bluuebunny/crossref_metadata_2025 huggingface dataset:
Masking patterns for comodulated and uncomodulated narrow-band noises as a function of masker band separation
Authors: Fischer Rebecca M. , Grantham D. Wesley
Published: October 2005

Masking patterns were determined for a pair of narrow-band (25 Hz wide) noises whose envelopes were perfectly correlated, uncorrelated, or negatively correlated. In one condition, noise bands were centered at 962.5 and 1037.5 Hz in order to examine the effects of comodulation masking release (CMR) when paired noise bands were within one critical band; the second condition used noise bands centered at 887.5 and 1112.5 Hz, a separation of greater than one critical bandwidth. Signal thresholds were determined for a range of frequencies (750–1250 Hz) above and below those of the maskers in order to examine the process of CMR outside the immediate spectral region of the maskers. Preliminary data suggest that CMR gradually decreases as the signal is moved away in frequency from the center of either noise band. A comparison of CMR in positively and negatively comodulated conditions is expected to shed light on the type of mechanisms underlying CMR within and across critical bands.
Perception of pitch structure in pure-tone melodic sequences
Authors: Oram Nicholas , Cuddy Lola L.
Published: October 2005

Musically naive subjects were tested in two experiments investigating the perception of pitch structure in pure-tone sequences. Each sequence contained 20 successive samples from either a diatonic or a nondiatonic toneset of seven tones with probability of occurrence of each tone systematically varied. Sequences were presented in either of two frequency ranges (262–494 Hz, or 1047–1976 Hz). Frequency range was varied between subjects in experiment 1 and within subjects in experiment 2. On each trial, a sequence was followed by a probe tone, one of the 12 chromatic tones within the same octave as the sequence. Subjects rated the goodness-of-fit of the probe tone to the sequence. In both experiments, probe-tone ratings were positively related to probability of occurrence of the probe tone in the sequence. Moreover, for sequences generated from diatonic tonesets and presented in the high-frequency range, ratings also reflected the tonal hierarchy implicated by the key of the toneset. [Work supported by NSERC.]
Binaural signal detectability in noise amplitude-modulated (NAM) noise as a function of masker bandwidth
Authors: Hedrick Mark , Grantham D. Wesley
Published: October 2005

Thresholds were determined for a 500-Hz tonal signal in the presence of a noise amplitude-modulated (NAM) noise, where the carrier was a wideband Gaussian noise and the modulator was a 20-Hz-wide noise band centered at 20 Hz. The NAM noise was filtered after modulation to produce a band centered on the 500-Hz signal. Both diotic (So) and interaurally phase-reversed (Sπ) signal thresholds were determined as a function of masker bandwidth (10–1000 Hz). When the masker was presented diotically, So thresholds (and to a lesser extent, Sπ thresholds) first increased, then decreased as a function of masker bandwidth, thus showing the effects of comodulation masking release at the wider bandwidths. The masking-level difference (MLD, i.e., the difference between So and Sπ thresholds) increased as masker bandwidth decreased, as it did with a random (unmodulated) noise masker. When the masker was presented such that the noise carrier was diotic but the noise modulators were independent at the two ears, a sizable MLD still occurred (5–10 dB), but the MLD decreased with decreasing bandwidth. The decreased MLD at small bandwidths resulted from the higher Sπ thresholds with this stimulus than with a true diotic masker, which can be explained by the slowly fluctuating interaural intensity differences that this type of masker produces at narrow bandwidths. [Work supported by NIH.]
Binaural frequency-following response (FFR) to lateralization stimuli
Authors: Ballachanda B. B. , Moushegian G.
Published: October 2005

The cues for localization or lateralization are interaural disparities of time and intensity, and neurophysiological studies have elegantly demonstrated that lower-brain-stem neurons, in conformity with anatomical observations, are differentially sensitive to the stimulus parameters that mediate binaural phenomena. Brain-stem responses were obtained using low-frequency (500 Hz) stimuli in various combinations of interaural phase (time) and interaural phase and level presented in-concert and in-opposition through earphones. The findings show that the amplitudes and morphologies of the volume-conducted evoked FFR potentials are differentially altered as a function of time and level variables. There are significant reductions, for example, in the amplitude of the evoked responses due to interaural time differences. Furthermore, for in-concert and in-opposition time and level stimuli, the amplitude changes are distinctly different, reflecting the complex neuronal interactions at the level of the brain-stem nuclei and tracts.
Human auditory localization performance in azimuth
Authors: McKinley Richard L. , Ericson Mark A.
Published: October 2005

Human auditory localization performance in the free-field, over headphones via a simulator and over headphones via a real-time, head coupled, digital auditory localization cue synthesizer will be described in the presentation. In the headphone conditions, the head position is monitored to update the sound source position relative to the listener's orientation. The synthesizer presents sounds over headphones that are perceived to be outside of the listener's head in the horizontal plane. The performance experiments were designed using a within subject, repeated measure study. The stimuli used were wideband pink noise, male speech bandlimited to 3.5 kHz, female speech bandlimited to 6 kHz and octave band noise from 125 Hz to 8 kHz. The mean magnitude error and mean response time were measured at 72 random points for each stimulus. Five male and five female subjects participated. A 4.8-deg mean magnitude error and a 4.4-s mean response time were collected with the synthesizer.
Current and future topics in modal analysis at the University of Cincinnati
Authors: Rost Robert W.
Published: October 2005

