bluuebunny/crossref_metadata_2025 huggingface dataset:
Speaker adaptation in articulatory speech analysis by synthesis
Authors: Schroeter J. , Larar J. N. , Sondhi M. M.
Published: October 2005

Using an articulatory approach for low-bit rate speech coding [J. Schroeter, J. Larar, and M. M. Sondhi, J. Acoust. Soc. Am. Suppl. 1 80, S19 (1986); Proc. IEEE ICASSP 308–311 (1987)], means for adapting an analysis/synthesis scheme to different speakers was investigated. Since a crucial feature for this approach to speech coding is the use of an articulatory codebook of tract shapes and related transfer functions [M. M. Sondhi, J. Schroeter, and J. N. Larar, J. Acoust. Soc. Am. Suppl. 1 81, S78 (1987); J. N. Larar, J. Schroeter, and M. M. Sondhi, J. Acoust. Soc. Am. Suppl. 1 82, S54 (1987)], the codebook entries have to be modified to accommodate different nominal vocal-tract sizes. Two different ways of speaker adaptation are compared. In the first approach, the codebook is comprised of several subcodebooks, each for a different size of the vocal tract. In this case, a method for determining the optimal subcodebook for each speaker is needed. Alternatively, in the second approach, the LPC-representation of the original speech is scaled for access to a constant tract-size codebook. Then, the codebook entry found is inversely scaled for synthesis. Here, a method for determining the optimal scale factor is required.
The effects of pedestal level on incremental auditory responses in humans: Brainstem response (ABR) study
Authors: Riquimaroux Hiroshi , Dallos Peter
Published: October 2005

The effects of pedestal level (LB) on incremental auditory responses were examined in normal hearing human adults with amplitude modulated (AM) white noise by using the auditory brainstem response (ABR). The wave V latency of ABR, yielded by a noise increment (ΔL) with various LBs (30–80 dB SPL), was measured. The hypothesis of this study is that the wave V latency derived by ΔL is invariant with LB. The results show that the wave V latency appears to be almost independent of LB for very small ΔL (⩽ 6 dB). The data are compared with those of the companion loudness matching study. Similar tendencies are seen in the characteristics of iso-wave V latency contours and equal loudness contours for relatively small ΔLs or for low LBs. The growth functions obtained from the two experiments, however, are different. In the loudness matching study the slope of loudness growth functions increases as LB increases, while in the ABR study the slope of growth functions of wave V latency increases only for low LB.
Proteins of the organ of Corti (OC)
Authors: Thalmann Isolde , Thallinger Gertraud , Thalmann Ruediger
Published: October 2005

The majority of proteins found to date in OC were identified by immunohistological methods. While these techniques are powerful tools for localization of proteins, they frequently lack satisfactory specificity. Two-dimensional polyacrylamide gel electrophoresis (2D-PAGE) is able to demonstrate the presence in the OC of specific subtypes of certain proteins. In the case of contraction associated proteins, for instance, the presence of nonmuscle actin (beta and gamma), three types of nonmuscle tropomyosin (Tm:4, Tin:5, and Tm:8) and nonmuscle lactic dehydrogenase (LDH B) was documented. Another important role of 2D-PAGE is the demonstration of previously unknown proteins. The presence in the OC of two highly prominent proteins of low molecular weight and strongly acidic pH (assumed to be specific to the OC) was shown. Accordingly, they were termed OCP-I and OCP-II [I. Thalmann et al., Arch. Otorhinolaryngol. 226, 123–128 (1980)]. It was found that small amounts of OCP-I and OCP-II are present in spiral ligament and spiral limbus and more substantial amounts in basilar membrane. Neither protein is detectable in stria vascularis. Likewise, no proteins resembling OCP-I or OCP-II were found in tissues other than inner ear. While some biochemical features of the two proteins have in the meantime been characterized, their functional significance remains obscure. [Work supported by NIH.]
Lateral suppression and auditory frequency selectivity
Authors: Dreschler Wouter A. , Leeuw A. Rens
Published: October 2005

For measuring frequency selectivity, the notched-noise procedure, introduced by Patterson and Nimmo-Smith, is regarded at the moment as the best method to avoid cueing effects and to take off-frequency listening into account. In this study, a similar approach was adopted for the measurement of lateral suppression. The amount of suppression for a two-tone masker, as a function of the frequency of the suppressor, was measured in the presence of a notched noise. Also in this paradigm, a fair amount of suppression was found in a group of 15 normal-hearing subjects. For hearing-impaired subjects, and, surprisingly, also for subjects with a conductive loss, a gradual loss of suppression as a function of hearing loss was observed. Finally, the results are related to auditory filter shapes obtained in simultaneous and forward masking.
Frequency masking patterns and “excess masking” in normal-hearing and hearing-impaired listeners
Authors: Garrison Margery A. , Schroder Anna C. , Nelson David A.
Published: October 2005

