bluuebunny/crossref_metadata_2025 huggingface dataset:
Perception of critical-band adjusted vowel continua by sensorineural hearing-impaired listeners
Authors: Espinoza-Varas B. , Jamieson D. , Wahn Judy
Published: October 2005

In front-back vowel continua cued by variations of F2, a critical-band (Z) representation of formant frequency was found to relate linearly to the judged “phonetic goodness” of the vowels: changes in F2 (ΔF2) which are equal along the Z scale, produce nearly equal changes of judged “phonetic-goodness” (ΔG) [Espinoza-Varas, J. Acoust. Soc. Am. Suppl. 1 74, S66 (1983)]. We replicated the previous experiments in sensorineural impaired listeners (low-frequency hearing loss, gradually sloping audiograms), who typically exhibit broadened critical bands. The vowel continua ranged from /u/ to /i/, /ɔ/ to /ɛ/, or /a/ to /æ/. On each continua the frequencies of F1 and F3 were fixed, and the frequency of F2 varied from the value appropriate for the back vowel to that appropriate for the front vowel in 6–12 steps of 0.5 Z each. F0 was 120 Hz. The 400-ms, steady-state vowels were generated by parallel synthesis; listeners categorized the stimuli and rated their “phonetic goodness.” A highly consistent deficit (across subjects and vowel pairs) in the impaired listeners' performance is the poorer ability to assess how good an examplar of a vowel class a given stimulus is. The relation between ΔF 2 and ΔG is no longer linear. This deficit is observed even though the ability to identify the vowels may be “normal.” The results appear to corroborate the possibility of a close relation between the judgment of “phonetic goodness” of vowels and the critical-band level of frequency analysis [Supported by AHFMR and NSERC.]
Ultrasonic measurements of pore concentration in aluminum cast materials
Authors: Adler Laszlo
Published: October 2005

An ultrasonic method was developed to measure the average pore size and volume fraction of porosity in aluminum silicon cast materials. The method is based on the measurements of backscattered amplitude of ultrasonic waves with frequencies from 2–20 MHz, evaluated in the frequency and time domain. The pore sizes range from 20 to 200μ and the volume fraction is less than 6%. Experimental results are compared to existing theoretical predictions. [This work was supported by the Center of NDE operated by Ames Laboratory for the Air Force Wright Aeronautical Materials Laboratory.]
Ultrasonic characterization of microstructure in powder metal alloy
Authors: Tittmann Bernhard R. , Ahlberg Lloyd A. , Fertig Kenneth
Published: October 2005

The ultrasonic wave propagation characteristics were measured for IN-100, a powder metallurgy alloy used for aircraft engine components. Attenuation, velocity, and backscattered microstructural noise were measured on several research samples. The data were obtained with a variety of different transducers placed either in a waterbath or in sample contact for a frequency range from about 2 to 18 MHz and statistically averaged over numerous volume elements of the samples. Micrographical examination provided size and number distributions for grain and pore structure. The results showed that the predominant source for the ultrasonic attenuation and backscatter was a dense (∼ 100/mm3) distribution of small micropores (∼ 10 μm radius). Two samples with different micropore densities were studied in detail to test the feasibility of calculating microstructural parameters from ultrasonic backscatter data in the regime for which the wavelength is much larger than the size of the individual scattering centers. The results suggest a way towards the nondestructive detection and characterization of anomalous distributions of micropores in a variety of materials when conventional ultrasonic imaging is difficult. The findings are potentially significant toward the application of the early detection of potential sites for stress-induced void coalescence leading to crack initiation and subsequent failure.
Optimal design of an electret earphone
Authors: Busch-Vishniac Ilene J.
Published: October 2005

Although foil electret microphones are widely available, their earphone counterparts are scarce. This is because of the low sensitivity of conventional electret earphones relative to electrodynamic or magnetic earphones. In the work presented here, an optimal design approach was used to maximize the sensitivity of a foil electret earphone while meeting constraints imposed by stability, frequency response, and low distortion. A generalized reduced gradient algorithm applied to leading order behavior of circular and annular electret segments indicates that an optimal earphone suitable for telephony should consist of a center circular segment surrounded by an annular segment. Optimal choice of parameters leads to a prediction of a sound pressure level of 103 dB for a 1-V rms drive. More accurate analysis indicates that this leading order estimate is somewhat high, but that it should be possible to produce a sound pressure level of at least 90 dB with a 1-V rms drive.
Transient response of volumetric sources and line array systems
Authors: Koenigs Paul D. , Stepanishen Peter R.
Published: October 2005