The University of Cincinnati, Structural Dynamics Research Laboratory, has been involved in the area of modal analysis for over 20 years. Currently, there are 5 faculty members and 15 graduate students who work with UC/SDRL. UC/SDRL has worked with the United States Air Force, NASA, General Motors, and many other industries on a variety of research and consulting topics. Current and future technology in the area of modal analysis will be discussed. Current topics to be discussed will include multi-input/multi-output testing techniques, parameter estimation for multi-input/multi-output testing, and work station environment. Future topics to be discussed will include step sine testing, active control, and new signal processing techniques.
Modal identification for hysteretic structure
Authors: Inman Daniel J.
Published: October 2005

This work describes a method of analyzing the vibration response of a viscoelastic beam. The equation of motion of the transverse vibration of a beam with viscous damping and temporal hysteresis is given. Several methods for measuring the constants of a given hysteresis kernel are presented. One is based on a least-squares fit to data using inverse procedures for partial differential equations. The second approach discussed is that of introducing an auxiliary coordinate. This coordinate combined with a transfer function model of the hysteresis effect yields a modal analysis solution for the viscoelastic beam. It is shown that identification is then possible by using modal analysis techniques. This theoretical derivation is compared with test data.
Sound sources of low-speed jet impingement on a large flat plate
Authors: Powell Allan
Published: October 2005

Jet impingement on a large flat normal plate produces a noisy turbulent divergent wall jet. As in the more general case, the image flow in the plate can be postulated and the “three sound-pressures theorem” applied (neglecting dissipation effects): For long wavelengths (“compact” flow), the aerodynamic sound sources must be reducible to lateral quadrupoles alone, with maxima normal to and in that plane. Additional (usually weak ??) shear dipoles associated with the viscous traction forces lie in the plane of the plate. Specifically, the large scale axisymmetric unsteady flow associated with the flow resonances that sometimes occur resembles a periodic series of ring vortices that approach the plate and then spread out, symmetrically. Regardless of the details of this symmetrical unsteady flow, the associated source field must be periodic annular distributions of the lateral quadrupoles and traction dipoles; in the long-wavelength (compact) low-speed case these degenerate into symmetrical point quadrupoles, one lateral and one longitudinal, yielding “U8 laws” for simple similarity. Dissipation results in “U8-type” monopoles due to heat addition fluctuations.
Modeling of fluid-injection effects on flow noise
Authors: Christoph George H.
Published: October 2005

Based on Naval Underwater Systems Center [communications with E. Payne of NUSC] and Soviet [Sov. Phys. Acoust. 30(5), 394–397 (1984)] water-tunnel data, the Corcos wall-pressure model was modified to include fluid-injection effects. Boundary-layer parameters were used to model the wall-pressure frequency spectral density and the longitudinal cross-spectral density. Using predictions from a Martin Marietta boundary-layer code, favorable comparisons were made to the NUSC measured boundary-layer profiles and wall-pressure frequency spectra. Excellent agreement was obtained for both the effect of injection rate and the reduced effectiveness of fluid injection with distance from the injection source. Calculated wavenumber spectra showed that at low frequencies (less than about 200 Hz) spectral levels increased with injection, while at higher frequencies spectral levels decreased. The predicted convective peaks shifted to higher wavenumbers due to the decrease in convective velocity with increasing injection rate. The rms wall pressures are shown to increase for all fluid-injection rates. The model predicts that the large flow-noise reductions (over 20 dB) obtained in water-tunnel experiments will not be achieved in full-scale applications.
Modal analysis of rotating machinery
Authors: Ying S. P.
Published: October 2005

Modal analysis of rotating machinery and its supporting structure is discussed. This paper presents the natural frequencies, damping ratios, and mode shapes of various modes of rotating machinery systems such as fans and radar antenna foundations as examples. For a rotating machinery, it is necessary to have a dynamic correction factor for the dynamic natural frequency of a rotor (critical speed). The dynamic correction factor for ventilation fans is presented with examples. The overall natural frequency of an entire system depends on the dynamic combination of the rotor and its supporting structure. If a heavy duty rotor bearing is used, the mode for the rotor having a rigid support becomes significant, which simplifies design, and the dynamic sensitivity of the system is reduced. For a large industrial fan, an early foundation design is compared with an improved foundation design in later stage from a mode shape viewpoint. Based on suggestions developed from modal analyses, a supporting structure was improved, gear noise resulting from torsional vibration was eliminated, and a rotor resonant problem was resolved.
Acoustic wave scattering by inhomogeneous obstacles in layered media
Authors: Lu I. T. , Jung H. K.
Published: October 2005