The presence of “excessive upward spread of masking” in high-frequency sensorineural hearing-loss (HFSNHL) subjects has been previously demonstrated by other investigators. This excess masking, defined by predictions from a “noise” model of hearing loss, occurs in the transition region between normal and impaired hearing. To examine this phenomenon further, frequency masking patterns (FMPs) were obtained from two normal-hearing subjects and one HFSNHL subject. Using a 200-Hz narrow-band noise (NBN) masker (90 dB SPL) with an upper edge at 520 Hz, FMPs from the impaired ear demonstrated excess masking, which it was hypothesized might be explained by an inability to “listen in the valleys” of the masker envelope because of poor temporal resolution. To test this notion, the experiment was repeated using a 40-Hz NBN masker (90 dB SPL), which has less rapid envelope fluctuations. No differences in the upward spread of masking were seen between the 200- and 40-Hz bandwidth-equal SPL maskers. When the experiment was repeated with equal spectrum level maskers, the 40-Hz NBN masker produced less upward spread of masking in both normal-hearing and hearing-impaired ears. These data suggest that, for the HFSNHL subject, excessive upward spread of masking cannot be completely explained by an inability to “listen in the valleys.” [Work supported by NINCDS.]
High-frequency audiometry in platinum ototoxicity
Authors: Dreschler Wouter A. , Hulst Rob J. A. M. v/d
Published: October 2005

In a group of 144 subjects (288 ears) medically treated with platinum drugs, high-frequency audiometry was applied at frequent intervals in order to compare the effects of three different drug administrations: high-dose cis-platinum cures, low-dose cis-platinum cures, and carboplatin cures. In all subgroups, high-frequency audiometry considerably enhanced the early detection of ototoxicity. Marked differences between cures were established, both in the pattern of onset of the damage and in the relation between dose and damage severity. For subjects treated with platinum derivatives, especially, the thresholds at 14 kHz prove to be important. The results suggest that, for these subjects, a single measuring frequency should be taken into consideration. With a minimum of effort, most of the increased sensitivity for a complete high-frequency audiogram can be reached. Finally, the predictive value of a threshold deterioration above 8 kHz for a threshold deterioration in the conventional range of audiometric frequencies will be considered.
Sinusoidal modeling of speech: Its use in simulating and compensating for recruitment of loudness
Authors: Lias Janet R. , Clements Mark A.
Published: October 2005

Amplitude compression techniques utilizing both single and multi-channel filtering systems have been proposed for use in compensating for recruitment of loudness in hearing aids. Due to a variety of limitations, these techniques have shown only marginal success. The sinusoidal model of speech developed by Quatieri and McAulay [IEEE Trans. Acoust. Speech Signal Process. ASSP-34, 744–754 (1986)] has been used to simulate and compensate for recruitment, using the general theory described by Villchur [J. Acoust. Soc. Am. 56, 1601–1611 (1974)] and others. Specifically, the model synthesizes speech as the sum of sinusoids with time-varying amplitudes and phases. Amplitude compression and amplification can be performed on individual sinusoids to shape speech to fit an impaired person's residual hearing. Conversely, expansion can be used to Simulate the effects of recruitment. This model offers almost unlimited flexibility in areas such as consonant boosting and spectral shaping in any portion of the spectrum. With a real-time implementation of this model, the parameters can be easily adjusted to suit each person's needs.
Effects of fluctuating conductive hearing loss in infancy on auditory brain-stem responses and binaural interaction measured at age 5
Authors: Gunnarson Adele , Finitzo Terese
Published: October 2005

The purpose of this research was to examine long-term auditory electrophysiologic effects of early conductive hearing loss. Animal deprivation studies suggest brain-stem anatomic and physiologic alteration during critical periods. Comparable human research is limited. The research questioned whether auditory brain-stem responses (ABR) and binaural interaction (BI) at ages 5–7 differed depending on documented patterns of early hearing. Stimuli were rarefaction clicks at 21.1/s, presented binaurally at 60 dB and monaurally at 30, 60, and 80 dB HL. The BI was derived by subtracting summed ipsilateral and contralateral monaural responses from binaural responses, and defined as deflections whose amplitude exceeded 3 s.d. from derived prestimulus amplitudes. Results indicated that waves III and V, and I–III and I–V interpeak latencies (IPL) differed when controls were compared to experimental subjects. Wave I, the latency/intensity functions at 30–60 and 30–80 dB, and III–V IPLs were unchanged. The BI was present in 8 out of 9 controls and 8 out of 9 children with fluctuating mild losses in infancy. Only 4 out of 9 children with moderate asymmetric losses in infancy had BI at age 5. [Work supported by NIH No. 5 ROI NS 19675-03.]
An accurate numerical solution to ASA Benchmark Problem I using the finite element method
Authors: Murphy Joseph E. , Chin-Bing Stanley A.
Published: October 2005