An impulse response approach to evaluate the transient response of volumetric sources and line arrays with beamformers is presented. A spatially and time-dependent impulse response function resulting from either a continuous or discrete distribution of sources with amplitude shading and arbitrary time delays is defined. The transient pressure at a field point is then obtained by convolving the time derivative of the impulse response with an excitation waveform. The characteristics of the impulse response for discrete steered and/or focused line array systems are first presented and related to the geometry of the array and beamformer. The characteristics of the impulse response for continuous line arrays or volumetric sources with spatially dependent amplitude and initial excitation times are then presented and related to the geometry of the source, speed of propagation within the array and media, and the location of the field point. Several applications and numerical results for volumetric sources and line array systems are presented.
Modeling predictions of the effect of bottom interaction on very-low-frequency (VLF,1<f⩽20 Hz) acoustic propagation
Authors: Schreiner Henry F. , Tunnell T. W.
Published: October 2005

The propagation of very-low-frequency [VLF, 20 Hz and less) acoustic signals in the deep ocean has been studied [A. N. Guthrie et al., J. Acoust. Soc. Am. 56, 58–59 (1974)] without taking bottom interaction into account. To investigate the effect of bottom interaction, implicit finite difference (IFD) techniques are used to solve the parabolic equation to study VLF (1 <f⩽20 Hz) acoustic propagation in continental shelf and slope environments. Fast field program (FFP) models are used to estimate what impact shear waves in the subbottom have on VLF propagation. Single interactions with the ocean bottom are studied with a plane wave multilayer (PWM) model. Normal mode calculations are used to interpret the IFD, FFP, and PWM results in terms of mode structure. Coherence and array signal gain are found to be high in regions where the density of IFD intensity contours is low. Numerical results for coherence, array signal gain, and transmission loss are presented. [Work supported by NORDA.]
A multi-source/receiver simulation computer program for studying acoustic fields
Authors: Seybert A. F. , Casey D. K.
Published: October 2005

In this paper we present some results of an interactive source/receiver computer program written for a personal computer. The program has been used to simulate simple acoustical environments and to study limitations of the sound intensity method. The program is also useful for showing students how source position, strength, and phase angle determine the acoustic field in a multi-source environment. The user can select interactively the characteristics of the sources, including sources on lines and sources on surfaces such as planes and spheres. Receivers may be selected individually by specifying their Cartesian coordinates; receivers may also be generated automatically to lie on certain simple surfaces such as planes, spheres or rectangular paralle-lopipeds. A pair of such receiver surfaces, separated by a fixed distance, can be generated automatically to study the sound intensity method. Examples of the above described features will be presented.
Teaching interior noise prediction using computers
Authors: Hurst C. J.
Published: October 2005

In most situations the prediction of noise levels inside rooms is complicated by several factors. First, the sound field may not have a truly uniform reverberant component. This is particularly true in many industrial environments where the ceiling height may be disproportionately low for the size of the floor plan. Secondly, most rooms have several sources present at once. Some sources may be large in extent, or may best be represented by a line sound source. A computer program, originally written for use in acoustic consulting work using a small computer, has been modified and used to help students explore the features of these sound fields, and to learn something about the efficacy of noise control measures such as absorption and barriers. The program will be described and the ideas behind its development discussed.
An interactive computer program for the acoustical analysis of small scale physical models of auditoria by architectural students
Authors: Siebein Gary W. , Battle John
Published: October 2005

A package of microcomputer software was developed to allow architectural design students to understand the acoustical ramifications of their design decisions. The package consists of a series of programs and instructional booklets to take architectural students (who typically have a very limited background in both acoustical and computers) step by step through the acoustical analysis of small-scale physical models using ultrasonic impulsive testing methods. Students build small-scale chipboard study models of a proposed auditorium design. The software package helps to lead them through the acoustical analysis of the design. Both graphic and tabular output of many objective acoustical design criteria are printed. Students optimize their schemes after several iterations are performed. The program was developed to be a computer-based worksheet so that the student could actually see the operations being performed by the computer. A minimal amount of computer experience (turning the machine on and off and inserting a disk) is required to operate the package. Comparisons between basic architectural design typologies (fan-shaped versus rectangular) of auditoria, the use of concert shells, and the configuration of the acoustical enclosure of the room can all be quickly evaluated by the students in model form. [Funded by NSF.]
Quantifying syntactic and semantic redundancy effects in the recognition of words in sentences
Authors: Nittrouer Susan , Boothroyd Arthur
Published: October 2005

The goal was to measure the contribution of semantic and syntactic constraints to the probability of word recognition in sentences. Normally hearing adult Ss listened to sets of four-word sentences at four S/N ratios. There were three types of sentences: (1) zero predictability, i.e., nonsense word strings; (2) low predictability, i.e., syntactically correct but semantically anomolous sentences; (3) high predictability, i.e., syntactically and semantically correct sentences. Word recognition probabilities were measured for each condition. The contribution of syntax was computed as the ratio of the logarithms of error probability in the low and zero predictability sentences. The contribution of semantics was computed as the ratio of the logarithms of error probability in high and low predictability sentences. The resulting parameter k is the amount by which the number of independent channels of acoustical information would need to be increased in order to produce the measured increase of recognition probability. Average values of k were 1.4 and 2.0 for syntax and semantics, respectively. Statistical analysis revealed that k was independent of S/N ratio. [Work supported by PSC. CUNY award 6-63137.]
Effects of spectral alternation upon intelligibility of words and sentences
Authors: Bashford James A. , Warren Richard M.
Published: October 2005

