay for female subjects. A large effect of stimulus position was also noted, with stimuli coming from central locations remembered better than those coming from the extremes, an effect not accounted for by azimuthal differences in localization accuracy. Analysis of the error distributions indicated that this pattern was largely due to the presence of right–left reversals. These reversals suggest the existence of two separate components to auditory spatial memory: side and azimuth, and that these two parts can be dissociated.
On the potential of fixed arrays for hearing aids
Authors: Stadler R. W. , Rabinowitz W. M.
Published: October 2005

Microphone arrays with fixed (time-invariant) weights are directed at enhancing a desired signal from one direction (straight ahead) while attenuating spatially distributed interference and reverberation. Using the theory of sensitivity-constrained optimal beamforming [Cox etal., IEEE Trans. Acoust. Speech Signal Process. ASSP-34, 393–398 (1986)], free-field arrays of head-sized extents were studied. The key parameters affecting array design and performance are the set of transfer functions from the target direction to each array microphone and the intermicrophone cross-spectral densities for isotropic noise. Design variables included the orientation of the array, the number and [as motivated by Soede, Ph.D. thesis, Delft Univ. of Technol. (1990)] the directionality of the microphones within the array, and the complexity and robustness of the required processing. For broadside orientation, a variety of arrays based on cardioid and hypercardioid microphones gave very similar performance. They can provide broadband intelligibility-weighted directivities (direct-target gain against isotropic noise) of 7–8 dB with easily implemented weights (simple scalars). For endfire orientation, as Soede recognized, similar directivities result with weights based on analog gains and pure time delays. However, with weightings that are chosen independently for each frequency, directivities up to ∼11 dB may be practical. [Work supported by NIH.]
Jet noise generation mechanism
Authors: Lee Seungbae , Meecham William C.
Published: October 2005

The breakdown process of the initial large-scale structures into substructures near the end of the potential core is examined and the production of aerosound as a result is discussed. The quadrupole-like directivity pattern of jet noise is numerically obtained by applying the large-eddy simulation (LES) technique for the fluid flow. The Lighthill analogy is used to obtain the acoustic far field. Most of the sound comes from a region near the end of the potential core. A discussion to explain this quasiaxisymmetric directivity pattern is given. The directivity maximum is obtained at about 45° from the axis of the jet exhaust. This will be explained in terms of turbulent velocities. The convective effect on sources is introduced by ray tracing to calculate retarded times. The result is to obtain more radiation downstream, as is observed. The relation of this correction to the Lighthill theory will be discussed.
Testing instruments that measure transient sounds
Authors: Young Robert W.
Published: October 2005

An unattended instrument for audible environmental noise must measure and report whatever and whenever sounds occur at the monitoring location. Test signals for steady sounds are here voltage-equivalents of sound pressure levels; the signals are sine waves at one-third octave intervals, 10 Hz to 20 kHz. Test signals for responses to transient sounds are sound exposure levels of single-cycle sinebursts, 10 Hz to 20 kHz. For a wide-band system and 1 kHz, the sound exposure level of a single cycle relative to steady sound pressure level is theoretically −10 log(1000)=−30.0 dB. With A-weighting, the relative sound exposure level is theoretically −30.7 dB; with CEL-493 integrating sound level meter No. 198107 the relative level was also −30.7 dB. At 32 Hz (more precisely 31.6228 Hz), the theoretical relative A-weighted sound exposure level is −10.6 dB [not −10 log 31.6228=−15.0]; with the CEL-493 the relative level was −10.8 dB. For a conventional sound level meter, at 1 kHz the theoretical fast A-weighted sound level of a single cycle, minus steady sound pressure level, is −21.8 dB; for CEL-493 the difference was −21.7 dB. At 32 Hz the theoretical fast A-weighted sound level of the single cycle, minus steady level, is −2.2 dB; for CEL-493 the difference was also −2.2 dB.
Stop bands in periodic and random media
Authors: Economou E. N.
Published: October 2005