Acoustic wave scattering by homogeneous obstacles in layered media has been studied by a new method that combines the hybrid ray-mode and boundary element methods [I. T. Lu, IMACS Meeting, Princeton Univ., March 1989]. The latter has been employed to formulate the scattering process and the former to provide the Green's function of the layered environment. Here, the new method is extended to analyze wave scattering by inhomogeneous scatterers by including the finite element method. By finite element method, the inhomogeneous obstacles are divided into small elements and the field on these elements are formulated in terms of a system of algebraic equations. By employing the boundary element method, the field distribution along the boundaries of the obstacles are formulated in terms of integral equations which are subsequently reduced to algebraic equations. In the integral equations, one needs to compute the Green's function of the layered environment for various arrangements of locations of source and receiver. The newly developed hybrid ray-mode method is best suited for this purpose because it combines rays and modes self-consistently within a single framework and optimizes the advantage of each. Numerical implementation illustrates these aspects. [Work Supported by NSF.]
Angle spreading of acoustic energy on bottom bounce paths
Authors: McCammon Diana F.
Published: October 2005

A model for the spatial distribution of bottom returning energy has been developed that relies on a combination of surficial roughness and sublayer roughnesses to produce a non-Gaussian probability distribution of returns. Sediment inhomogeneities and sublayering are modeled by the artifice of assuming that the rms slope of the surficial layer increases with grazing angle. Deep region inhomogeneities are modeled by choosing a high value of 22° for the rms slope of the basement. These simplifying assumptions keep the model inbounds computationally and produce the correct behavior for angle spreading in both thick and thin sediment regions. Excellent agreements are shown with data. [Work supported by NAVSEA.]
Cross correlation of the repeated transmissions of signals scattered at a rough sea floor
Authors: Clay C. S.
Published: October 2005

Real measurements of the cross correlations of repeated transmissions from a source ship to a receiver on another ship must contend with drifting ships. Over a rough sea floor, the cross correlations depend on the geometry of the experiments, the nature of the roughness, and transmission techniques. Green [J. Acoust. Soc. Am. Suppl. 1 68, S72 (1980)] described multiple transmission techniques that can reduce the effects of receiver displacements between transmissions. Since it is easy to control drifting ships in numerical calculations, it was decided to use numerical simulations. The wedge assemblage and facet ensemble method were used to compute impulsive signals scattered at a rough bottom [C. S. Clay and W. A. Kinney, J. Acoust. Soc. Am. 83, 2126–2133 (1988)]. The results show that one can expect cross correlations as high as 80%–90% for receiver displacements of 25 m between transmissions. [Research supported by Office of Naval Research and the Geophysical and Polar Research Center of the University of Wisconsin-Madison.]
Observations on perturbation theory of scattering by surface waves
Authors: Mellen Robert H.
Published: October 2005

First-order perturbation theory of scattering by surface waves is usually expressed in terms of the wavenumber spectrum of the surface roughness. Bistatic reverberation is calculated from the Bragg condition and coherence loss by integration over the wavenumber range corresponding to real scattering angles. An alternative angular-spectrum approach is presented that provides additional insight into both the elementary acoustics and the nature of the approximations involved. Scattering is shown to be equivalent to dipole reradiation by a random ensemble of coherent arrays of area πL2, where L is correlation length. Phase delay of the excitation then produces a scatter beam pattern having horizontal broadside and vertically steered endfire components. Some of the possible hazards involved in comparing predictions with experiment are discussed.
Improved duration rules for text-to-speech synthesis
Authors: Syrdal Ann K.
Published: October 2005

The durations of phonemic segments in an utterance synthesized by rule are important because they affect both intelligibility and prosodic quality. There are many factors that influence segmental duration, including the identity of the segment, and extrinsic factors such as surrounding segments, lexical stress, syntax, and speaking rate. Several approaches to modeling the combined effects of the contributory factors have been proposed. Until recently, the duration rules of the experimental AT&T dyad synthesis-by-rule system have been primarily additive. Starting with a minimum duration for each segment, duration was augmented incrementally by each contributory extrinsic factor. However, in natural speech, the effect of an extrinsic factor tends to vary depending upon other contributory factors. Analyses of natural speech [H. S. Gopal and A. K. Syrdal, J. Acoust. Soc. Am. Suppl. 1 82, S16 (1987)] indicated that Klatt's incompressibility model was more accurate than strictly additive or multiplicative approaches. Consequently, a new duration rule program was developed for the AT&T text-to-speech system, using an incompressibility approach. Comparative evaluations indicate that segment durations determined by the new duration rule system are more accurate and are preferred significantly more often by listeners than durations determined by the additive rules.
A speech synthesizer for rule-based synthesis
Authors: Mitome Yukio , Umeda Noriko
Published: October 2005