In the study of a novel speech interference paradigm, it was found that periodic alternation between nonoverlapping filtering conditions—1100 Hz low pass and 1700 Hz high pass—significantly reduced intelligibility. Thus, although repetition accuracy for CID sentences was approximately 98% in either the low-pass or high-pass filtering condition alone, subjects averaged only 48% correct when sentences were subjected to spectral alternation at a rate of 2.67 Hz. Interestingly, the intelligibility of spectrally alternated sentences was dramatically improved by the addition of appropriately filtered noise bands. Thus, when low-pass noise was mixed with segments of high-pass speech, and high-pass noise was mixed with segments of low-pass speech, repetition accuracy increased by 38.8%. A smaller but statistically significant increase was also found for isolated words. The intelligibility loss produced by spectral alternation appears related to other verbal and musical phenomena. The results from various control conditions indicate that the enhancement of intelligibility produced by alternating noise bands involved separate phonemic restorations for each band. [Work supported by NIH and NSF.]
Influence of speaking rate and lexical status on word identification
Authors: Miller Joanne L , Dexter Emily R. , Pickard Kimberly A.
Published: October 2005

At a recent meeting of this Society, we reported that two factors which contribute to the identification of a word within a sentence, the speaking rate at which the sentence is produced, and the semantic relation between the sentence and the target word, operate quite differently: Slight changes in task demands were found to readily eliminate the semantic effect, whereas the rate effect was obtained regardless of the specific task conditions used. These results suggested that in the course of identifying a word, the processing system optionally takes into account the semantic content of the context sentence, but necessarily takes into account its rate. In the present investigation, we have been asking whether a comparable distinction holds at the level of a single word. Specifically, we are comparing how the lexical status of a syllable (whether it is a real word or a nonsense word) and its speaking rate (specified by syllable duration) influence identification. As expected under the hypothesis that rate information plays an obligatory role during processing, we have found that identification is influenced by speaking rate under the variety of task conditions tested. We have also found, however, that the lexical status of the syllable influences identification under all the conditions tested. This result, if supported by additional tests currently underway, would suggest that lexical information, unlike word-external semantic information, may necessarily be accessed during processing and, like speaking rate, obligatorily influence identification. [Research supported by NIH.]
Stress centers are not perceived categorically
Authors: Cooper Andre M. , Fowler Carol A. , Whalen D. H.
Published: October 2005

The perceived onset of spoken stressed syllables (“stress centers”) is primarily sensitive to the duration and type of the initial consonants. To determine whether this phenomenon behaves categorically, three [sa]-[sta] continua were created by inserting silence between the /s/ noise and the vocalic section. In the first continuum, the overall syllable duration was allowed to vary as silence was added. In the second, the overall stimulus duration was held constant by excising compensatory amounts of frication from within the /s/ noise. In the final continuum, overall stimulus duration was held constant by excising compensatory amounts of the vocalic segment. While each of the continua was discriminated categorically, listeners' stress-center judgments were not categorical: the perceived stress center moved linearly toward stimulus offset as the overall duration of the prevocalic segment (frication + silence) increased. These results indicate that stress centers are not a post-perceptual illusion based on phonemic identity, but are instead sensitive to subphonemic acoustic/articulatory properties of the initial consonant(s). [Work supported by (NIH) HD-16591.]
Accuracy of doubly asymptotic approximations in acoustic scattering problems
Authors: Haberman Robert C. , Smith Preston W.
Published: October 2005

Numerical methods for predicting acoustic scattering from cylindrical shells generally consist of large finite element models of the shell coupled with finite element or integral representations of the fluid. In order to reduce the computational effort, the fluid can be represented by doubly asymptotic approximations [T. L. Geers and C. A. Felippa, J. Acoust. Soc. Am. 73, 1152–1159 (1983)]. The accuracy of first-and second-order DAA's is demonstrated for the two-dimensional scattering problem of a plane acoustic wave incident on a thin cylindrical shell with and without an attached oscillator. Quantities compared to the exact solutions are fluid modal impedance and farfield radiated pressure. By appropriately adjusting a parameter value in the second-order DAA, the impedance and pressure can be determined with reasonable accuracy. The first-order DAA is demonstrated to be not as accurate. The effect of the oscillator on the radiated pressure is also discussed.
Temporal resolution in monaural and binaural just-noticable phase changes
Authors: Zwicker E. , Henning G. B.
Published: October 2005