In periodic media, frequency regions may exist at which no bounded at infinity solution of the wave equation exists. In random systems frequency regions may appear at which all bounded solutions are localized, i.e., decay exponentially at infinity. Both regions give rise to stop bands. Such stop bands appear easily for acoustic waves in bubbly liquids. They are also predicted to exist for elastic waves in binary composites consisting of gold or lead spherical inclusions periodically placed in a Si, Be, or SiO2 (for gold inclusions only) matrix and occupying about 10% of the volume. The midfrequency ωg of the stop band was found to satisfy the relation ωg=ycil/Ri, where cil is the longitudinal velocity in the inclusion material, Ri is the radius of each inclusion, and y is a number between 1 and 2.
Modeling and simulation of tympanograms
Authors: Chen Lin
Published: October 2005

Tympanometry has become a standard evaluation procedure for children and adults in the audiological clinic. Because of the acoustical complexity of both the middle ear system and the testing device, there is a need to study tympanometry not only from an experimental but also a theoretical perspective to achieve a full understanding of the underlying mechanism. In the present study, a model has been developed for simulation of different type of tympanograms. In the model, the acoustic characteristics of the ear canal are described by a set of acoustic equations and those of the middle ear are described by the Zwislocki electric analog model. Model parameters corresponding to the ear canal length, the stiffness of the middle ear, and the resistance of the cochlea are manipulated to simulate the effects of the applied pressure in the ear canal during a tympanometric procedure. Typical tympanograms such as 1B1G, 3B1G, 3B3G, and 5B3G have been demonstrated with the model and are in good agreement with those actually recorded from the human ear. Some tympanograms seen under pathological and abnormal conditions are also simulated.
Integrated mathematical model of the ear. II. Application and insights
Authors: Price G. Richard
Published: October 2005

The model, described by Kalb in the previous paper at this meeting, was initially designed to predict hazard from intense impulses [G. R. Price and J. T. Kalb, J. Acoust. Soc. Am. 90, 219–227 (1991)]. In this mode it correlates highly with auditory hazard from impulses and has provided a radically different approach to an impulse noise damage-risk criterion. A feature of major significance is the ‘‘movie’’ it produces showing basilar membrane displacement synchronized with temporal progression through the pressure history. This display allows visualization of intracochlear events, promoting insight into the ear’s response to spectrally and temporally complex signals. In addition to research applications, the model has a potential pedagogical role. For example, if the basilar membrane is given a ‘‘Q’’ associated with active processes in the cochlea, then an additional range of demonstrations is possible. The ‘‘half-octave shift’’ for TTSs, the traveling wave, frequency localization, and the upward spread of masking can be shown. The model runs on a PC and with demonstrations is easily contained on a single floppy disk.
An assessment of current acoustical measurements in auditoriums in relation to previously established guidelines derived from perceptual experiments in auditorium synthesis
Authors: LoVerde John J.
Published: October 2005

Paul S. Veneklasen and Associates has used auditorium synthesis and acoustical modeling as a guide in the design of auditoria over the past 25 years [Paul S. Veneklasen, Int. Symposium on Arch. Acoustics, 21–42 (1974)]. Optimal settings for ratios of direct to envelopmental and direct to reverberant sound level were established based on perceptual data for a variety of program materials including classical music, music drama, speech, and pipe organ by experienced listeners including conductors, musicians, music critics, concert goers, and acousticians [P. S. Veneklasen, Acoust. Soc. Am. 77th Meeting, Session P, 27–28 (1969)]. Current auditorium acoustic parameters were measured using impulse response techniques in the auditorium synthesis facility for a variety of direct, envelopmental, and reverberant sound level settings. The range of values for the various auditorium synthesis settings is shown for the measured acoustic parameters [source strength (G), early decay time (EDT), early/late energy (C), and lateral energy fraction (LF)]. Based on the guidance from auditorium synthesis, two additional parameters are introduced that should be considered for a more complete assessment of auditoria acceptability. These are the direct to envelopmental and direct to reverberant energy ratios.
The control of reverberation
Authors: Veneklasen Paul S. , Atwal Mahabir
Published: October 2005

The variety of programs that a good modern auditorium should accommodate, requires that reverberation should be controllable if each program is to merit optimal acoustical environment. Guidance from auditorium synthesis suggests that, in contrast with the usual doctrine stressing reverberation time, the sound level of the reverberant sound relative to the direct sound is a dominant factor. Following on from the seminal work of Carl Eyring, this paper describes experiments and experience over the past 25 years that suggest a very practical way to achieve the necessary control.
Deflections and vibrations of the unimorph flexural disk transducer
Authors: Warren Daniel M. , Lafleur L. Dwynn , Shields F. Douglas
Published: October 2005

