bluuebunny/crossref_metadata_2025 huggingface dataset:
Updating UPSID
Authors: Maddieson Ian , Precoda Kristin
Published: October 2005

UPSID—the UCLA phonological segment inventory database—is a database containing the phoneme inventories of a large genetically based sample of languages [I. Maddieson, Patterns of Sounds (1984)]. Each phoneme is specified in terms of a comprehensive set of phonetic features. The first version of the database has proven useful to scholars interested in phonological universals and theories concerning the structure of phonological systems [e.g., B. Lindblom and I. Maddieson, in Language, Speech & Mind, edited by L. M. Hyman and C. N. Li (1988); K. Stevens and S. J. Keyset, Language 65, 81–106 (1989)]. An expanded and corrected second version is currently in preparation. This version improves the sample, increasing coverage of previously undersampled language families and correcting a few oversampling errors, and correcting errors in individual language inventories. A new custom-written software package for MS-DOS systems provides economical and flexible means of storing and modifying this enhanced database and outputting subsets of the data for further analysis. The database is stored as several separate but interrelated modules. One contains a listing of character codes for each distinct segment type occurring in the database paired with a standard phonetic description and with the list of features assigned to that segment. Another contains the phoneme inventories as a set of character codes for each language. The database is principally used by mating information from these two modules, for example, by creating a file containing the fully specified feature descriptions of the segments in a group of languages, or all segments defined by a selected set of feature values. Such files can be exported to a standard statistics package for sophisticated processing, but simpler counting operations can be performed within the UPSID program.
A high-resolution beamformer for acoustic imaging
Authors: Knudsen Donald C.
Published: October 2005

This paper describes the development of a high-resolution imaging sonar system based on innovative digital beamforming technology. The new beamformer provides rapid access to a high-resolution, three-dimensional image field, scanning simultaneously in both azimuth and elevation, to produce a “staring” image as well as traditional range/bearing presentation. The new digital beamformer operates on sampled data, and uses time delays rather than phase shifts for broadband signal capability. It also features dynamic, pulse-tracking focusing from within one aperture of the array. An experimental imaging sonar with a 64-channel digital beamformer has been constructed. A highly parallel, pipelined architecture is employed to obtain a processing rate of four million beam output samples per second, delayed and summed across 64 hydrophone channels. This paper describes the technology and presents results obtained with the prototype system, including samples of imagery.
Characterization of the properties of underwater acoustical materials
Authors: Ting Robert Y.
Published: October 2005

Increased emphasis on the development of new materials for underwater acoustical applications requires that material properties be carefully characterized during the research phase of the investigation. Once measured in the laboratory, the chemical, physical, and acoustical properties of the materials are correlated in order to enable one to design new acoustical materials. This approach has been successfully employed at NRL-USRD for the past years in the study of both new transduction materials and acoustical elastomers. Piezoelectric and dielectric properties of new ceramics and piezocomposites are characterized. Dynamic moduli in shear, tensile, and bulk modes are also determined for elastomers, while their chemical properties are analyzed by using a number of instrumental techniques, including gel permeation and liquid chromatography, Fourier transform infrared spectroscopy, and thermal gravimetric analysis. These measurement techniques will be described, and their frequency, temperature, and pressure capabilities will be discussed.
Proportional target-tone duration as a limit on pattern discriminability: Multicomponent targets
Authors: Watson Charles S. , Kidd Gary R.
Published: October 2005

Under high levels of stimulus uncertainty, pattern discrimination performance is often degraded so severely that it would be reasonable to question whether the same processing mechanisms used under minimal-uncertainty conditions are still available to the listeners. It has previously been reported that the degrading effects of high stimulus uncertainty depend strongly on the proportional duration of a pattern component that is subject to change, relative to the total duration of the pattern [J. Acoust. Soc. Am. Suppl. I 83, S40 (1987)], but only weakly on the absolute durations of the stimuli. Again using tonal sequences and a frequency discrimination task, this finding was extended to the case of multiple-component target sounds. Results suggest that detectability of pattern changes is a function of both the proportional duration of each changed tone and the number of changed tones, and not solely a function of the total proportion of the pattern that has been changed. [Work supported by AFOSR and NIH.]
Multiple looks and temporal integration
Authors: Viemeister Neal F. , Wakefield Gregory H.
Published: October 2005

Recent research on temporal resolution indicates that a person can listen using a temporal window with a time constant of less than 10 ms. A basic question is how information from these brief “looks” is combined and processed. The initial focus is on the old problem of temporal integration and data are reported from an experiment similar to Zwislocki's two-click experiment [J. Zwislocki, J. Acoust. Soc. Am. 32, 1046–1059 (1960)]. The data are well described by assuming true integration for brief click separations and by assuming two independent looks for separations longer than approximately 10 ms. Thus the data are more consistent with multiple looks than with true, long-duration temporal integration. To account for more general “integration” phenomena, a simple multiple-look strategy is compared with neural summation using spike data recorded from the auditory nerve of chinchilla for CF tones. The decision statistic is either an optimally weighted combination of the spike count from contiguous short bins (looks) or the total spike count summed over the duration of the tone. The time-intensity trading functions (threshold for d′ = I versus duration) for these schemes are very similar. This demonstrates, consistent with intuition, that a multiple-look scheme can show integration-like behavior and that such behavior does not imply true integration. [Work supported by NS12125.]
Recognition memory for arbitrary, complex waveforms
Authors: Yost William A.
Published: October 2005