The just-noticeable (sinusoidal) amplitude modulation (JNAM) and the just-noticeable (sinusoidal) frequency modulation (JNFM) were measured by four subjects at 250 Hz and 60 dB SPL using the method of adjustment. For JNFM, two types of stimuli—monaural and binaural with unmodulated tone at the right ear—were presented through a matched pair of TDH-39 earphones driven in phase. The rate of modulation was either 1, 2, 4, 8, 16, 32, or 64 Hz. The results are plotted as a function of modulation rate and are expressed in terms of a modulation index which is similarly related to the amplitude of the first spectral side tones for both types of modulation. A comparison of the data allows the following conclusions: JNAM, JNFM (mon), and JNFM (bin) diverge strongly at low modulation rates but are indistinguishable at 32 Hz, in agreement with earlier monaural AM/FM data describing the limit of validity of Ohm's acoustical law. In the binaural situation, sensitivity to FM (which produces interaural time-delay modulation) remains high, up to unexpectedly large modulation rates (32 Hz). Individual differences in just-noticeable interaural time delay produce BMLD's of different magnitude which, however, still exist at modulation rates as large as 16 Hz; BMLD's appear to show temporal characteristics similar to those of normal thresholds.
Propagation of beluga whale echolocation signals
Authors: Au Whitlow W. L. , Penner Ralph H. , Turl Charles W.
Published: October 2005

The propagation characteristics of high-frequency echolocation signals (peak energies above 100 kHz) of the beluga whale, Delphinapterus leucas were determined by a series of measurements conducted while the animal performed a target-detection task. The whale was trained to station on a “bite-plate” so that its projected beam could be measured in the vertical and horizontal planes using a vertical and horizontal array of hydrophones, respectively. The transitional region between the acoustic near- and farfields was also located by using an array of hydrophones that extended directly in front of the animal, in the horizontal plane. Three distinct modes of signals were observed. Mode 1 signals had interclick intervals that were longer than the time required for the signals to travel to the target and back (two-way transit time). Mode 2 signals had interclick intervals shorter than the two-way transit time, and mode 3 signals had high-repetition rates with interclick intervals from 1–3 ms. The vertical and horizontal beam patterns of the mode 1 signals had similar 3-dB beamwidths of approximately 6.5°. The major axis of the vertical beam was directed at an angle of 5° above the plane defined by the animal's teeth. The near- to farfield transistion region was found to be approximately 0.88 to 1 m from the tip of the animal's mouth.
Sonar discrimination of metallic plate composition and thickness by dolphins and humans
Authors: Au Whitlow W. L. , Martin Douglas W.
Published: October 2005

Previous experiments have demonstrated that a blindfolded echolocating dolphin and humans using a diver sonar which projected a dolphin-like signal could discriminate the composition and thickness of metallic plates. No satisfactory explanation of the discrimination cues has yet been presented. In this study, the same plates were acoustically examined using a dolphin-like signal at various incident angles to determine cues present in the echoes. At normal incidence the echoes resembled the incident signal and did not seem to contain useful information for discrimination. However, at angles greater than about 10°, the echoes contained multiple highlights that could be used to discriminate the plates. These multi-highlight echoes were probably related to “leaky Lamb waves” and although they were between 20 and 30 dB lower in amplitude than the echoes at normal incidence, they could still provide useful discrimination cues. A human listening experiment was performed and the results were consistent with the presence of “leaky Lamb wave” echoes. The subjects could not discriminate the echoes received at normal incidence; however, the discrimination task was trivial for echoes obtained at a 14° incident angle.
Program for calculating acoustic fluctuations from internal waves
Authors: Flatte Stanley M.
Published: October 2005

A fast code was written to calculate the effects of a Garrett-Munk internal-wave field on fluctuations in acoustic transmission. Input parameters include acoustic frequency, source depth, ray launch angle, receiver range, and experimental profiles of sound-speed and buoyancy frequency, which are fit by cubic-spline techniques. Output includes ray geometry; time scales for phase and intensity fluctuations; vertical correlation length of the acoustic field; frequency decorrelation bandwidth of the field; descriptions of tails for pulse transmission; whether the ray is in the unsaturated, partially saturated, or fully saturated region; and whether the calculation is valid from the point of view of inhomogeneity and anisotropy effects.
Internal wave-induced azimuthal fluctuations in the acoustic field
Authors: Penland Cecile
Published: October 2005

In describing the effects of internal waves on acoustic normal modes excited by a time-harmonic point source, all previous descriptions have neglected the existence of azimuthal fluctuations in the acoustic field. This study compares this azimuthally symmetric case with the case where azimuthal fluctuations in the acoustic field are taken into account. Although azimuthal variations enhance the randomization of interference effects so that different depth modes decorrelate more quickly in range, this enhancement is generally small. The mean energy flux vector (j) becomes radial at long ranges; the vertical components of (j) decay somewhat more quickly with range when azimuthal fluctuations are considered than when they are neglected. [Work supported by Office of Naval Research.]
Currents and the parabolic approximation in underwater sound channels
Authors: Robertson John S. , Siegmann William L. , Jacobson Melvin J.
Published: October 2005