A new speech synthesizer is proposed that is suitable for speech synthesis by rule. This synthesizer is based on a source-filter model and consists of two sources—a glottal source and a noise source—and three cascade filters. Each source can be connected with any of the filters dynamically. Each filter is used as a speech component, which is produced as a source signal filtered by a slow varying system and has a limited duration. The speech signal consists of some components overlapping each other in time-frequency domain. The experimental system was developed on a 32-bit personal computer. The software system has two subsystems, a speech synthesizer with a format pattern interpreter and a speech analysis system with a speech database. A formant pattern interpreter reads a formant pattern definition file, and a controls speech synthesizer routine. Speech analysis is based on the maximum entropy method (MEM), and a database contains 25 min of recorded male voice samples. Experimental results showed that this synthesizer can easily realize the spectral discontinuity between phonemes.
TFT: An algorithm for the spectral compression of natural speech signals
Authors: Hurtig Richard R.
Published: October 2005

An earlier report [R. R. Hurtig, J. Acoust. Soc. Am. Suppl. 1 81, S78 (1986)] using synthesized syllables demonstrated that naive subjects can discriminate and identify spectrally compressed vowel segments under auditory and vibrotactile conditions. These findings are consistent with the view that the identification of the spectral shape of the speech segment may be independent of its frequency range. A computational algorithm was developed to achieve spectral compression of natural speech. The algorithm includes calculation of an n-point FFT, padding the result with the spectrum of a Hamming window, calculation of a 2n-point IFFT, and outputting the first half of the resultant time domain signal. The size of the pad determines the amount of compression achieved while the placement of the pad determines the direction of the frequency shift. Naive subjects had no difficulty recognizing simple sentences in a closed set for speech signals compressed to 2500 or 1250 Hz bands. After a few hours of listening, open set recognition is achieved. The implementation of the algorithm for sensory aids will be demonstrated.
The noise environment under low-altitude, high-speed military aircraft training routes
Authors: Plotkin Kenneth J. , Chavis Alton
Published: October 2005

Low-altitude, high-speed training operations are routinely conducted along specially designated Military Training Routes (MTRs). Design of new routes and/or realignment of existing routes requires an environmental assessment to determine the community noise impact. Nominally, aircraft on these routes navigate from point to point along defined segments. Key elements required for noise prediction are the frequency of flights, the statistical variation of position relative to the nominal centerline, and the operational noise emission levels of the aircraft. Noise measurements were conducted on three routes: one operated by the Strategic Air Command and two by the Tactical Air Command. On each route, 20 automatic noise monitors were deployed on a 2- to 4-mi array across the route centerline. Major findings were: noise emission levels are fully consistent with predictions from USAF's NOISEFILE database; aircraft tend to fly within the central part of a route; aircraft follow nominal tracks corresponding to defined route centerline or corresponding to prominent visual references; the lateral distribution about each track is Gaussian; and multiple tracks can exist. Noise events were infrequent (typically less than three or four per day), and the highest Ldn measured was less than 65 dB. [This work was sponsored by USAF AAMRL/BBE.]
ROUTEMAP model for predicting aircraft noise exposure along military training routes
Authors: Lucas Michael J. , Plotkin Kenneth J.
Published: October 2005

A model and PC-based computer program has been prepared to calculate noise levels along low-altitude, high-speed military training routes. The program is designed for use by environmental planning personnel who are familiar with MTR operations and with noise, but are not necessarily expert. The program provides options for selecting general types of operations (visual or instrument navigation), aircraft types and speeds, altitudes, and nominal track centerlines. Up to 20 track/altitude/aircraft types may be defined within a 20-mi-wide corridor. Aircraft on each track have a Gaussian lateral distribution about the centerline. The program contains nominal standard deviations based on the type of operations, or the user may specify a site-specific value. Aircraft noise emission levels are derived from the USAF NOISEFILE database. The program calculates Leq, Ldn, and Ldnmr, where Ldnmr is Ldn with an adjustment to account for the onset rate of MTR aircraft noise. Program output is available in tabular form or in graphs suitable for inclusion in reports. [This work was sponsored by USAF AAMRL/BBE.]
Sonic boom spectra of Atlantis landing 6 December 1988
Authors: Young Robert W. , Awbrey Frank T.
Published: October 2005

After circling the world 69 times, orbiter Atlantis came to a stop on Runway 17 of Edwards Air Force Base in California, at 1537U on 6 December 1988. Some 10 mi to the west, and 4 min previous to landing, its 394-ms sonic boom swept over our measurement site at latitude 34.886°N, longitude 118.036°W, elevation about 2.5 kf above sea level. When the decelerating Atlantis heading southeast passed nearest to the microphones at lateral slant range of 100 kft, it was gliding at Mach 1 about 48 kft above ground. The peak flat sound pressure level of the sonic boom was 129 dB; the peak C-weighted sound pressure level, 125 dB; the peak A-weighted sound pressure level, 110 dB. The flat sound exposure level was 118 dB; the C-weighted and A-weighted sound exposure levels of the initial transients were, respectively, 102 and 84 dB. If sound exposure level is wanted for a 400-ms sonic boom with primary emphasis in the range 0.8–3 Hz, flat sound exposure level is appropriate; with primary emphasis in the range 10–50 Hz, C-weighted sound exposure level is appropriate; with primary emphasis in the range 80–1000 Hz, A-weighted sound exposure level is appropriate.
Revision of a dosage-effect relationship for the prevalence of noise-related annoyance
Authors: Fidell Sanford
Published: October 2005