A Special Session on Numerical Solutions of Two Benchmark Problems was presented at the 113th Meeting of the Acoustical Society of America in Indianapolis, Indiana, 11-15 May 1987. The underwater acoustics community was invited to present solutions to these problems and compare them against a numerically “exact” solution obtained from the R. B. Evans' coupled mode model [F. B. Jensen, J. Acoust. Soc. Am. Suppl. 1 81, S40 (1987)]. Our finite element ocean acoustic propagation model [J. E. Murphy and S. A. Chin-Bing, J. Acoust. Soc. Am. Suppl. 1 81, S9 (1987)] to Problem I is applied here. (Problem I dealt with shallow water upslope propagation in a wedge-shaped channel.) This finite element model gives a full-wave solution for range-dependent acoustic propagation with accuracy that rivals coupled mode models. The model's capability will be demonstrated by presenting a numerical solution to Problem I that compares favorably to the Evans' coupled mode model solution. [Work supported by ONR/NORDA.]
Sound transmission experiments from an impulsive source near rigid wedges
Authors: Li Saimu , Clay C. S.
Published: October 2005

Experimental sound transmissions in air from a spark source to a small microphone were made near rigid wedges. Two types of experiments were made. The first experiments were transmissions within an approximately 12° wedge and a 52° wedge. The Biot-Tolstoy wedge solution [I. Tolstoy, Wave Propagation (McGraw-Hill, New York, 1973)] was used to calculate the theoretical impulse response. The “free air” transmission from the spark source was convolved with the theoretical transmission. The transmissions within the wedge gave finite sets of multiple reflections or image arrivals and a diffraction arrival. The diffraction was sensitive to leaks at the wedge apex. Theory and experiment matched. The second set of experiments was made near a 270° wedge. Arrivals were the direct reflection and the diffraction. The image reflection when the specular “reflection point” was very near the wedge apex was of interest. Comparisons of data and theoretical signals using Biot-Tolstoy theory were excellent. Theoretical diffraction signals calculated with Trorey's theory had poor matches [A. W. Trorey, Geophysics 35, 762–784 (1970)].
Impulse response of density contrast wedge using normal coordinates
Authors: Chu D. , Clay C. S.
Published: October 2005

The exact impulse response of a point source to a rigid wedge was derived by Biot and Tolstoy [I. Tolstoy, Wave Propagation (McGraw-Hill, New York, 1973)]. The solution by the experiments has been verified. The wedge having free (pressure release) boundaries has the same form as the rigid boundary solution [W. A. Kinney et al., J. Acoust. Soc. Am. 73, 183–194 (1983)]. These solutions cannot be applied directly to many actual cases such as ocean floor, where the ratio of density of the base sediment to that of the water approaches neither infinite (rigid) nor zero (pressure release). The solution is the impulse response to an isovelocity (v1 = v2) and density contrast (ρ1≠ρ) wedge. It is an extension of the solution given by Biot and Tolstoy. The reflection part of the solution is an impulse series weighted by reflection coefficient with different order. The diffraction part is the summation of the diffractions due to the individual image sources. The amplitudes are a function of the reflection coefficient and the number of multiple reflections. [This work was partly supported by the People's Republic of China and the Office of Naval Research.]
Estimation of surface noise source level from low-frequency seismoacoustic ambient noise measurements
Authors: Schmidt Henrik , Kuperman W. A.
Published: October 2005

The extraction of noise source levels from ambient noise measurements requires accounting for the seismoacoustic propagation from the surface generated noise sources to the field measurement positions. At frequencies below 100 Hz, the waveguide nature of the environment strongly influences the distribution of ambient noise. Only when these propagation effects are considered can measurements from different experiments taken in different environments be combined to ascertain the global properties of surface noise sources. Here, such a previous analysis of a shallow water experiment [Schmidt et al., in Natural Mechanisms of Surface Generated Noise, edited by B. Kerman (Reidel, Dordrecht, The Netherlands, 1988)] is used and the results are combined with Kibble-white and Ewans' lower-frequency results [J. Acoust. Soc. Am. 78, 981– 994 (1985)], the latter also containing a summary of previous experimental results obtained by others. These data are treated in the same way by accounting for the environments using a wave theory model of distributed noise [W. A. Kuperman and F. Ingenito, J. Acoust. Soc. Am. 67, 1988– 1996 (1980)]. When the particular environments are used, the spread in the reprocessed noise levels is substantially reduced, showing more consistency in the frequency dependence of the noise sources, suggesting that there are only a few (or possibly only one) dominant natural noise source mechanisms that primarily contribute to ocean ambient noise below 100 Hz. The importance of appropriately including propagation factors in processing noise data for estimating source levels is therefore strongly demonstrated.
Boundary element solution for a coupled elastodynamic and wave equation system to predict forced response of a plugged acoustic cavity
Authors: Ciskowski Robert , Royster Larry H.
Published: October 2005

The subject of this investigation is the use of the boundary element method (BEM) to predict the steady-state and transient response of a plugged acoustic cavity to external excitation. The plug is an elastodynamic material subject to external excitation. The excitation of the plug is transmitted to the acoustic cavity. No internal sources of excitation are considered. A three-dimensional BEM model is developed for the analysis, both for the fluid in the cavity and the elastodynamic material in contact with it. Transient response is determined by recovering the time domain response from the Laplace transform domain solution using an FFT-based inversion scheme. Steady-state response is obtained from the Laplace transform domain solution by setting s = − iw. The numerical code is designed to run on the IBM family of personal computers and workstations.
Plane-wave excitation of an infinite circular cylinder reinforced by a periodic set of rings
Authors: De Maigret Francois
Published: October 2005