Recognition memory for arbitrary complex stimuli was measured using the operating characteristic technique developed by Egan [J. P. Egan, AFCRC TN 58-51, AD 152650 (1958)]. Ten complex stimuli (500 ms in duration), each consisting of six, randomly chosen, equal amplitude sinusoidal components spanning frequencies from 300 to 3000 Hz, were used as a training set. After 15 min of listening to the training set, listeners were presented a 20-stimulus test set consisting of the ten training stimuli along with ten additional complex stimuli (generated in the same way as the ten training stimuli). In a recognition memory paradigm, listeners used a five-point rating scale to rate their confidence that a stimulus presented from the test set was from the training set. Different constraints were placed on the selection of the complex waveforms to determine how those constraints affected recognition memory. For one condition there were no constraints except as described above; for a second condition the six-tonal components were harmonics (randomly chosen) of a common fundamental; and for a third condition the six-tone complex was sinusoidally amplitude modulated at rates of 4, 16, or 32 Hz. The results will be discussed in terms of auditory object perception and the use of receiver operating techniques to measure recognition memory. [Work supported by a Program Project Grant from the NINCDS and a grant from AFOSR.]
A psychophysical evaluation of a model for suppression and excitation in remote masking
Authors: Goldstein J. L. , Gilkey R. H. , Quin̄ónez R. E.
Published: October 2005

In the classical literature both the masking and the suppression of one tone by a second tone of lower frequency have been shown to be nonlinear functions of overall level. Here, the relation between these two tasks is further constrained by measuring both remote masking and suppression with the same subjects. The remote masking experiment measures the effect of a 602-Hz sinusoidal masker on the detectability of a simultaneous 1500-Hz sinusoidal signal, while the suppression experiment measures the effect of a 602-Hz sinusoidal suppressor on the forward masking produced by a 1500-Hz sinusoid. The data agree with earlier studies and are interpreted in the context of the multiple band pass nonlinearity model [J. L. Goldstein, “Updating cochlear driven models of auditory perception: A model for nonlinear auditory frequency analyzing filters,” in Working Models of Human Perception, edited by B. Elsendoorn and H. Bouma (Academic, New York, 1989), pp. 19–57], which suggests that both excitatory and suppressive mechanisms influence remote masking. [Work supported by NSF and AFOSR.]
Area under power-law receiver operating characteristics
Authors: Pollack Irwin
Published: October 2005

The receiver operating characteristic, or ROC curve, describes performance under a wide range of hit (H) and false alarm (FA) probabilities. In actual practice, with many different experimental conditions, only a single hit-false alarm (H-FA) combination may be available to describe performance under each condition. One way to describe each performance level is in terms of the area under an ROC curve passed through the obtained H-FA combination. James Egan astutely observed that most ROC curves are well approximated by power-law functions of the form: H = [FA]λ. Egan's power-law functions also satisfy the requirement of passing through the coordinate pairs: 0,0; H-FA; and 1.0,1.0. The area under power-law ROC curves is simply 1/(1 + k). An alternative approach is to assume an ROC curve with linear segments between 0,0 and H, FA; and between H, FA and 1.0, 1.0. The calculated area by the linear assumption is not constant for a fixed ROC curve; it varies with the H,FA level. The linear assumption underestimates, and at times seriously underestimates, the area under power-law ROC curves. The minimum error of underestimation is over 4% for k between 0.05 and 0.5. The maximum error of underestimation approaches the maximum possible error. Clearly, the linear approximation procedure can grossly distort the area under the ROC curve.
Acoustics of creaky voice
Authors: Titze Ingo R.
Published: October 2005

Creaky voice seems to be perceived as some combination of low pitch and roughness. Preliminary perceptual experiments suggest that whenever the fundamental frequency drops below about 70 Hz, pulses are perceived in phonation. The combination of several closely spaced harmonics of different (and varying) amplitudes may then add roughness if the harmonies fall within the same critical band. In addition to relaxing the laryngeal muscles to produce low F0, the larynx can create asymmetric vibrations and mode changes that produce subharmonic components. These contribute to the perception of low F0. Subharmonics are a result of bifurcation of a single stable vibratory state into multiple quasistable states and can lead to chaotic vibratory behavior if asymmetries or nonlinearities in the system are increased.
In vitro ultrasonic heating of fetal bone
Authors: Drewniak J. L. , Carnes K. I. , Dunn F.
Published: October 2005