More than a decade has passed since Schultz (1978) derived a relationship between noise exposure and the prevalence of annoyance from the findings of various social surveys of community response to general transportation noise sources. Numerous measurements of the prevalence of noise-related annoyance have been published in subsequent years. A least-squares quadratic fitting function to a set of 427 data points (developed by addition of 266 new data points to the original set of 161 data points) differs little from the third-order polynomial fit to the original data set. [Work supported by U.S. Air Force Noise and Sonic Boom Impact Technology Program under Contract F33615-86-C-0530.]
Thermoacoustic refrigerator for space applications
Authors: Garrett S. L. , Hofler T. , Fitzpatrick M. , Susalla M. P. , Yolkert R. , Harris D. , Byrnes R. B. , Cameron C. B. , Murray F. M.
Published: October 2005

Long-lived, space-based cryocoolers with low vibration levels are necessary in a variety of applications, including the cooling of infrared sensors and high Tc superconductors. A thermoacoustic refrigerator designed to function autonomously in a Space Shuttle Get Away Special (GAS) canister will be described that has no sliding seals and oldy 15 gm of reciprocating mass. It is capable of cooling to 100 K below room temperature with a single stage “stack” consisting of rolled plastic film and short lengths of 10-lb test monofilament fishing line as the spacers. This presentation will concentrate on the modifications to the basic refrigerator design (U.S. Patent No. 4,722,201, issued 2 Feb. 1988) that improve both its electroacoustic and thermoacoustic efficiencies. These improvements include the use of a helium-xenon gas mixture as the thermodynamic working fluid and a neodymium-iron-boron electrodynamic driver with a titanium suspension. Custom electronic circuits necessary to control the refrigerator (microprocessor, bubble memory data recorder, phase-locked loops, automatic gain controls, amplifiers, multiplexed diode thermometers, etc.) and special fabrication techniques necessary to confine the helium for years while permitting electrical feed through will also be disclosed. [Work supported by the Office of Naval Research, Office of Naval Technology, and the Naval Postgraduate School Research Foundation.]
Source characteristics: A study of hand clapping
Authors: Tousman Stuart A. , Pastore Richard E. , Schwartz Steven
Published: October 2005

The present research evaluated the capabilities of listeners to determine source characteristics from acoustic signal properties. Repp [J. Acoust. Soc. Am. 81, 1100–1110 (1987)] demonstrated that naive listeners have some ability to determine the source characteristic of hand position given the acoustic signal of a clap. The purpose of this hand clapping experiment was (1) to more directly assess the capabilities of naive listeners to determine source characteristics, (2) to determine the physical aspects of the acoustic signal that are highly correlated with the source, and (3) to train subjects so that performance could be improved. The production phase of the experiment consisted of recording claps in five different hand positions. The perceptual phase consisted of a within clapper same/different task with listeners asked to judge whether the hand position was the same (never physically identical) or different. Half of the listeners tested received feedback. Preliminary results indicate that naive listeners could distinguish hand position from only a few clappers. Furthermore, training over 360 trials was only marginally beneficial in improving performance. Results will be discussed in terms of different methods of assessing and improving listener performance. [Work supported by BRSG program grant.]
Accuracy and reliability of formant frequency measurements in infant CV-like utterances
Authors: Daniloff Raymond G. , Roussel Nancye C. , Miller Creighton J.
Published: October 2005

The accuracy and reliability of measurements of infant formant frequencies were investigated. A total of 40 CV-type tokens (C = labial- or velarlike stop closant, V = /i/-like or /ɛ-like vocalic segment) were selected from recordings of vocalizations of two infants, ages 2–4 months. The tokens were of good acoustic quality and were taken from recordings of postfeeding mother-child interactions. Formant frequency measurements were made from both single FIT cross sections using a 10-ms sampling window and averaged, time-advanced, spectra. Time-advanced FFTs were computed using three successive FFTs taken at 2.5-ms rightward shifts of the 10-ms window. Three investigators independently measured formant frequencies in all tokens. Wideband 500-Hz spectrograms of each token were used to establish baseline values of formant frequencies for comparison with measured FFT values. Inter- and intraobserver reliability was also determined.
Speech recognition using codebooks trained from a synthesized seed
Authors: Smith L. A. , Scott B. L. , Lin L. S. , Newell J. M.
Published: October 2005