Analytical expressions are derived for the scattered field by an infinite fluid-loaded cylindrical shell reinforced with the periodic set of rings and acoustically excited by an incident harmonic plane wave. Vibrations of the shell are treated using thin bending cylindrical shell theory by Kennard. The stiffeners interact with the cylindrical shell only through normal forces. Unknown quantities (i.e., scattered acoustic field, displacements of the shell) are expanded on the natural Bloch-Floquet basis induced by the structure periodicity. Numerical simulations on a wide range frequency are reported, where the dimensionless product “ka” of the incident fluid wavenumber by the shell radius typically extends from 10−2 to 10−2, using a unique algorithm. Numerical predictions are compared with experimental data on different physical situations showing a good agreement. [Work supported by GERDSM.]
Prey detection by means of passive listening in bottlenose dolphins (Tursiops truncatus)
Authors: Burros Nélio B. , Myrberg Arthur A.
Published: October 2005

Free-living bottlenose dolphins (Tursiops truncatus) are primarily piscivorous and the use of active echolocation has been implicated in prey capture [K. S. Norris and B. Møhl, Am. Nat. 122, 85–113 (1983)]. Recent evidence based on the examination of stomach contents of more than 70 individuals from the southeastern United States has suggested that bottlenose dolphins also likely use passive listening for the purpose of detecting and possibly orienting to their prey. The most frequently occurring prey species (i.e., those found in more than 15% of the sample) were invariably those known to be highly active sound producers and, in all instances, producers of sounds of high level (e.g., croakers and drums of the family Sciaenidae, grunts of the family Haemulidae, toadfishes-midshipmen of the family Batrachoididae, and mullets from the family Mugilidae). Although in several instances members of some species could have been captured by active echolocation and visual means, it is highly unlikely that members of other species, by their highly secretive habits, would have been captured by such methods (i.e., members of the family Batrachoididae). Data from other regions corroborate the abovementioned findings that fish producing loud sound comprise the bulk of the diet of bottlenose dolphins.
Designing for the performing arts: An historical overview
Authors: Forsyth Michael
Published: October 2005

A relationship developed historically between the acoustics of different auditorium types and corresponding styles of musical composition. This resulted from the composer's intuitive awareness of room acoustics, and from the architect's limited knowledge, based on trial and error, of the acoustic performance of building materials and room shapes in providing for different purposes. Except where composers have had auditoria specifically designed for their own music, this relationship became tenuous in the present century when the musical repertoire broadened and when nonmusical criteria came to dominate the design process. An auditorium's success now depends on the initial selection of acoustical and other criteria in relation to the building's defined purpose and on the translation of acoustic theory into three-dimensional form. These factors together determine the audience-performer relationship and the shape and dimensions of the enclosing surfaces. Historical examples illustrate early attempts to project theory into built form, and others from the present day describe the problem of integrating opposing acoustic demands.
Sound-speed structure of the Arctic Ocean and the Greenland Sea Marginal Ice Zone
Authors: Newton John L.
Published: October 2005

Within the central ice-covered Arctic Basin, well away from marginal influences, the vertical sound-speed profile can be characterized by four positive sound-speed gradient layers. These layers (and the approximate gradients) are a surface mixed layer extending to a depth of a few tens of meters (∼0.016/s); a layer from the base of the mixed layer to the depth (200–500 m) of the temperature maximum in the Atlantic water (∼ 0.080/s); a layer from the temperature maximum to the bottom of the Atlantic water at about 1000 m (∼0.010/s); and the deep water extending to the ocean bottom (∼0.016/s). The consistent and notable differences evident in the general sound-speed structure of the Canadian and Eurasian Basins can be related to the large scale Arctic Ocean circulation. Profile-to-profile variability and the occurrence of small vertical scale sound-speed features, such as relative minima/maxima and steplike structures, are significantly higher in the Eurasian Basin than in the Canadian Basin. The sound-speed distribution within the Greenland Sea Marginal Ice Zone is dominated by the presence of a strong oceanographic front, the East Greenland Polar Front, which marks the boundary between the cool, fresh Polar water exiting the Arctic Ocean and the adjoining warmer, more saline Atlantic water. Because of this strong ocean front, profile-to-profile sound-speed variability in the Marginal Ice Zone is high over short distances and times. The complex sound-speed structure and its high variability result from various interactions between the two water masses which can take the form of eddies, intrusive layers, and steplike features.
Static and dynamic compliances of the rat tectorial membrane under in vivo-like conditions
Authors: Frommer Glenn H.
Published: October 2005