Estimates of temperature increase in soft fetal tissues exposed to medical ultrasound can be made with existing knowledge. However, little knowledge is available to make a resonable estimate of the temperature increase in fetal bone and surrounding tissues during in vivo exposures. An experimental study of the temperature increase measured in vitro in human fetal femurs exposed to 1-MHz, continuous wave ultrasound at 37°C has been conducted. The temperature increase in the femur specimens is measured with a small implanted thermocouple probe. Specimens range in gestational age from 59 to 150 days. The temperature increase measured in a 59-day gestational age specimen at 20 s following the initiation of exposure at 1 W/cm2, for example, was 0.10°C, while the temperature increase for a 108-day specimen with the same exposure conditions was 2.9°C. A temperature increase of 4.0°C was measured in the 108-day specimen for a 60 s exposure at 1 W/cm2. The temperature rise is found to increase linearly with intensity over the range of 0.5 to 10 W/cm2. The initial rate of the temperature increase, evaluated at 0.2 s following initiation of exposure, has also been measured and is found to increase rapidly with gestational age. For example, I−1 dT/dt at 0.2 s measured for the 59-day specimen is 0.06°C/s/(W/cm2), while for the 108-day specimen 0.81°C/s/(W/cm2) was measured. The values of initial temperature increase measured in the fetal femur specimens can be compared to that obtained for soft tissue using the quantity I−1 dT/dt = 2α/ρCρ. For α = 0.05 cm−1, and ρCρ = 3.78 J/°C/cm3, I−1 dT/dt = 0.026°C/s/(W/cm2). The values of I−1 dT/dt measured for the 59- and 108-day gestational age specimens are approximately 2 and 30 times greater than that for soft tissue.
Temperature rise in a focused Gaussian beam of ultrasound
Authors: Wu Junru , Du Gonghuan
Published: October 2005

The steady-state temperature elevation generated by a focused Gaussian beam including the effect of perfusion has been calculated along the beam axis. The medium is assumed to be a homogeneous absorbing one. The results indicate: (1) the temperature rise is an increasing function of the intensity gain of the focusing transducer, but seems never to exceed twice of that at the interface of the transducer and the medium generated by its unfocused counterpart; (2) the temperature rise at the interface of the transducer and the medium is not affected by focusing significantly. [Work supported by the American Cancer Society (IN-156) and by NSF and Vermont Epscor.]
The psychoacoustics of loudspeaker sound reproduction—Past achievements and present problems
Authors: Toole Floyd E.
Published: October 2005

The design of loudspeakers is gradually being put on a scientific basis. Art and intuition have given way to engineering guidelines as the relationships between perceptions and technical measurements have been elucidated. Indeed, within limited circumstances, loudspeakers can be designed to meet specific engineering design objectives, with considerable confidence in how they will be subjectively evaluated. In practice, however, all of the relevant conditions are not controlled, and several factors conspire to preclude universal satisfaction among listeners. Most of the uncertainty appears to be in the interactions between loudspeakers, rooms, listeners, and program material. This paper reviews the present state of knowledge, and outlines the areas most in need of further work. It is clear that additional psychoacoustical data and suitable technical innovations can alleviate some of the remaining problems. Others, though, may be better treated by standardization.
High-frequency acoustic imaging of the seafloor
Authors: de Moustier C.
Published: October 2005

For decades, sidescan sonars have been the primary tool to obtain acoustic images of the seafloor. Such images provide qualitative information on the seafloor surveyed based on amplitude variations of the backscattered acoustic signals received. In the 1980s, bathymetric sidescan sonar systems, capable of simultaneously producing acoustic images and measuring depth at numerous points across the swath, added a quantitative description of the seafloor in the form of a depth contour map. Similar claims can be made with multibeam echo sounders well known for their high-resolution swath bathymetry capabilities. Taking advantage of this high bathymetric resolution, the beamformed acoustic backscatter data can also be displayed as a geometrically correct acoustic image of the seafloor and provide textural information not available in the contoured bathymetry of the same area. Likewise, knowledge of the bathymetry, particularly bottom slopes, is needed to correct for the angular dependence of seafloor acoustic backscatter and construct a map of acoustic backscattering strength over the area. Such a map will give clues to regional variations in lithelogies.
Estimation of sediment volume scattering strength and absorption coefficient
Authors: Stewart Robert A. , Chotiros Nicholas P.
Published: October 2005

A simple theoretical model for estimating sediment volume scattering strength and absorption coefficient from the reverberation tail of an acoustic pulse observed by a point receiver within the ocean sediment is described. This model assumes an infinite plane wave entering the sediment at normal incidence and scattering isotropically from a uniform field of random inhomogeneities. The model predicts the magnitude and decay rate of the volume scattering at a point receiver in terms of the volume scattering strength and the absorption coefficient of the medium. By fitting the predictions of the model to experimentally observed volume scattering data, the volume scattering strengths and absorption coefficients of real sediments were inferred. Expeimental data were obtained from insediment acoustic probes deployed at sites off Kings Bay, GA, and off Panama City, FL. The absorption coefficient results were compared with corresponding measurements from core samples. [Work supported by ONT with NORDA management.]
Remote sensing of particle motion by cross correlation of acoustic backscatter
Authors: Chotiros Nicholas P.
Published: October 2005