The effect of currents on the acoustic pressure field in an underwater channel is investigated. Based on the fundamental fluid equations, model equations are formulated which predict sound pressure while including nonuniform currents in the source-receiver plane. Application of parabolic-type approximations yields a collection of parabolic equations. Each of these is valid in a different domain determined by the magnitudes of current speed, current shear, and depth variation of sound speed. Under suitable conditions, it is useful to interpret current effects in terms of an “effective” sound speed. Using this effective sound speed in an existing numerical code, we examine propagation in a shallow isospeed channel with a simple shear flow and a lussy bottom. It is found that even small currents can induce very subtantial variations in relative intensity. The degree of variation depends upon current speed, source and receiver geometry, and acoustic frequency. Particular emphasis is placed on intensity-difference predictions in reciprocal sound transmissions in the presence of an ocean current. [Work supported by ONR.]
Estimation of geoacoustic parameters using towed array response to bottom-interacting ship noise
Authors: Kuperman W. A. , Werby M. F. , Gilbert K. E.
Published: October 2005

The response of a towed array to noise from its tow ship is simulated in a bottom-limited ocean environment. It is shown analytically and numerically that the beamformed response to tow-ship noise is related to the local properties of the ocean bottom through the depth-dependent Green's function. The transition region between the discrete and continuous parts of the Green's function is indicative of the acoustic properties of the bottom. The array size, tow distance, and element weighting affect the response of the array to this region. The optimum array configuration and processing scheme is identified through a numerical sensitivity study. Finally, to demonstrate the utility of the method, we apply it to several diverse bottom types.
Inverse techniques applied to the characterization of model rough surfaces
Authors: Bailey Ricky S.
Published: October 2005

This study represents an inverse problems approach to rough surface scattering, an objective being the inference of certain statistical characteristics of the rough surface directly from the coherently scattered pressure field. A combined theoretical and experimental approach is used. Eckart's physical-acoustic model is applied to derive relationships for the probability density function and correlation function of surface heights. The results are expressions which depend solely upon the scattered pressure field. Verification of the theory is accomplished through experiments conforming to the constraints outlined in the theory. Experimental measurements of the steady-state complex scattered pressure field from a pressure release model surface are made with use of a single-frequency small beamwidth source. Several signal processing techniques are used to obtain estimates of the probability density function and correlation function. Comparisons are then made between the inferred characteristics and the actual characteristics of the surface determined by direct measurement techniques. [Work supported by the Office of Naval Research.]
Limitations on ray-path identification due to topographic features with application to the Straits of Florida
Authors: Seem D. A. , Lawson L. M. , Palmer D. R.
Published: October 2005

The Florida Current, which flows northward through the narrow Straits of Florida, transports a significant fraction of the heat which must be transferred from low latitudes to polar regions in order for the Earth to remain in overall radiative balance. Ocean acoustic tomography is one of the remote sensing technologies being considered for monitoring this transport. Tomography, as developed by Munk and Wunsch, is dependent, however, on being able to resolve and identify individual ray paths. We have investigated the influence of the topographic features present in the Straits on ray-path identification. Using a fast, specialized ray-tracing program we calculated the fluctuations in ray-path geometry resulting from small random perturbations in the bathymetry. We found that the topographic features amplify these fluctuations to such a degree that identification of individual rays becomes difficult, if not impossible, after only a few bottom bounces. These results may have relevance to other environments where rays encounter range-dependent boundaries such as propagation under ice or above a mid-ocean ridge.
Horizontally random bottom structure and acoustic intensity variations in a shallow ocean
Authors: Ashley C. E. , Jacobson M. J. , Siegmann W. L.
Published: October 2005

The effects of a model for random bottom structure on acoustic intensity in isospeed shallow water are studied. The randomness results from stochastic variations in bottom density and sound speed in the horizontal direction beneath a plane water-bottom interface. Ray geometry, spreading loss, and bottom loss and phase shift are examined in order to derive formulas for intensity mean and variance. The expressions obtained are sufficiently general to permit their use with different bottom-acoustic models of sound reflection. For illustrative and comparative purposes two such models, those of Mackenzie and Rayleigh, are considered. The distinctive acoustic consequences of bottoms of different density mean, variance, and horizontal correlation are discussed, as are comparisons of results for the two bottom-reflection models. Intensity moments are obtained for differing source-receiver range and water depth. [Work supported by ONR.]
Labial movement dynamics for speech
Authors: Caligiuri Michael P. , Abbs James H.
Published: October 2005