Compliances were examined qualitatively in 17 rat cochleas prepared minutes after death. Forces were applied perpendicularly to the membrane by a 10-μtin-thick microelectrode that could also vibrate. Sinusoidal displacements were 10-μm peak to peak. White and strobe light illumination was used [G. H. Frommer, Acta Oto-Laryng. 94, 451–460 (1982)]. Static depression of the membrane at points between the spiral sulcus to the outer hair cells produced an elongated hollow, while almost circular depressions were noted in the lip region. Dynamic characteristics were frequency dependent. Under 60 Hz, depression patterns were similar to the static ease. However, at 200 Hz, the region between the sulcus and outer hair cells was quite rigid and displaced at the limbal attachment. Local vibrations of the membrane in the lip region and over the sulcus were almost circular. The mechanical characteristics correlate with ultrastructural studies. It is concluded that static analyses alone cannot describe the viscoelastic behavior of the tectorial membrane.
The relation between the impedance at the microphone and the transfer impedance of an occluded ear simulator
Authors: Kuhn George F.
Published: October 2005

The ANSI S3.25-1979 Standard for an occluded ear simulator specifies the transfer impedance, that is the ratio of the pressure at the measuring microphone to the volume velocity 12.7 mm in front of the microphone, from 100$ndash;10 000 Hz. This paper presents analytical relationships between the transfer impedance and the simulated eardrum impedance. Limit of error calculations are used to derive the expected tolerances for one impedance when the tolerances are specified for the other impedance. [Work support by NIH, NINCDS.]
Longitudinal resonances of the human, external ear; A first approximation
Authors: Kuhn George F.
Published: October 2005

The characteristic equation for a simple model of the external ear is used to demonstrate the individual effects of the radiation impedance, of the pinna, and of the ear canal, including a wedge-shaped termination at the tympanic membrane, on the resonance frequencies of the ear. The results show that a denotation such as “first quarter wave resonance,” suggesting a physical dimension of the ear, which is a quarter wavelength long at the resonance frequency, is a misnomer. [Work supported by NIH, NINCDS.]
Absorption, phase velocities, characteristic impedances and transmission matrices for tubes of small diameter
Authors: Kuhn George F.
Published: October 2005

Numerical results for the absorption, the phase velocities, the characteristic impedances and the transmission matrices, relating input pressure and velocity to output pressure and velocity, are presented for tubes with small radii relative to the boundary layer thickness. These exact results are compared to results based on other approximate analytical models and their range of validity in terms of frequency and tube radius is examined. The exact analytical model yields electroacoustic analogs, for certain sizes and frequencies, which contain nonrealizable negative resistances. It is shown that a change in the length of the tube(s) comprising the transmission line overcomes this difficulty. [Work supported by HIH, NINCDS.]
Backward and forward masking for direct electrical stimulation of the VIIIth nerve in two profoundly deaf subjects
Authors: Dent L. J. , Townshend B. S.
Published: October 2005

Stanford Electronics Laboratories, Stanford, CA 94305 Two profoundly deaf multielectrode implant subjects were required to detect a probe signal (10 ms in duration) in a temporal gap between two pulse-train maskers (each 300 ms in duration). The detection threshold was measured for a probe centered temporally in the gap, as well as for a probe offset from center by up to 97.5%. Also presented were the pure backward and pure forward masking cases. Qualitatively, both subject's forward and backward masking functions approximated those observed for normal hearing subjects [L.L. Elliott, J. Acoust. Soc. Am. 34, 1116–1117 (1962)] in that forward masking decayed more gradually than backward masking as a function of probe-masker separation. Because mechanical (cochlear) contributions to masking [H. Duifhuis, J. Acoust. Soc. Am. 54, 1471–1488 (1973)] can be excluded in the case of direct VIIIth nerve stimulation, these data support the attribution of nonsimultaneous masking phenomena to VIIIth nerve or higher neural mechanisms. [Work supported by NIH.]
Nonorthogonality of measured normal modes in shallow water
Authors: Rayborn Grayson H. , Ioup George E. , Ioup Juliette W.
Published: October 2005

The importance of surmounting the nonorthogonality of measured normal modes and processing shallow-water data in such a way that modal compositions are effectively recovered for matched field processing has been demonstrated by several investigators. The potential for improvement using this technique is greatest when the nonorthogonality of the measured modes is largest. The amount by which the normal modes fail to be orthogonal for a variety of ocean bottoms, array lengths and discretizations, and array positions for the Pekeris model has been studied. Environments are selected to reflect sediment types characteristic of the continental shelf. It has been found that the nonorthogonality is greatest for water depths and frequencies at and just above modal onsets, for sound-speed ratios close to 1, and for arrays which span only a fraction of the water column. Judicious placement and length selection for short arrays, however, can give orthogonal measured modes for some combinations of frequencies and environments when a small number of modes are present.
Spatial matched processing for multipath propagation
Authors: Dzieciuch Matthew , Birdsall T. G.
Published: October 2005

Underwater acoustic propagation is characterized by multipath or multimode propagation. Ray theory and mode theory are not fully adequate for modeling physical reality. Impulse responses can be more accurately calculated using Gaussian beam theory. Signal processors can be designed to take advantage of the channel complexity if the propagation is actually known so that detectability is increased. The proposed technique, channel matched filtering, synthetically backpropagates the wave front to a hypothesized source location. Accurate passive estimates of source location can be made without knowledge of the signal characteristics. GB theory can easily accommodate a range-dependent deep water environment. [This research supported by the Office of Naval Research.]
Limitations to source description set by a finite receiving aperture
Authors: Stocklin Philip L. , Stocklin Norma L.
Published: October 2005