In a previous design study, it was determined that clouds of passive riders in the upper ocean, such as bubbles and suspended matter, could produce detectable and unique acoustic backscatter signatures. It is postulated that the acoustic signatures may be processed to yield the movement of the passive riders, hence image the motion of the water itself. Methods based on cross correlation processing were developed. Computer simulations and experiments were carried out to test the methods. [Work supported by ONR.]
The organ as a teaching resource in architectural acoustics
Authors: Kinzey Bertram Y.
Published: October 2005

The study of architectural acoustics must include an acquisition by students with limited or no musical background of a range of knowledge beginning with the harmonic structure of sound and its effect on timbre and continuing through an awareness of the ways, for better or worse, the architectural enclosure creates a reverberant field that affects various attributes such as blend, clarity, and fullness of tone and completes the musical sound that is heard. Aural experiences are essential if the student is to understand fully musical sound and the role of an architectural enclosure in its proper audition. The organ is an ideal instrument for demonstrations that give this experience. A series of live and recorded demonstrations using the organ is described which have been used over several years to show the relationship between music and architectural acoustics.
Changes in vocal fold length with nerve stimulation in canine larynges
Authors: Jiang Jiaqi , Titze Ingo R.
Published: October 2005

Vocal fold length change and adductory movement of the glottis were obtained from seven canine larynges when the recurrent and the superior laryngeal nerve were stimulated bilaterally. A surgical procedure was used to make marks on the vocal fold for length measurements. Frame by frame analysis was used to get static and dynamic data on vocal fold length. The mean maximum elongation of the membranous vocal folds was 44.7% with 120-ms elongation time and 140-ms relaxation time. The mean maximum shortening of membranous vocal folds was 17.8%, with 110-ms contraction time and 115-ms relaxation time. When both agonist and antagonist muscles were in maximum contraction, the mean elongation was 24.3%. The length-time curve approximates an exponential function. Closing of glottis occurred in less than 35 ms. The temporal results agree with earlier pitch change results. [Work supported by NIH Grant NS16320.]
Comparison of ANSI noise dosimeters and IEC personal sound exposure meters
Authors: Krug Robert
Published: October 2005

ANSI and IEC have prepared draft standards on meters to measure workers sound exposure. The standards have a number of similarities. Both are based on IEC 804, integrating-averaging sound level meters, and both are intended to specify meters worn on a worker and to measure noise throughout a work day. The ANSI draft specifies meters with exchange rates of 3, 4, or 5 dB, with a fast or slow time constant and a threshold. The IEC draft specifies only a 3-dB exchange rate with no time constant or threshold. The ANSI draft specifies a read out in percent dose while the IEC draft is developed around Pascal squared hours. For continuous noise above the threshold, the reading can be converted from one set of units to the other. For intermittent and impulsive, a noise significant difference will occur.
Binaural simulation technique for scale modeling
Authors: Xiang N. , Blauert J.
Published: October 2005

Room simulation for the purpose of predicting acoustic behavior and quality has recently become a popular topic in room acoustics. At the Ruhr University, the experience of the binaural human listener in a room simulation using both physical and computer modeling has been authentically recreated. This paper reports on the latest stage of the present work in binaural room simulation using a scaled down physical model, where a sensitive miniature dummy head with accurately scaled pinnae (scale factor 1:10) serves as a receiver to pick up the model sound field. A versatile PC-based system measures the binaural impulse responses in model space according to the m-sequence transform and also carries out the convolution of the room-impulse responses with anechoic speech or music signals. An especially wide broadband ultrasonic transmitter system is necessary to provide the desired complex stimulation of the sound field. After the necessary signal processing, the resulting signals can be listened to binaurally via headphones. [Work supported in part by Deutsche Forschungsgemeinschaft.]
Observation of acoustic cavitation in excised canine urinary bladders
Authors: Fowlkes J. B. , Carson P. L. , Chiang E. H. , Rubin J. M.
Published: October 2005

A high-intensity ultrasonic field has been used to generate bubbles within urinary bladders excised from dogs. Following the exposure, bubbles were easily visualized using an ATL UltraMark 9 diagnostic scanner with a 5-MHz mechanical sector scanhead. Scattering of the high-intensity ultrasound by the bubbles was also observed as uniformly high amplitude scan lines during the irradiation. The bladders were removed surgically after tying off the ureters and urethra to prevent urine loss and exposure to external contaminants. Each bladder was sealed in a plastic bag filled with a degassed aqueous solution of sodium chloride (0.9% by weight) and sodium azide (1 g/liter). A bag containing a bladder was centered in a sealed degassed water path at the common focal point of a 7-cm-diam, 500-kHz transducer, and a 10-cm-diam brass reflector. The transducer and reflector were both focused at 10 cm and were aligned coaxially. Exposures were 10 s in duration but intensity or pressure values were not available as of the writing of this abstract. No significant discoloration of the exterior bladder wall was observed within the beam path. Results of this study suggest that bubbles might be externally generated in vivo for use as echo contrast agents and for studying the bioeffects of acoustic cavitation.
Measurement of the local power reflection coefficient in the ear canal using the two-microphone method
Authors: Murphy W. J. , Tubis A. , Long G. R.
Published: October 2005