Previously, a simple electroacoustic model of a transducer constructed of flexural mode piezoelectric disks [Warren etal., J. Acoust. Soc. Am. 91, 2325 (A) (1992)] and a derivation of the governing differential equations for the flexural disk as a layered piezoelectric plate [Warren etal., J. Acoust. Soc. Am. 92, 2292 (A) (1992)] have been presented. In the current work, solutions of the differential equations are sought for static deflections and dynamic vibrations, under a variety of boundary conditions. The technique used to treat laminated plates with lamina of differing radii, of which unimorphs are examples, is emphasized. The results are compared to the classic theories [S. Hanish, NRL Rep. 5259 (1959) and R. S. Woollett, USL Res. Rep. 490 (1960)] of which this work is an extension. [Work supported by ONR.]
Split-dual beam array transducer design and computer simulation
Authors: Huang Dehua , Lan J. , Jeffers R. K. , Boucher S. G.
Published: October 2005

Both dual-beam and split-beam array transducers are widely employed in underwater sonar systems. The split-beam principle is used to find the position of individual targets, for target strength measurements and fish stock assessment. The dual-beam array transducer can transmit signals on one beam and receive the echoes with either one or two beams and has been widely used for target strength measurements and fish stock assessment. Here, a new array transducer has been developed, which combines the strengths of split-beam and dual-beam array transducers. The dual-beam mode is achieved by driving the linear array elements with Chebyshev weighting, while the split-beam mode is obtained by driving the quadripartite subarrays. Furthermore, this new array transducer can transmit signals in split-beam mode and receive the echoes in the dual beam mode, or vice versa. The configuration of the new array transducer will be presented. The beam patterns at different transmitting and receiving situations have been numerically simulated.
Electroacoustic evaluations of 1–3 piezocomposite rings
Authors: Howarth Thomas R. , Rittenmyer Kurt M. , Ting Robert Y.
Published: October 2005

Two 1–3 piezocomposite rings have been recently electroacoustically evaluated for sonar application. The two rings have outer diameters of 345 mm, 101.6-mm height and a wall thickness of 5 mm. Both have a 5% Navy type II piezoceramic volume concentration loading. The host epoxy of one ring is soft while the other is hard. Both rings had 17.78 by 25.4 mm rectangular elements etched into a 3 by 3 element pattern in their outer electrode for direct in-water performance comparison with a previous design utilizing individual elements of the same dimensions [Howarth etal., J. Acoust. Soc. Am. 91, 2325–2326 (A) (1992)]. The in-water measurements include wide-band free-field voltage sensitivity (FFVS), transmitting voltage response (TVR), and radiation patterns in both the XY and XZ planes. A laser Doppler vibrometry was used for in-air evaluation of the piezoelectric charge constant and surface displacement. Interelement coupling data will also be presented. [Work sponsored by the Office of Naval Research.]
Spacing effects in large-area hydrophone arrays
Authors: Brown Richard N. , Berkman E. F.
Published: October 2005

It has long been known that large hydrophones filter out the high-wave number, convective components of turbulent boundary layer (TBL) noise. A two-dimensional array constructed of N×M hydrophones might be expected to increase the filtering by a factor of N×M, but, in practice, inter-hydrophone spaces (‘‘gaps’’) diminish this benefit. An analysis, based on a wave-vector approach, has been done to study the effect of gaps in two-dimensional arrays. It is shown that the effect of gaps is to allow leakage into the array response function at wave numbers corresponding to aliasing lobes. This leakage can extend to moderately high wave numbers, although the details of the leakage are sensitive to the exact gap size. Numerical estimates of array self-noise due to flow noise are made using models of the TBL wall pressure wave-vector spectrum. It is shown that, at low values of k0L (k0 is the acoustic wave number and L is characteristic array length), the leakage from the convective and subsonic components of the TBL can be substantial, even for relatively small gaps.
Effects of discontinuous F3 transitions on phonetic identification and fusion
Authors: Hall Michael D. , Pastore Richard E.
Published: October 2005