Spatial sampling function theory results are stated for a monochromatic acoustic field and for a bandwidth-limited acoustic field. These results are applied to a finite receiving aperture to estimate the physical limitations to source descriptions obtainable from the outputs of elements of the aperture. Effects of variations in propagation conditions, received signal bandwidth, received SNR, and available processing time are shown. Relationships to inverse field processing are developed.
Near-ground sound fields and surfaces of finite impedance
Authors: Embleton Tony F. W. , Daigle Gilles A.
Published: October 2005

Studies over the past 30 years have elucidated the effects and interactions of many mechanisms involved in sound propagation outdoors. Interactions with the ground surface are especially important for many applications to practical problems. It is particularly significant that ground surfaces are porous, and thus have a finite complex acoustic impedance. The history of sound field measurements to deduce values of impedance, and its direct measurement, is reviewed. Such data prompted, and were then used to validate, models for ground impedance, first in terms of a single parameter (flow resistivity of the ground surface) and later in terms of one to three additional parameters. Impedance values for many grounds at low frequencies are difficult to measure accurately and several new techniques have been developed. Nonisotropic and nonhomogeneous ground has been modeled as a layered medium or as a fluid-filled porous matrix. These efforts have increased our understanding of the acoustical properties of the ground as an air-solid interface from a purely empirical basis to a detailed physical description based on the microscopic properties of the ground.
Airborne sound to seismic coupling: Background for development of the models
Authors: Sabatier James M. , Bass Henry E. , Bolen Lee N.
Published: October 2005

The initial measurements of the acoustic-to-seismic coupling phenomena using loud speakers as sound sources were made by the University of Mississippi and Waterways Experiment Station. For these measurements, a speaker was suspended from a crane, the sound level in the farfield was measured with a microphone above the surface, and the geophone response was measured below the surface. The results of this work largely indicated that the geophone had a response 1000 times greater than one would expect from calculations based upon simple acoustic transmission through a boundary between two perfect fluids. Another finding was that the transit times from speaker to microphone and speaker to geophone were approximately the speed of sound in the air. Success in our lab with water-saturated sediments suggested the application of the Biot model to the ground. Microphones were developed that were used as a pore fluid probe in the soil, and the attenuation and phase speed in the pore fluid were measured. This work led to the understanding of the commonly assumed locally reacting property of the ground.
Application of ray theory to propagation of low-frequency noise from wind turbines
Authors: Hawkins James A. , Blackstock David T.
Published: October 2005

Ray theory has been used to attempt to explain the propagation of very low-frequency (1–20 Hz) noise generated by downwind wind turbines. Two NASA field experiments with a large downwind machine at Medicine Bow, Wyoming, show that the downwind sound decays by spherical spreading near the source, but by cylindrical spreading downrange. Ray theory calculations explain this behavior. The favorable wind gradient downwind causes rays to refract downward and bounce along the ground. A sound channel is built up by repeated jumps in the number of ray arrivals at the receiver as downwind range is increased. Near the source, where no multiply reflected rays arrive, the sound field spreads spherically. The first jump (onset of multiple arrivals) is predicted to occur at about 2 km for the conditions of the NASA experiments. The second NASA experiment confirms this prediction. For the upwind sound, a shadow zone is predicted because of the unfavorable wind gradient. The NASA measurements show only that the upwind SPL in the predicted shadow zone is of order 7–15 dB less than the downwind SPL at comparable ranges. [Work supported by NASA.]
Acoustic wave interactions at a mean-flow stagnation point
Authors: Thompson Charles , Manley Martin
Published: October 2005

The interaction between acoustic disturbances and the mean-flow near the stagnation point of a bluff body will be examined. The stability of such flows will be investigated. It will be shown that streamwise worticity generated by the Stokes layer can enhance the receptivity of the mean-flow boundary layer to free-stream disturbances. The downstream evolution of the vortical motion of the fluid will also be modeled and the results will be presented. [Work supported by Analog Devices Professorship.]
Effects of carbon dioxide and humidity on some acoustic and thermodynamic properties of air
Authors: Wong George S. K.
Published: October 2005

The theoretical data on the combined effects of humidity and carbon dioxide content on some physical properties of air, such as the characteristic impedance ρc and the sound speed c, the specific heats Cp and Cv, and their ratio γ, and the density ρ, have been studied. In general, over the temperature range 0°–30 °C, the normalized values Cp/(Cp)0, Cv/(Cv)0, and c/c0, become larger with the inclusion of humidity and rising temperature; and they are also inversely proportional to the CO2 content. Similarly, ρ/ρ0 and ρc/(ρc)0 are proportional to the CO2 content, but they are inversely proportional to humidity and temperature. However, the normalized value γ/γ0 becomes smaller with the increase of humidity, temperature, and CO2 content. The above reference values, which are indicated with a zero suffix, such as (Cp)0 and ρ0, refer to dry standard air (314 ppm CO2 content) at 0 °C and at a pressure of 101.325 kPa.
Three-dimensional analysis for radial and thickness mode excitation of piezoceramic disk
Authors: Brissaud Michel
Published: October 2005