The local power reflection coefficient in the ear canals of humans has been measured using the two-microphone technique [W. J. Murphy, A. Tubis, and G. R. Long, J. Acoust. Soc. Am. Suppl. 1 82, S70 (1987)]. The accuracy of the calibration technique and the assessment of the random and bias errors associated with the reflectance have been investigated. The use of bandlimited noise as a stimulus results in decreased coherence between the two microphone signals relative to the coherence obtained with sinusoidal or other deterministic signals (e.g., a chirp signal) at high frequencies. The effects on the coherence and the reflectance due to sinusoidal chirp and bandlimited stimuli are presented. The measurements were performed in an acrylic hard-terminated tube. As a test of the assumption of local one dimensionality of the ear canal, the reflectance was measured as a function of the insertion depth of the two-microphone probe in several subjects. At high frequencies, the measured reflectance was found to vary significantly in ear canals with contorted geometries. In relatively straight ear canals, the measured reflectance was found to vary by less than 10% for changes of 5 mm in the insertion depth. [Work supported by a grant from the Deafness Research Foundation.]
Some implications of nonlinear passive models of spontaneous otoacoustic emissions
Authors: Talmadge C. , Tubis A. , Long G. R.
Published: October 2005

The nonlinear passive cochlear model of M. Furst and M. Lapid [J. Acoust. Soc. Am. 84, 222–229 (1988)] has recently been criticized by Wit [J. Acoust. Soc. Am. 85, 2217 (1989)] on grounds that this model is not able to account for the observed statistical [Bialek and Wit, Phys. Lett. A 104, 173–178 (1984)] and external-tone entrainment [Long et al., Basic Issues of Hearing, edited by H. Duifhuis et el. (Academic, London, 1988), pp. 93–100] properties of spontaneous otoacoustic emissions. The argument of Wit and the response to it by M. Furst [J. Acoust. Soc. Am. 85, 2218–2220 (1989)] are critically examined. On the basis of numerical simulations and exact solutions [Caughey and Payne, Int. J. Non-Linear Mech. 2, 125–151 (1967)] of the Fokker-Planck equation for stochastically driven nonlinear passive oscillators, it is concluded, in agreement with Wit, that nonlinear passive cochlear models of the type used by Furst and Lapid are not able to adequately describe the observed behavior of spontaneous otoacoustic emissions. [Work supported by NIDCD.]
Forward masking of the frequency following response
Authors: Krishnan A. K. , Durrant J. D.
Published: October 2005

The frequency following response (FFR) reflects sustained neural activity within the brain stem, phase locked to the stimulus at low frequencies. Attempts at resolving the issue of the origin of FFR along the basilar membrane via the subtractive high-pass masking technique have been unsuccessful. A tone-on-tone forward masking paradigm is described as an effective alternative method for assessing the place specificity of FFR. Recordings were made between forehead and the ipsilateral mastoid using normal adult subjects. FFRs were elicited by a 500-Hz tone burst with masking frequencies from 500–4000 Hz presented at 10 ms before the probe. Results indicate that the FFR (500 Hz) was most affected by the 1000-Hz masker, with less masking at 500 Hz and little or no effects at frequencies above 1000 Hz. Additionally, some subjects exhibited an enhancement of the FFR for maskers about 1000 Hz. Results suggest that our method is effective in evaluating frequency specificity of the FFR. A serendipitous finding revealed a possible means for discriminating between what might be separate on-set and sustained components of the FFR.
The representation of synthetic stop consonants in a computational model of the dorsal cochlear nucleus
Authors: Pont Michael J. , Damper Robert I.
Published: October 2005

A computational model of part of the mammalian auditory nervous system is described. This consists of three scissile stages simulating afferent neural processes within the cochlea, auditory nerve, and dorsal cochlear nucleus (DCN). The model derives its input from a 128-channel cochlear filter bank. The artificial neuron units employed are simple encoder models whose responses and interconnections follow closely those reported in recent anatomical and physiological studies. The activity of the low-frequency DCN projection units in the model to three (labial, alveolar, and velar) synthetic voice onset time (VOT) stimulus series is shown to mimic the behavioral responses of human and chinchilla subjects [P. K. Kuhl and J. D. Miller, J. Acoust. Soc. Am. 63, 905–917 (1978)] to these stimuli. The results suggest that DCN processing may have an important role in the perception of voicing contrasts. [Work supported by SERC and NICHD Contract N01-HD-5-2910.]
Duration models and segmental quality in a text-to-speech system
Authors: Carlson Rolf , Granström Björn
Published: October 2005