Speech sounds generated by a simple waveform synthesizer were used to create a 16-vector codebook for vector quantization (vq). This codebook was used as a seed for training over the TI-20 isolated word database, which was then used to test the resulting codebooks using speech recognition. The database was split into two sets of speakers and two codebooks were created for each set, one trained on one pass over the speech and the other trained iteratively. All codebooks were trained adaptively, resulting in codebooks of 31 vectors—16 high-energy and 15 low-energy vectors. Recognition was performed using a conventional DTW algorithm. The baseline recognition accuracy (using no vq) was 97%; the accuracy for the 16-vector synthesized codebook was 91.8%. Accuracy using trained codebooks ranged from 94% (two-sided vq) to 95.1% (one-sided vq with a normalized match distance). Crossing speaker sets (resulting in speaker-independent codebooks) yielded 94.9% recognition accuracy. The results indicate that speech synthesis can be used to decrease the training needed to develop multispeaker and speaker-independent codebooks.
Effects of tree structure and statistical methods on broad phonetic classification
Authors: Delmege James W. , Hillenbrand James , Gayvert Robert T.
Published: October 2005

The goal of this project was to develop a system for assigning individual frames of a speech signal to one of four broad phonetic categories: vowel-like, strong fricative, weak fricative, and silence. Classification results were compared from a K-means clustering algorithm and a maximum likelihood distance measure. In addition to the comparison of statistical methods, this study compared classification performance using several tree-structured, decision-making techniques. Training and test data consisted of various combinations of 98 utterances produced by five male and five female speakers. Results showed very little difference between the K-means and maximum likelihood methods. However, the nature of the decision tree had a significant effect on the performance of the classifier. [Work supported by Rome Air Development Center and the Air Force Office of Scientific Research as part of the Northeast Artificial Intelligence Consortium (Contract No. F3060285-C-60008) and by Redcom Laboratories, Victor, NY.]
A comparison of x-ray microbeam and ultrasound measurements of the tongue surface during speech
Authors: Stone Maureen
Published: October 2005

Tracking points on the tongue surface provides unique information about its movement, such as surface expansion/contraction, differences in rate of movement along the tongue's surface, and X-Y tongue position. Scanned images of the tongue surface provide different but equally unique information, such as posterior and anterior tongue relationships, tongue shape, tongue rotation, and asymmetry. The present study used two quite different techniques (x-ray microbeam and ultrasound) in a comparative study of tongue movement. A single subject repeated /VCVC/ combinations of /s/, /l/, /i/, /a/, and /o/. The phonemic effects of the consonants on tongue shape were examined, as were the allophonic effects of the three vowels. Tongue minus jaw movements (ultrasound) were compared to tongue plus jaw movements (XRMB). The data indicated that tongue rotation, which was quite common, occurred about a fulcrum whose locus was shifted or obscured by jaw movement. Other results indicated that there was considerable expansion/contraction of the tongue surface during speech. In addition, the surface of the tongue moved at nonuniform rates that were both physiologically and phonemically based.
Volume determination using acoustic resonance
Authors: Rothman Neil S.
Published: October 2005

Cavities of irregular geometry can be modeled as Helmholtz resonators provided they adhere to specific dimensional limits. If the characteristics of the entrance region or “neck” of a cavity are held fixed, its resonant frequency will be a function only of its volume. This property of a Helmholtz resonator enables the determination of the volume of various cavities through resonant frequency measurements. Using an electronic feedback technique and hand-held adaptor, this methodology has been applied to determine the volumes of both simple and complex cavities including castings of human nasal passages. The aim of this research is the development of a laboratory system of measurement of nasal passage volume in live humans. This will allow the determination of volume changes resulting from various stimuli and provide further insight into the heat and mass transport phenomena that occur in upper respiratory airways.
Smart acoustically active surfaces
Authors: Shields F. Douglas , Hendrix James E. , Lafleur L. Dwynn
Published: October 2005

Bilayered transducing surfaces have been constructed and used to control the reflection of sound in both liquids and gases. The acoustic signal generated in one of the layers is amplified, passed through a digital filter, and used to drive the second layer with the proper phase and amplitude so as to eliminate the reflections of single-frequency tone bursts. The tone burst frequencies were between 20 and 40 kHz in liquid and between 5 and 20 kHz in air. Feedback is controlled by the digital filter so as to eliminate instabilities. A theoretical analysis predicts the performance of such smart surfaces, but as yet there are discrepancies between theory and experiment. [Work supported by ONR.]
Independence of spatial and temporal tuning in the NP I cutaneous mechanoreceptor system
Authors: Van Doren Clayton L.
Published: October 2005

Results from a previous study that measured tactile detection thresholds for spatiotemporal sinusoids suggest that the spatial and temporal tuning of the P and NP I cutaneous mechanoreceptor systems are independent. In other words, provided that the spatial configuration of a vibratory stimulus is independent of temporal frequency, changing the spatial configuration should not change the shape of the temporal tuning curve of either system. To test this hypothesis for the NP I system, vibrotactile thresholds were measured as a function of temporal frequency (10, 12, 15, 20, 25, 30, 40, 55, 80, 100, 175, and 250 Hz) for 200-ms stimuli delivered to the thenar eminence via a 0.75-cm2 circular contactor, with and without a rigid surround. At frequencies where the threshold was mediated exclusively by the NP I system, the presence of a rigid surround did not significantly alter the shape of the temporal tuning curve, but did change the overall sensitivity. That is, NP I thresholds were uniformly higher without the surround than with the surround. Similar results were obtained when the experiment was repeated on the forearm. The observed invariance in the shape of the low-frequency tuning curve is consistent with the spatiotemporal independence hypothesis, but is consistent with conclusions from prior studies. The apparent discrepancy will be discussed.
Further tests of “composers' pulses” in computer performances of piano music from the classical period
Authors: Repp Bruno H.
Published: October 2005