Standards on piezoelectricity [IEEE Trans. Sonics Ultrason. SU-31 (March 1984)] describe the analysis of vibrations on piezoelectric materials having simple geometrical shapes. The results are based on linear piezoelectricity, and resonance modes are treated as noncoupled vibrational modes. However, in general, real materials involve coupled modes. In addition, they may have one dimension that is not greater than the others. Therefore, it is not possible to state concisely a specific set of conditions under which the definitions and equations contained in the standards apply. However, for pure-mode direction propagation, the mechanical displacements are only related to a single coordinate, and the electrical impedance and general electromechanical coupling factor equations can be determined. With the same assumptions as in the thickness excitation of plates [M. Brissaud, J. Acoust. SOc. Am. Suppl. 1 79, S46 (1986)], that is, completely stress-free boundary conditions and lossless piezoelectric materials, the impedance equation of the disk can be given. It should be noted that, when the piezoelectric coefficients c13D and h33 vanish, the impedance reduces to an equation that is similar to that given by Mason's one-dimensional model [Phys. Acoust. 1 (1964)].
Auditory factors in the perception of stops and glides
Authors: Walsh Margaret A. , Diehl Randy L.
Published: October 2005

There is some disagreement in the literature about the relative contribution of formant transition duration and amplitude rise time in cuing the stop-glide distinction (e.g., [b] vs [w]). Lack of resolution on this point is partly due to the confounding of these two variables in most studies. For a series of identification experiments, sets of synthetic [ba] and [wa] stimuli were created in which transition duration and rise time varied orthogonally. Both variables affected labeling performance in the expected direction, but transition duration was by far the more important factor. A similar pattern of results was obtained for single sine-wave stimuli that modeled the rise times, frequency trajectories, and durations of the first formant in the [ba] and [wa] stimuli. A likely auditory basis for both the speech and nonspeech effects is that rapid frequency transitions and short rise times both contribute to a higher degree of spectral splatter (i.e., dispersion of energy across a wide frequency band), which may be a principal cue for abrupt onsets in general. [Work supported by NICHD.]
Identification of stop consonants from continuous speech in limited context
Authors: Lamel Lori P.
Published: October 2005

This paper describes a series of perceptual experiments aimed at evaluating listeners' ability to identify singleton stop consonants. The stimuli consisted of portions of speech extracted from a corpus of about 3600 sentences spoken by over 4.50 talkers. A variety of factors, such as the stress pattern, syllable position, and phonemic environment, were systematically varied to determine the extent to which these factors influence the listeners' decision. At least 20 listeners heard syllable-initial stops in one task and nonsyllable-initial stops in another. In each task, only the immediate surrounding context was available. Overall identification rates were better than 97% for syllable-initial stops and 85% for the noninitial stops. In the syllable-initial case, 75% of the errors were in voicing, predominantly voiced stops being heard as their voiceless counterparts. For the noninitial stops, 88% of the errors were also in voicing. However, in this task, most of the errors were voiceless stops heard as voiced. Almost 70% of the errors for the noninitial stops involved alveolars, whereas no particular place of articulation accounted for most of the errors for the initial stops. Factors other than syllable position will also be discussed, and comparisons will be made to other stop identification studies. [Work supported by DARPA under contract N00014-82-K-0727, monitored through the Office of Naval Research.]
Perceptuomotor adaptation to natural speech adaptors
Authors: Shuster Linda I.
Published: October 2005

Perceptuomotor adaptation has been demonstrated using synthetic adaptors [Cooper and Nager, J. Acoust. Soc. Am. 58, 256–265 (1975); Jamieson and Cheesman, J. Phonet. 15, 15–27 (1987)]. However, similar results have not been obtained when natural adaptors were employed [Summerfield et al., J. Phonet. 8, 491–499 (1980); Shuster and Fox, ASHA 28, 72 (1986)]. In the present study, subjects produced examples of the token [rəthi] prior to adaptation and then after perceptual adaptation with a naturally produced [rəphi]. Voice onset time was then measured for each spoken utterance. It was found that VOTs were significantly shorter after adaptation to the bisyllable than they were prior to adaptation. These results are similar to those obtained with synthetic adaptors and support the notion that perceptuomotor adaptation reflects a perception-production link.
Acoustic versus phonological context effects in the perception of formant transition continua
Authors: Nearey Terrance M. , Shammass Sherrie E.
Published: October 2005

The effect of the following vowel on listener's judgments of place of articulation in formant transition continua is perhaps the archetypal example of context sensitivity in speech perception. Is this sensitivity explicable in terms of acoustic context alone or must the judged phonological identity of the following vowel also be considered? Shammass [unpublished dissertation, University of Alberta (1985)] designed transition-continua experiments involving ambiguous and nonambiguous vowel stimuli to examine this question in detail. Linear logistic analysis of these data confirms the systematic role of acoustic context effects, but indicates that the phonological context effects are of a highly restricted nature and, furthermore, are both task and listener dependent. A summary of this analysis is presented and the results are discussed in the context of other results from the literature (including those of Mermelstein, Massaro, Ohala, and Repp). The evidence is consistent with the hypothesis that phoneme-sized units (rather than diphones or larger units) serve as the primary interface between acoustic cues and phonological units.
Aural signal processing in the context of a digital hearing model
Authors: Chabries D. M. , Christiansen M. W. , Christiansen R. W. , Brey R. H. , Harris R. W. , Robinette M.
Published: October 2005