We have had the privilege of working together with Dennis Klatt for many years. In our presentation we will refer to some of Klatt's work that has had an influence on our own work. Modeling of segmental duration was a central part in Klatt's work during the 1970s. This work resulted in a duration model in 1979 that captures many of the basic effects found in speech. This model has been used as a framework in the KTH text-to-speech system. The use of quantity in Swedish demands expansions of the model. The Swedish study has been done in the context of a speech database from different speakers reading different text materials. In Klattalk, Klatt [J. Acoust. Soc. Am. 82, 737–793 (1987)] addressed all levels in a text-to-speech system, but special effort was placed on a general improved segmental quality. The quality of the best speech synthesis is, however, still far from that of human speech. Some recent efforts to improve the segmental intelligibility in our system will be described. This includes experiments with new synthesis strategies with an emphasis on modeling contextual variability. Analysis and synthesis of positional variants of the Swedish consonants are reported, and new strategies for synthesis are discussed. Based on analysis of the speech database, consonant rules affecting both source and resonator features were formulated and tested. Special efforts were made to handle the realization of consonant clusters. In this development work, diagnostic tests were used at regular intervals. Results from the last years' evaluation will be reported and discussed.
Acoustic interactions with buried objects
Authors: Lim Raymond , Hackman Roger H.
Published: October 2005

The theoretical description of the scattering from objects near medium boundaries has been the subject of a number of recent articles [e.g., G. Kristensson and S. Ström, J. Acoust. Soc. Am. 64, 917–936 (1978); G. S. Sammelmann and R. H. Hackman, J. Acoust. Soc. Am. 82, 324–336 (1987)]. These articles consider scatterers in nonattenuating, liquid media. When objects buried in sediment are considered, however, sediment attenuation introduced fundamentally new features into the description of the scattering process. The interaction of an incident plane wave with the sediment-seawater interface distorts the phase-amplitude relation of the wave incident on a scatterer in the sediment. The description of the scattering of these inhomogeneous waves requires the introduction of new, “suitably modified” basis states for the solution of the scattering problem. The modification of the T-matrix and waveguide theories developed at the Naval Coastal Systems Center is discussed and a numerical study of the scattering by a thin spherical shell buried in an attenuating liquid sediment is presented.
Parabolic equation starting field for a low-frequency source near an interface
Authors: Gilbert Kenneth E. , Huang Dehua
Published: October 2005

Experiments in underwater acoustics often use a low-frequency source close to the bottom. In such situations, the standard Gaussian starting field for parabolic equation calculations can grossly violate the boundary conditions at the water-bottom interface. In this paper, a simple operator method is derived that gives the correct boundary conditions in the starting field and also properly excites the “high-angle” modes and the continuum. As limiting cases, the standard Gaussian starting field of Brock [NORDA Tech. note 12, Naval Ocean Research and Development Activity, NSTL Station, MS (1978)] and the high-angle starting field of Greene [J. Acoust. Soc. Am. 76, 1764–1773 (1984)] are obtained analytically. The method used here is compared with numerical results from a fast-field program for a low-frequency source near the bottom in shallow water.
Teleconferencing—The acoustical environment and directional transducers
Authors: West J. E. , Elko G. W.
Published: October 2005

The performance of teleconferencing systems are largely a function of the acoustical environment, the type, and the location of loudspeakers and microphones. Acoustical treatment in most conference rooms is limited to carpeted floors and an acoustical ceiling. For the most part, present microphones are designed to work best under close talking conditions and have omnidirectional, or first-order unidirectional characteristics. Two small experimental conference rooms have been constructed in which the effects of room parameters such as reverberation time, noise level, and discrete echo control can be independently studied. Broadband (8-kHz) directional microphones and steerable directional arrays, mostly of second order, are being investigated for a variety of acoustical environmental conditions. The easily adjustable conference rooms allow us to investigate these microphone systems with respect to their directional properties, placement, and steering algorithms. Loudspeaker directional characteristics, and advantages of stereo transmission are also under investigation. The experimental conference rooms and new directional microphones will be described and a discussion of trade-offs between major parameters for speech reception and transmission will be given.
Microphones and feedback suppression in teleconferencing
Authors: Petterson Michael , Julstrom Steve
Published: October 2005

The effect of room acoustics on microphone placement and selection for teleconferencing systems and current approaches to electroacoustic feedback and echo control in teleconferencing systems will be discussed.
Timbre and tone height
Authors: Patterson Roy D.
Published: October 2005

A sound composed of all harmonics of 131 Hz in cosine phase produces the note C3 with a buzzy timbre. One can produce similar notes that are all C's but that sound “higher” than the original C, in at least three different ways. One can (a) attenuate the lower harmonics, (b) attenuate the odd harmonics, and (c) shift the phase of either the odd or the even harmonics. Traditionally, these perceptual changes would be regarded as timbre changes because tone chroma is fixed, but these manipulations also affect the tone height of the sound. Accordingly, experiments were performed to investigate the relationship between timbre and tone height. The stimuli were presented at six octaves with fundamentals ranging from 32–1024 Hz. The listeners judged the octave of the sounds on a scale from 1 to 6. The results show that waves with the same period can produce tone-height judgments that consistently differ by more than an octave, and that a substantial component of many presumed timbre differences (e.g., between a piano and a harpsichord) is actually a tone-height difference.
Pitch and timbre interaction in isolated tones and in sequences
Authors: Iverson Paul , Krumhansl Carol
Published: October 2005