Composers' pulses are patterns of expressive microstructure (i.e., timing and amplitude modulations) said to convey individual composers' personalities when implemented in computer performances of their music [see M. Clynes, in Studies of Music Performance, edited by J. Sundberg (1983), pp. 76–181]. The present research continued the perceptual evaluation of such pulses, which so far has yielded inconclusive results [B. H. Repp, Music Perception (in press)]. The initial bars of five piano pieces by each of four composers (Beethoven, Haydn, Mozart, and Schubert) were played on a computer-controlled electronic piano with each of the four composers' pulses (following Clynes' specifications as closely as possible) and also in a deadpan version. Listeners judged to what extent each computer performance conveyed the composer's characteristic expression, relative to the deadpan version. Pilot results provide occasional positive support for the Beethoven and Haydn pulses, but none at all for the Mozart and Schubert pulses. Additional data will be reported from musically trained subjects who also listened to computer performances in which the timing and amplitude components of the pulses were applied separately. [Work supported by NIH-BRSG.]
Ear canal cross-sectional acoustic modes
Authors: Friedrich M. T. , Rabbitt R. D.
Published: October 2005

The cross-sectional acoustic modes for real ear canal geometries are determined using a WKB method in combination with a Rayleigh-Ritz procedure. This model, which applies the usual assumptions of linear acoustics, can be used to analyze the three-dimensional sound-pressure distribution in any long, slender, rigid-walled waveguide. The method is particularly well suited to obtain the higher modes, which are trapped near both ends of the ear canal. The analysis was confirmed by comparing computed results with the exact solutions for fundamental waveguides. Results were also obtained for geometries measured from real ear canals. [Work supported, in part, by the Whitaker Foundation.]
Nonlinear aspects of the acoustic levitation of compressible spheres
Authors: Church Charles C.
Published: October 2005

The use of acoustic levitation cells has become increasingly popular in both basic research and industrial materials processing. Most theoretical studies of the forces involved have assumed a linear response of the levitated object to the acoustic field. In this study some of the effects of the nonlinear responses of highly compressible spheres (i.e., gas bubbles in a liquid) on the time-average radiation force are described. The results show that, for example, as the pressure amplitude increases (1) the radius at which the force changes sign decreases; (2) the positions in the field at which the force is maximal shift relative to the positions of the maxima determined using linear theory for radii less than the fundamental resonance; (3) at a fixed position in the field, harmonic resonance responses produce local maxima in the acoustic force. The presence of the higher-order harmonic resonances also may be seen in calculations of the levitation number for a suspended bubble. Implications for levitated liquid drops will be discussed. [Work supported by NIH.]
Oscillatory motion near a shape edge
Authors: Thompson Charles
Published: October 2005

The problem of linear transmission of sound over a sharp edge is one of the oldest in acoustics. However, only recently has interest been taken in the nonlinear attenuation properties of such structures. A theoretical model for the scattering and dissipation acoustics will be presented. In particular, the role that vorticity generation and transport plays will be discussed.
Acoustic streaming generated by surface oscillations
Authors: Thompson Charles , Mehta Vineet
Published: October 2005

The results of a numerical analysis of acoustic streaming caused by the oscillatory flow over a two-dimensional boundary will be presented. The effect of spatial and temporal irregularity will be addressed. The validity of models of the oscillatory and time-averaged boundary layer will be explored.
Coherent acoustic field computations using first-order multiple scattering and the generic sonar model
Authors: Bates Susan M.
Published: October 2005

The generic sonar model (GSM) is modified to include, as an option, the computation of the coherent acoustic field in an ocean environment with internal waves. This modification to GSM permits estimation of coherent signal processing performance. GSM computes the acoustic pressure in a horizontally stratified ocean in the absence of environmental inhomogeneities using WKBJ solutions to the Helmholtz equation. Using the first-order multiple scattering approximation and the Garrett and Munk internal wave model, the coherent acoustic field is included in GSM for frequencies from 100 to 10 000 Hz. Results are presented using the modified GSM with internal waves. Comparisons are made with earlier results, which used the parabolic equation method [B. J. Bates and S. M. Bates, J. Acoust. Soc. Am. 82, 2042–2050 (1987)], and GSM without internal waves.
Adaptive vibration control for structural mounting systems
Authors: Sommerfeldt Scott D. , Tichy Jiri
Published: October 2005