A nonlinear model of the human aural system is proposed. Utilizing a homomorphic transformation, the model is implemented in a digital computer and accounts for the Fletcher critical bands in hearing, the nonlinear relationship between input sound intensity and perceived loudness known as recruitment, as well as the thresholds of audibility and the maximum tolerance threshold. The transformed or output signal domain is identified as a perceptual space. It is shown that the model can be used to reproduce the Fletcher Munson equal loudness contours for normal hearing populations. Modification of easily measured parameters results in the generation of equal loudness contours for hearing impaired individuals. Through a combination of the forward homomorphic transformation for normal hearing to the perceptual domain and the inverse mapping utilizing parameters of a hearing impaired individual, it is shown that hearing intelligibility is dramatically improved. It is postulated that signal processing performed in the perceptual domain will produce results more pleasing and intelligible to the listener. Results of auditory testing utilizing the proposed model for both normal and hearing impaired populations are presented. Speech intelligibility scores are shown to increase by more than 40% (i.e., from 40% to over 80%) over preprocessed scores under headphones for hearing impaired subjects used in these tests.
The role of static spectral properties in vowel identification
Authors: Hillenbrand James , McMahon Brendan J.
Published: October 2005

The classic study of Peterson and Barney [J. Acoust. Soc. Am. 24, 175–184 (1952)] demonstrated that vowels in CVC context are identified with a very high degree of accuracy by human listeners. A number of attempts have been made to classify the Peterson and Barney vowels using some combination of F0 and the three lowest formant frequencies. Although these attempts have been reasonably successful, the accuracy of the classification algorithms has been well below that of human listeners. Human listeners, however, have access to dynamic (duration and spectral change) as well as static information (F0 and “target” formant frequencies). The purpose of the present study was to determine the identifiability of the Peterson and Barney vowel set based exclusively on static information. A formant synthesizer was used to generate isolated, steady-state versions of all 1520 vowels (76 speakers × ten vowels × two repetitions) in the Peterson and Barney data base using measured values of F0 and F1–F3. Preliminary results from a small group of trained listeners suggest that the error rate for the resynthesized stimuli is 3.5–4.5 times greater than the error rate reported in the original Peterson and Barney study. Pilot data also suggest that the identification task may be extremely difficult for some untrained listeners. [Work supported by Rome Air Development Center under contract ♯ F3060285-C-0008.]
Vowel and consonant judgments are not independent when cued by the same information
Authors: Whalen D. H.
Published: October 2005

Vowels with ambiguous formant patterns can be perceived as changing from [æ] to [ε] as duration decreases. Similarly, ambiguous final consonants can be perceived as [d] when the preceding vowel is long and [t] when it is short. Mermelstein [Percept. Psychophys. 23, 331–336 (1978)] had subjects identify four syllables (“bad,” “bat,” “bed,” and “bet”) that were synthesized by varying only vowel duration. He claimed the consonant and vowel judgments were independent, implying distinct vowel and consonant systems. The present replication indicates that Mermelstein's analysis lacked power and that, in fact, the judgments are dependent on each other: Subjects were more likely to report hearing [æ] and [t] together, since more of the vocalic segment duration is attributed to the vowel and/or less is attributed to the consonant. Similarly, subjects are more likely to report hearing [ε] and [d] together, since there is less duration attributed to the vowel and/or more attributed to the consonant. This indicates that listeners hear a coherent syllable, in which two distinctions that depend on the same information must compete for that information. [Research supported by NIH Grant HD-01994.]
Molecular basis of the acoustic properties of polyurethanes
Authors: Hartmann Bruce , Lee Gilbert F. , Duffy James V.
Published: October 2005

The acoustic properties of polyurethanes as a function of frequency are dominated by a very broad relaxation process in which the polymer makes a transition from the rubbery to the glassy state. It is convenient to use an analytic representation of the data that expresses the complex shear modulus in the form G* = G∞ − (G∞ − G0)/[1 + (iωτ)α], where G0 is the limiting low-frequency modulus, G∞, is the limiting high-frequency modulus, τ is the relaxation time, and α is a dimensionless parameter, with values between zero and one, that determines the width of the relaxation. It will be shown that τ is related to the glass transition temperature Tg of the soft segment in the polyurethane. In turn, Tg is found to depend on the degree of phase separation that occurs between the chemically incompatible hard and soft segments, as well as the degree of crystallinity developed. These same molecular factors determine G0, but G∞ is found to be essentially constant. Attempts to relate a to the width of the glass transition as measured in a differential scanning calorimeter have been only qualitatively successful. [Work supported by the NSWC Independent Research Program and the Office of Naval Research.]