The interaction of pitch and timbre of tones in isolation and in sequences was studied in a series of experiments. Experiment 1 investigated whether pitch, timbre, and duration interact in the detection of changes between tone sequences. It was found that variations in timbre interfered with the detection of pitch changes, as did the variation of pitch when detecting timbre changes. Changing durations did not interfere with detection of either timbre or pitch changes. Experiment 2 investigated whether the interaction between timbre and pitch is a consequence of changes of relations between tones in the sequences. Contextual variation between sequences in pitch interfered with pitch judgments, but no other contextual interference was found. Experiment 3 measured reaction times of various classifications of isolated tones varying in pitch and timbre and found that pitch and timbre interact. These results indicate that pitch and timbre interact at the level of individual tones, they do not interact at a structural level, pitch is coded relationally, and timbre is not coded relationally. [Work supported in part by a Grant-in-Aid of Research from Sigma Xi, The Scientific Research Society.]
Interaction of timbre and pitch in spectral discrimination tasks using complex tones
Authors: Singh Punita G.
Published: October 2005

The dual nature of pitch (“spectral”/“virtual”) and its relation to such timbral percepts as “sharpness” and “roughness” is explored in a set of experiments designed to ascertain perceptual cues facilitating discrimination tasks with complex tones having flat spectral envelopes. Three types of complexes are employed: (1) harmonic “residue” tones comprising four harmonics, (2) ten-component harmonic and inharmonic complexes with all components shifted from some reference frequency, and (3) ten-component complexes with a single component shifted from its harmonic frequency. Using two-tone sequences as stimuli, listeners are asked to judge if the second tone is (1) the same, (2) higher in pitch, (3) lower in pitch, (4) different in “something else,” (5) different in “something else” and higher, or (6) diffferent in “something else” and lower in pitch than the first (“something else” is taken to be synonymous with “timbre”). For residue tones, data indicate that changes in spectral locus yield changes in timbre. Further, direction of locus change can indicate pitch change, despite little or no change in F0 (0%–2% for 200- and 400-Hz F0). This implies that changes in timbral “sharpness” may be construed as pitch changes, given the absence of other cues. For stimuli in which components are shifted from harmonic frequencies, the unitary sensation of a complex may be replaced by one of multiple sources. Low components (n = 1–3) are more susceptible to being “heard out” as individual entities, while changes in higher components may yield changes in timbre, such as “roughness.” Thus comparisons of components based on factors such as (1) magnitude, (2) location, and (3) direction of spectral change in a sequence can guide grouping operations, in addition to indicating changes in an overall property of a complex as a whole, such as its pitch, timbre, or both of these percepts simultaneously. [Work supported by AFOSR.]
Effect of spectrum and attack properties on the evaluation of concurrently sounding timbres
Authors: Sandell Gregory
Published: October 2005

The perception of concurrently sounding musical timbres underlies musical orchestration, but current knowledge about psychological attributes of timbre emphasizes successive comparison (e.g., similarity, analogy). The psychological attribute “blend” was investigated by having eight listeners use a ten-point scale (a continuum of “separated” to “fused”) to rate all possible simultaneous presentations of synthesized instrumental tones used in an earlier study [J. Grey, J. Acoust. Soc. Am. 61, 1270–1277 (1975)]. Following Grey's three-dimensional model, the data were analyzed for the effects of harmonic synchrony, spectrum content (centroid), and attack characteristics. The latter two attributes showed a significant relationship to blend: in general, “dark” instruments and instruments with quick, clean attacks tended to blend well. The amount of blend reported for such instruments decreased as their pairs increased in brightness or attack length, and pairs of bright or noisy-attack instruments did not blend well. The results found here corroborate some of the recommendations for desirable blends in traditional orchestration manuals. Other aspects likely to pertain to orchestration are interactions between sounds (e.g., masking, acoustic consonance) and the differential salience of timbres (e.g., recognizability, loudness, stream segregation); these will be evaluated.
Quasiperiod of variations in the backscattering and total cross sections of spherical shells
Authors: Kargl Steven G. , Marston Philip L.
Published: October 2005

Quantitative ray methods for leaky surface elastic waves are applied [P. L. Marston, J. Acoust. Soc. Am. 83, 25–37 (1988)] to approximate the quasiperiod of structures in backscattering and total scattering cross sections. The relevant amplitudes in this model contain a Fabry-Perot resonance denominator, which for spheres is [1 + exp(−2πβl + i2πxc/cl)] for the lth class of SEW; x = ka = ωa/c and cl(x) and βl(x) are the SEW phase velocity and radiation damping coefficient. The increment in x between resonances for a given l follows from the condition Δ(xc/cl) = 1. From the Taylor expansion [cl(x + Δx)]−1 ≈ cl(x)−1 + Δxd(cl−1)/dx, the approximate resonance spacing Δx≈cgl/c for weakly dispersive SEW having a group velocity cg (x) is obtained. (This approximation has been noted by others.) The quasiperiod of a broad structure in the total scattering cross section is also approximated as Δxl≈2π[2 cos θl − (c/cl) (π − 2θl)]−1, sin θl = c/cl, when dispersion is neglected. This follows from a condition on the phase increment of the forward amplitude. Both types of variations due to Lamb wave contributions can be seen in exact and ray-synthesis calculations for a spherical shell. [Work supported by ONR.]
New rigorous and ray-acoustic traveling wave formulations for source-excited thin elastic spherical shells immersed in fluids
Authors: Felsen L. B. , Ho J. M.
Published: October 2005