According to general perceptual principles, figural goodness and feature integration of stimuli should reflect the continuity between components; in CV syllables this continuity might be between phoneme-distinguishing glide (formant transition) and vowel formant. Two experiments examined continuity effects on identification and fusion for sinewave syllables based on common CV parameters. Experiment 1 provided a perceptual basis for equating isolated glides and stationary tones. Independent of glide direction, perceptual matches tended to be based upon temporally adjacent portions of the stimuli. Using matched stimuli, experiment 2 evaluated the relative contribution to phonetic identification of (1) frequency continuity of transition and formant and (2) transition direction. Subjects discriminated the original syllables from binaural and dichotic syllables containing either continuous or discontinuous F3 transitions. Identification was influenced more by transition direction than by goodness of physical continuity. Furthermore, very different patterns of results were obtained for binaural conditions and dichotic conditions in which responses indicated fusion. The nature of information provided by transitions and operation of schema are discussed. [Work supported by NSF and AFOSR.]
The Academy of Music, Philadelphia—Preliminary model studies of the potential for coupled volumes
Authors: Marshall A. Harold , Kirkegaard Dana L.
Published: October 2005

A 1:24 scale model of the occupied hall is being used to explore the feasibility of coupling additional volume to the Academy of Music in Philadelphia. Data from several schemes will be compared with respect to both listening position and source position. Model measurements are being carried out with the MIDAS test system. The model was calibrated to measurements made in the occupied hall.
Attenuation of sound passing over audience at grazing incidence
Authors: Mommertz Eckard
Published: October 2005

In halls with horizontally arranged seats the direct sound propagates nearly parallel to the audience, which causes a frequency-dependent attenuation. Although the additional attenuation is remarkable not much experimental data are available in the literature so far; most measurements were carried out in empty halls. The present study examines the effects on sound passing over audience and unoccupied seats at grazing incidence. Measurements were carried out in two halls and for scale model seat rows, using maximum-length sequences in combination with the Hadamard transform. To improve the dynamic range the spectrum of the maximum-length sequences was digitally pre-emphasized in some cases. The investigation showed that the audience causes a considerable, almost constant attenuation in the range between 500 Hz and 1.5 kHz with an exponential relationship between distance and sound preasure decrease. For higher frequencies the attenuation was dependent on the horizontal angle of incidence with regard to the seat rows. In some cases a significant decrease of sound pressure due to interference effects was observed. In the low-frequency range the well-known ‘‘seat-dip’’ effect occurred.
Modified Klatt synthesizer (VSS–KS) and analyzer (INTANAKL) for control parameters
Authors: Ananthapadmanabha T. V. , Jayasimha H. N.
Published: October 2005

Klatt synthesizer (KLSYN88) is a successful hybrid formant filter model for speech synthesis [Klatt and Klatt, J. Acoust. Soc. Am. 87, 820–857 (1990)]. Several modifications are proposed to enhance the scope and quality of KLSYN88. For example, in VSS–KS, control parameters do not change abruptly; an impulse excitation is used for synthesis of stop consonants; aspirated stop consonants can be synthesized; microphonetic events with fine temporal resolution can be synthesized; parallel formant filter is correctly implemented; source–filter interaction (up to three formants) is implemented as formant frequency and bandwidth modulation rather than abrupt jumps; etc. INTANAKL estimates many of the control parameters like formant frequency, bandwidth, level, voice source parameters, aspiration noise component, pole-zero component, etc. The estimated parameters are validated by comparing the short-time spectra of the original and synthesized speech and source waveforms. VSS–KS and INTANAKL are described in this paper.
The impact of backscatter angular dependency on reverberation levels in shallow water under downward refracting conditions
Authors: Herstein Peter D. , Cole Bernard F. , Brown Kerry E. , Podeszwa Eugene M. , Browning David G.
Published: October 2005

For downward refracting conditions in shallow water over a sandy bottom it was found that propagation loss was dependent on the source and receiver depth [P. D. Herstein etal., J. Acoust. Soc. Am. 92, 2302(A) (1992)]. The analysis is now extended to reverberation levels, considering the grazing angles corresponding to various source and receiver depths and both the first and second power sine of the angle dependencies of the backscattering. This is compared to a compilation of shallow water backscattering results.