Interpreting central nervous system control of speech movement depends in part on understanding the detailed parameters of that movement. For example, the motor control significance of the often-reported reflex responses elicited during speech depends upon whether the external eliciting stimuli generate mechanoreceptor activity that is comparable to that generated naturally by speech movements [cf., Caligiuri and Abbs, Society for Neurosciences (1984)]. Similarly, kinematic analyses of articulatory movement confined to average velocity without considering instantaneous velocity and acceleration are likely to be misleading insofar as the critical neuromotor actions are likely to be misinterpreted. For these reasons the labial dynamics for speech were investigated in five young adults. The maximum peak acceleration and velocity obtained during normal speech production were 544.9 cm/s2 and 39.7 cm/s, respectively, with means of 260.7 cm/s2 and 25.5 cm/s. Further analyses of the effects of increasing speaking rate revealed that peak velocity and acceleration were not scaled proportionately. That is, for the labial closing gesture, peak acceleration increased up to 47%, while peak velocity increased only 12%. These and related findings underscore the importance of considering peak acceleration as well as peak velocity when assessing the motor control of speech movements. [Research supported by NS-13274 and NS-18797.]
Channel reverberation in shallow water
Authors: Herwig Heinz , Nuetzel Bernd
Published: October 2005

The results of experiments designed to measure the characteristics of channel reverberation in shallow water (30-m water depth) are presented along with supporting environmental data. The acoustic data were obtained using long continuous wave pulses with a monostatic transmitter and receiver system from a fixed platform in the North Sea. The characteristic values of channel reverberation such as frequency shift, linewidth, and reverberation time are described as a function of frequency (500 Hz to 10 kHz) and the wind-induced surface roughness. It is also shown for a more limited set of environmental data that channel reverberation is dependent on the angle between the acoustic propagation and the direction of the surface waves. The linewidth and the reverberation decay time show a strong dependence on frequency and surface roughness. These dependencies are presented for various sets of environmental parameters. [Work supported by MOD. West Germany.]
Experimental study of laser-induced underwater sound
Authors: Maccabee Bruce S. , Bell Charles E.
Published: October 2005

Experiments have been carried out to characterize the sound generated when a CO2 “TEA” (pulsed) laser, operated without nitrogen, illuminates a free water surface. Energy densities ranging from below to above the value needed for rapid surface evaporation have been used. Acoustic pulse risetimes, as measured with very broadband detectors, are typically less than 100 ns at shallow depths. The acoustic pulse shape changes during propagation. The exact nature of the change depends upon the laser energy density. Generally a pulse stretches and the trailing edge steepens so that it changes from a nearly sinusoidal shape into an “N” shape, often with the rarefaction having a larger amplitude than the initial compression by the time it has traveled downward 5 cm from the surface. Although nonuniformities (“hot spots”) in the laser intensity make it difficult to determine the exact relation, it appears that the peak pressures are roughly proportional to the energy density (J/cm2) of the laser pulse. An approximate energy density-to-pressure conversion constant for pressures measured at a depth in meters is 0.1 (m) (bar)/(J/cm2). The paper will provide an overview of these and other experimental results.
Optically stimulated sound from gas bubbles in water: A novel optoacoustic mechanism
Authors: Unger Bruce T. , Marston Philip L.
Published: October 2005

We detected high-frequency sound radiated by individual gas bubbles in response to modulated green light from an argon-ion laser. The bubbles were in clear water and were attached to a needle. Their radii were typically less than 100 μm. The sound radiated in response to single light pulses, of duration ≃10 μs and power ≃2 W, exhibits the ringing of the bubble's monopole resonance. The resonance frequency fR was found by Fourier transforming this signature; typical values were 30 kHz. Bubbles were subsequently illuminated by bursts of pulses with a frequency f≃fR. The resulting sound was characterized by an initial buildup of its amplitude. The mechanism for driving the monopole oscillations appears to be the modulated optical radiation pressure on the bubble's surface rather than thermal expansion. Related experiments on the optical levitation of bubbles in water and the emission of sound from dyed drops in water will be discussed. [Work supported by ONR.]
“Representing” inherently variable quantities: The artificial mastoid and other acoustical calibrators
Authors: Corliss Edith L. R.
Published: October 2005

Because human beings vary widely in their physical attributes, one often must describe a representative individual for the purpose of calibration and specifications. A common procedure is to summarize the median or mean at specific audiofrequencies of determinations made at those frequencies. Less commonly, parameters for each individual are derived, and median parameters taken for the group. The choice of alternative reduction processes yields significant differences. This is illustrated by reducing the same measured data by both alternative procedures. At least, the load the human head presents to a bone vibrator has a specified plane of contact, and can be represented by lumped parameters. However, similar problems exist for representations of the external ear and ear canal, where the load approximates a transmission line. Diffraction patterns about the ear, head, and body present comparable problems. Feasible processes for reducing the systematic errors will be discussed. [Access to National Bureau of Standards data and statistical advice is acknowledged with thanks]
Effect of measurement parameters on acoustic reflex thresholds
Authors: Robinette Larry N. , Thompson David J.
Published: October 2005