A continuous Legendré transform spanning the polar angle (θ) domain − ∞ < θ < ∞ is employed to derive a rigorous new integral representation for source-excited-time-harmonic pressure fields in the presence of a thin elastic spherical shell immersed in different interior and exterior fluids. The new formulation identifies directly the traveling waves with their multiple encirclements of the shell by extending the θ domain from its conventional 0⩽θ⩽π range with periodicity constraints into the unbounded (multisheeted) domain without these constraints. The new formulation also systematizes the treatment of spherical caps and other truncated sections. Periodicity for the closed shell is recovered by summing contributions from an infinite array of image sources located in the “non-physical” portion θ < 0 and θ > π of the angular space. The rigorous solution is obtained by synthesis over a complex spectral continuum, and various alternative representations are derived from it. Special attention is given to rigorous high-frequency asymptotic forms that describe the wave phenomena in terms of incident and geometrically reflected ray fields, and also in terms of surface guided ray fields. The latter are excited by phase matching of the incident ray field to the traveling wave modes in the shell, and they reach the observer by phase matched detachment [see, also, P. L. Marston, J. Acoust. Soc. Am. 83, 25–37 (1988)]. The phase matching applies to directly excited leaky waves as well as to waves that decay initially into the fluid; the latter are excited from an exterior source by evanescent tunneling. [Work supported by the Office of Naval Research and David Taylor Research Center.]
The optical illusion of sound
Authors: Swartz Bradford L.
Published: October 2005

The graphic depiction of sound is not a straightforward matter. Stationary representations using lines, waves, or circles leave the notion of movement (either particle or energy) and pressure fluctuations within the wave to the reader's imagination. The common sinuous wave shape evident in most all sources of study originates from Lissajous, and resulted from his transfer of tuning fork motions into a point of reflected light that traversed a screen. Lissajous performed his experiments with finely focused electric lamp light, tuning forks, and mirrors. Similar experiments can be replicated today using laser light and audio speakers. Thus one can more readily gain an appreciation for the familiar wave pattern representing sound, and see by the simple instruments used to create it, how this waveform is not a true depiction of the shape of sound waves in a medium. It is in reality a displacement versus time graph. By using multiple forks or speakers in opposing planes and tuned to various frequencies, the creation of the single- and multiple-ellipse Lissajous patterns that are often produced on an oscilloscope screen can also be accomplished.
Acoustic surface shape resonances
Authors: Maradudin Alexei A.
Published: October 2005

Acoustic surface shape resonances are vibrational excitations that are localized in the vicinity of an isolated protuberance or indentation on the otherwise planar, stress-free surface of a semi-infinite elastic medium. The protuberance may be fabricated from the same material as the substrate, or from a different material. In general, there is an infinite number of such resonances associated with a given surface perturbation. Their frequencies are discrete because of the loss of translational symmetry caused by the surface perturbation; they are complex because they overlap the range of frequencies allowed the vibrations of the substrate, into which they can decay; and they depend on the shape of the protuberance or indentation, and on the relation of the material properties of the protuberance (mass density, elastic moduli) to those of the substrate. Methods for calculating the frequencies of the acoustic surface shape resonances associated with protuberances or indentations of simple forms are described. Results are presented that suggest that acoustic surface shape resonances can be studied experimentally by the scattering of acoustic waves from the structure supporting them.
Localized wave transmission physics and engineering
Authors: Ziolkowksi Richard W.
Published: October 2005

Exact solutions of the scalar wave equation that describe localized transmission of wave energy will be reviewed briefly. These acoustic solutions can be optimized so that they are localized near the direction of propagation and their original amplitude is recovered out to extremely large distances from their initial location. Pulses with these very desirable localized wave transmission (LWT) characteristics may have a number of potential applications in the areas of directed energy applications, secure communications, and remote sensing. The following will be shown. (1) The underlying physics of the LWT effect is closely connected to the additional degree of freedom obtained by coupling the usually disjoint portions of phase space. Position and frequency spectra at different locations in the aperture are different but are highly correlated so that the effective frequency of the aperture is higher than expected and the resulting pulses from these aperture sources reconsitute the frequency content of the LWT packet, hence its pulse shape, as it propagates. (2) The LWT solutions do not violate any known uncertainty relations and satisfy a generalized full phase-space uncertainty relation. (3) The independently addressible element LWT arrays can be designed to dramatically outperform conventional cw-driven apertures. This includes pulse shape, peak amplitude, and energy fluence. New acoustic experimental data that corroborate these theoretical results will also be presented. [This work was performed by the Lawrence Livermore National Laboratory under the auspices of the U.S. Department of Energy under Contract No. W-7405-ENG-48.]