It is often desired to isolate a source of structural vibration from surrounding structures. Typically, the vibrating structure is attached to the foundation via some form of passive, flexible mount, as a means of accomplishing the desired objective. One weakness common to all passive schemes is that they produce relatively little (or no) vibration reduction at the lower frequencies. Active control provides a means of compensating for this weakness to achieve good vibration isolation over a wide frequency range. In this study, adaptive closed-loop control has been combined with passive control to give both active reduction at the lower frequencies and passive reduction at the higher frequencies. The adaptive nature of the controller allows the controller to converge to the optimal solution, as well as to track any variations in the system parameters that may occur. A controller has been developed, based on the LMS algorithm, and investigated to determine its stability and convergence properties. As well, the mechanical coupling that occurs when multiple control actuators are used has been studied. Real-time experimental results have been obtained from mounted vibrating structures, using a high-speed digital signal-processing board to run the control algorithm.
Sound transmission measurements through concrete block walls with attached drywall layers
Authors: Warnock A. C. C.
Published: October 2005

Sound transmission through a series of concrete block walls was measured in a reverberation room according to ASTM E90. The measurement frequency range was extended down to 63 Hz. The effects of the method of attaching the drywall and of adding sound absorbing material to the cavity were investigated during the series. The series produced a reasonably well-behaved set of data even at the lower frequencies. As expected, increasing the depth of the cavity and adding sound absorbing material moved the mass-air-mass resonance to lower frequencies and usually improved the sound transmission class rating. The highest sound transmission class measured was 73. Because of the extended range of the measurements, reductions in the low-frequency transmission loss values were quite evident. A simple method of predicting the results obtained will be presented and its extension to other cases discussed.
Musical pitch tracking based on a pattern recognition algorithm
Authors: Brown Judith C.
Published: October 2005

It can be shown that, when the Fourier transform is plotted against log frequency, the pattern obtained in the frequency domain is the same for any sound with harmonic frequency components. A constant Q Fourier transform has been calculated for frequencies corresponding to a musical quarter tone and plotted against log frequency. The resulting pattern can be “tracked” using cross correlation to give the musical pitch. Excellent results have been obtained for a variety of instruments including the violin, flute, and piano. [Work supported by a Brachman Hoffman Fellowship from Wellesley College.]
Application of an acoustic noise synthesis technique in the design of a duct system
Authors: Prasad M. G. , Ananthapadmanabha T. V.
Published: October 2005

This paper discusses the application of acoustic noise synthesis (ANS) for the design of a duct system, specifically, a muffler system. ANS is based on well-known principles of speech synthesis using a source-filter model. The designer can use an ANS technique to study the influence of design parameters of the duct system on the perceptual quality of noise. This approach is useful in studies relating to perceptual loudness criteria and noise control. The model considered in this study is a planewave white Gaussian noise source, a simple expansion chamber-type muffler, and an open-end tail pipe. The system transfer function is obtained based on the transmission coefficient expression derived from the chain matrix. Impulse response of the system is obtained by the application of an inverse FFT. The impulse response is convolved with the source signal to obtain the system output in the time domain. The output fed through a digital-to-analog convertor and a loudspeaker yields the simulated acoustic noise. This simulation procedure can be applied for any given model for source-muffler termination. The advantage of acoustic noise synthesis is that the source-muffler interaction, muffler design changes, and the effects of boundary conditions can be simulated based on theoretical models.
Anomalies in backscattering on large-scale inhomogeneities
Authors: Schhemelev Yevgeny G.
Published: October 2005

The backscattered field is investigated under conditions such that the plane monochromatic wave falls on a layer of chaotic statistically isotropic inhomogeneities. It is shown that the main part of the received signal comes from a specific direction & agr; (determined by geometry and scale), if the backscattering signal is received by a directed antenna and the condition d <l is realized (d is the receiving signal aperture, l is the characteristic scale of the inhomogeneities). This effect may be of use for determination of characteristic scales of large-scale inhomogeneities from analysis of the back-scattering signal.
Cross-frequency interactions in the precedence effect
Authors: Clifton R. K. , Zurek P. M. , Shinn-Cunningham B. G. , Durlach N. I.
Published: October 2005

The extent to which the precedence effect is observed when leading and lagging sounds occupy different spectral regions has been measured. Subjects, listening under headphones, were asked to match the lateral position of an acoustic pointer to that of a test stimulus composed of two binaural noise bursts with asynchronous onsets, parametrically varied frequency content, and different interaural delays. The precedence effect was measured by the degree to which the interaural delay of the matching pointer was independent of the interaural delay of the noise burst in the test stimulus. These results show an asymmetric frequency effect in which low frequencies dominate. For example, a stimulus composed of a high-frequency (1.25 kHz) leading burst and a low-frequency (0.45 kHz) lagging burst produces no precedence effect, whereas the reverse produces a strong precedence effect. These results are in qualitative agreement with those of Blauert and Divenyi [Acustica 66, 267–274 (1988)], despite numerous differences in stimuli and methods, including the restriction of this stimuli to the phase-sensitive (f < 1.4 kHz) region. [Work supported by NIH.]