Estimates of acoustic reflex thresholds (ARTs) may differ by 5 dB or more between lowest (“best”) and highest reported values. These discrepancies are likely due to differences in measurement characteristics (sensitivity, probe tone level, display mode, and temporal response) of acoustic immittance instruments in addition to criteria used for judgment of threshold and normal threshold variability. Continued development of ART measurement methods for clinical evaluation is hampered by such disparities. We therefore studied the effects of selected measurement parameters on ARTs. Contralateral acoustic-reflex thresholds were measured in ten subjects with normal hearing sensitivity using a digital acoustic-immittance instrument that allowed discrete changes in measurement parameters. In the first experiment, probe microphone sensitivity was varied in four steps from low to high values, in the second experiment probe tone (226 Hz) level was varied from 72 to 84 dB in 4-dB steps, and in the third experiment effective measurement time constant was varied from 8.8 to 141.6 ms in 8.8-ms steps. Three independent judgments of ARTs were made from plots of responses to a 1000-Hz tone and broadband noise and means of the judgments were used for analysis. Results indicate that instrument measurement parameters can affect ARTs and should be accounted for. [Research supported by Rehabilitation Research and Development Service of the Veterans Administration.]
Open-loop transfer function simulation as a means for understanding acoustic feedback in hearing aids
Authors: Egolf David P. , Haley Brett T.
Published: October 2005

The “squeal” of unstable acoustic feedback in hearing aids occurs when Nyquist's stability criterion involving the system's open-loop transfer function (OLTF) has been violated. To understand the causes and possible remedies of unstable feedback in hearing aids, one must first investigate the OLTF of such a system. Reported herein is a scheme for simulating, on the computer, the OLTF of an eyeglass-type hearing aid. This scheme consists of taking the serial product of two-port network models of the various components which transduce, amplify, and transmit sound to the eardrum. These components are (in order) the microphone, amplifier, receiver, sound tubes leading to the eardrum, earmold vent, and external pathway from the vent exit back to the microphone. Computer simulations were compared with laboratory measurements of the OLTF. These were obtained by applying an electrical signal to the amplifier input and simultaneously monitoring electrical output of the hearing-aid microphone. Computer results and experimental data followed the same trends but differed by as much as 6 dB at one frequency.
A perceptual voice taxonomy and its uses in evaluating speaker recognizability
Authors: Voiers William D.
Published: October 2005

The development of practical methods of evaluating speaker recognizability in processed speech has been hampered by the lack of an adequate system, either physical-acoustical or perceptual, for classifying voices. Previous efforts to develop taxonomies based on perceived voice traits (PVT) failed for a variety of reasons, including inappropriate scaling procedures and grossly inadequate voice samples. In the present attempt to remedy the situation, more than 500 potential voice descriptors were experimentally screened to yield a reduced set of rating scales with which listeners could reliably describe their perceptions of individual voices. Factor analysis of rating data for a sample of eighty voices revealed at least eight orthogonal PVTs, several of which had not been previously described. Methods of evaluating the effects of speech processing on speaker recognizability from its effects on the reliability and factorial structure of voice rating data are discussed and illustrated.
Objective estimation of perceptually specific subjective qualities
Authors: Quackenbush S. R. , Barnwell T. P.
Published: October 2005

A different approach to the design of objective measures of speech quality is presented: Objective estimates of the parametric, or perceptually specific, qualities of the Diagnostic Acceptability Measure (DAM) subjective quality test are used to construct more accurate objective measures of speech acceptability. Previous research [T. P. Barnwell III and S. R. Quackenbush, IEEE ICASSP, pp. 996–999 (1982)] has shown that many widely used objective measures produce poor estimates of speech acceptability when used over a variety of speech distortions. A multidimensional scaling analysis suggests that the reason for this is that the objective measures incorporate little of the speech quality information provided by the parametric quality scales of the DAM. Multiple linear regression is used to show that a subset of the parametric qualities can be combined in a linear model to provide very good estimates of speech acceptability. To this end, objective measures which provide dramatically improved estimates of the same subset of parametric qualities are described. The narrower scope of these parametric subjective qualities promotes a closer coupling between physical quantities and perceptual qualities of the given quality category, resulting in more accurate objective measures. Multidimensional scaling is used to graphically show the remarkable improvement in the ability of these new measures to estimate the parametric subjective qualities.
Intelligibility of speech and its relationship to the speech spectrum—French and Steinberg revisited
Authors: Englert Sue Ellen , Egolf David P.
Published: October 2005

Knowledge of the relationship between the speech spectrum and intelligibility of speech is a key factor in being able to maximize intelligibility for the hearing-impaired listener and for the normal listener under such adverse conditions as bandwidth attenuation or masking by noise. The present experiment was designed to allow the speech spectrum to pass, binaurally, to a normal listener, with an optimally flat response of the electroacoustic system. A real-time programmable digital filter was then used to allow only certain bandwidths of the spectrum to be heard by the listener. The bandwidth locations were chosen to include the formants of the talker's voice. The experimental results were then compared to the classic results of French and Steinberg. It was observed that high-pass and low-pass filter tests, generally, agreed with French and Steinberg's observations; however, it was also found that bandpass filter tests do not agree since comparison to the older experimental results is inappropriate. From these new results, a revised analytical procedure for intelligibility versus the speech spectrum is being developed.
Central representation and categorical perception
Authors: Daly D. M. , Daly D. D. , Drane J. W.
Published: October 2005

We report results of two persons with unusual differences in ability to classify standard sets of synthetic acoustic stimuli (SAS), findings perhaps analogous to extraordinary vibrotactile observers [J. C. Craig, Science 196, 450 (1977)]. Case 1, a 33-year-old woman, had normal hearing bilaterally; case 2, a 33-year-old man, had mild high-frequency hearing loss bilaterally. Sets of SAS were presented monaurally to each ear; responses were arranged in multidimensional contingency tables and analyzed for departures from homogeneity. Each classified /b/ vs /d/ vs /g/ consistently and appropriately (re: 45 normals). Case 1 could consistently classify /g/ vs /y/ to right ear (AD) as one of three sounds (g colored, an intermediate, and y colored), and to left ear (AS) as one of six sounds (four g colored, two y colored); AD classes were not composed of AS classes. Case 2 could consistently classify /g/ vs /y/ to either ear as one of five sounds (vocalic, two g colored, two y colored); AS and AD classes were equivalent. Case 2 could also consistently classify /b/ vs /w/ as one of five sounds (vocalic, two b colored, two w colored); however, AS and AD differed significantly between longer b and shorter w boundaries. These results require fine grained differences in rate or duration of basilar membrane movements be represented in auditory cortex.
The application of a hierarchical classification technique to speech analysis
Authors: Randolph Mark A.
Published: October 2005

In this paper we describe a hierarchical classification technique based on CART (Classification and Regression Trees, see Breiman et al., 1984) and its application to the task of speech analysis. Our investigation is motivated by two reasons. First, we believe that this technique provides an intuitive mechanism for quantifying the acoustic cues of phonetic contrasts. Second, the technique can potentially help us develop classifiers that are useful for automatic speech recognition. Towards these goals, we have added a number of features to the basic CART algorithm, and have expanded it into an exploratory data analysis tool. For example, CART uses a predetermined criterion for partitioning the feature space. We have added the capability for users to manually perform this partitioning. In addition, we have implemented a number of statistical functions for univariate and multivariate data analysis, and added graphical facilities for viewing data in different ways. Most importantly, we have integrated CART with SPIRE, our primary acoustic-phonetic analysis tool. Examples of how CART can be used for speech analysis and classifier design will be presented. Comparisons with other classification procedures will also be included. [Work supported by AT&T Bell Laboratories Cooperative Research Fellowship and by the Office of Naval Research under contract N00014-82-K-0727.]
Speaker characteristics influencing the intelligiblity of resynthesized LPC speech
Authors: Buiting Harry , Boves Louis
Published: October 2005

In applying analysis resynthesis techniques, much attention must be paid to the choice of the speaker, especially when a so-called speech-chip is used in the resynthesis process. To find an answer to a number of questions concerning this issue a perception experiment was carried out using ten Dutch speakers, five males and five females. The speakers read 50 phonetically balanced CVC nonsense words, which were embedded in carrier sentences. The 50×50 = 500 CVC words were analyzed by means of a linear predictive procedure which resulted in energy, pitch, and formant/bandwidth data. For a small subset of the speech material the analysis data were optimized by means of both visual and auditory inspection. This optimization process resulted in a set of rules to be applied to the analysis data before resynthesis. The optimization rules were used to resynthesize the CVC words with several bit rates. The resynthesized words made up the stimulus ensemble in a recognition experiment. Questions we address are:—What is the effect of the bit rate on the identifiability?—Is it possible to isolate criteria that “good” speakers should meet?—How successfully can rules established for a small set of words be applied to a larger set?
Mixed excitation for speech II: Naturalness of period-by-period reordered voiced fricatives
Authors: Allen George D. , Jamieson Leah H.
Published: October 2005

Previous attempts at synthesizing voiced fricatives have failed to yield acceptably natural-sounding segments. One possible reason for this failure is that the synthesis models have been global, whereas some important characteristics of these segments may reside in their period-by-period structure. Our goal in this study was therefore to compare the auditory quality of digitally manipulated /v, ð, z, and ȝ/ segments, in various V-V contexts. High-quality tokens were digitized at 24 kHz, and individual pitch periods were marked by hand using interactive software. Comparison stimuli were then created via the following manipulations: (1) reordering of alternate periods; (2) reordering triples of periods; (3) replacement of all odd- (even-) numbered periods by their even- (odd-) numbered neighbors; (4) similar replacements modulo 3; (5) random reordering of periods. Furthermore, these manipulations were sometimes restricted to the onset, steady state, or offset portions of the segments. These digitally spliced segments were then presented to listeners for discrimination, naturalness, and likeness judgments. Results of these comparisons and their implications for synthesis of natural-sounding voiced fricatives will be discussed.