bluuebunny/crossref_metadata_2025 huggingface dataset:
Digitized acoustic radar signals
Authors: Werner Ch.
Published: October 2005

To study the atmospheric structure, a data acquisition system for the monostatic acoustic radar was developed. It allows the direct comparison to in situ and other remote measurements. The data acquisition system consists of a transient recorder, an interface and a magnetic tape recorder. The transient recorder digitizes and stores the signal. The interface, containing an adjustable calender and a real time digital clock is connected between the transient recorder and the magnetic tape recorder. The tape recorder writes data in an IBM compartible mode. A special computer program to get lines of the same signal amplitude is used. There is more detailed information as the normal display unit shows. Examples demonstrating the possibilities of this digitized system are given.
Sonar velocity resolution with a linear period modulated pulse
Authors: Altes Richard A.
Published: October 2005

For linear period modulated (LPM) signals, the phase of the echo from a moving point target contains information about range rate. When an LPM signal is transmitted, a Doppler insensitive sonar processor is obtained by following a matched filter with an envelope detector. Alternatively, the same echo can be used for accurate velocity measurements by implementing a phase-sensitive detector. A receiver that envelope detects matched filter responses and also estimates the phase of an echo can therefore be both Doppler tolerant and Doppler resolvent. The relative simplicity of Doppler tolerant receivers is again realized when Doppler-sensitive processors are implemented for LPM echoes; if LPM is used, only two matched filters are needed to implement a sequence of M velocity hypotheses, rather than a bank of M different filters. LPM is thus an excellent signal format not only for Doppler invariant range estimation and initial detection, but also for fine velocity resolution with minimum receiver complexity. These observations may imply some hitherto unsuspected echolocation capabilities for bats that use LPM waveforms. The existence of velocity induced phase transformations in LPM signals also leads to a new synthetic aperture sonar technique. [Work supported by Naval Coastal Systems Laboratory.]
Airgun array design techniques
Authors: Teer T. L.
Published: October 2005

Since airguns were introduced to the oil exploration industry in the late 1960's, they have replaced dynamite as the primary acoustic source for marine seismic profiling. Arrays of airguns compare favorably in the seismic bandwidth to dynamite for depth of penetration and resolution. Multi-element arrays capable of peak-to-peak pressures exceeding 50 bars (at 1 meter) are presently being used in the industry. Independent control of chamber pressure and volume, operating depth and distance between airguns allows the design of airgun arrays which can meet particular resolution or depth of penetration objectives. Array design techniques are reviewed and data from a Shell array is presented to illustrate the degree to which success in meeting the design objectives of a high peak pressure, a high primary-to-bubble ratio and broad bandwidth is achieved. Far-field signatures from the array are shown to possess the necessary shot-to-shot stability required of modern signal enhancement techniques when proper care is taken in deploying and towing the array. Boundary effects (ghosting) are shown to reduce shot stability. Some mechanical deployment and towing considerations are discussed.
A computer model of the explosive acoustic source
Authors: Lauer R. B. , Maples L. C.
Published: October 2005

Explosives are widely used as broadband acoustic sources, but their spectrum characteristics are not adequately defined. A comprehensive experimental evaluation is not feasible economically and the problem can be solved only by a model, supported by a limited critical experiment. A computer model of the source waveform, using only analytic functions, is at last available. The model, known as DIAL-A-SUS, readily provides the desired spectrum for any charge size and detonation depth by Fourier transform techniques. Certain amplitude and time characteristics of the waveform, which are defined in terms of the source weight and depth, are used to evaluate the parameters of the model's analytic functions. The model is ideally suited to study the sensitivity of the spectrum characteristics to variations in the critical parameters of the waveform and determine the important factors requiring further experimental evaluation. The results and significance of such studies are discussed, with experimental implications, and model predictions are compared with measurements. An exciting spinoff is the characterization of the explosive as a multifrequency, narrow-band (e.g., 1 Hz) source, with important potential applications. [Work supported by NUSC.]
Low-frequency underwater explsoive source levels
Authors: Hughes R. C.
Published: October 2005

New experimental measurements of underwater source levels for explosive charges at depths ranging from 20 to 200 m are reported. The explosives used include 45 g (1.6 oz) TETRYL primer charges, 454 g (1.0 lb) bare TNT charges, 31 g (1.1 oz) TETRYL SUS charges and 816 g (1.8 lb) TNT SUS charges. Underwater pressure measurements were made at a nominal distance of 1500 m. For frequencies below approximately twice the bubble frequency, the experimentally measured acoustic energy spectra show a remarkable correspondence when they are appropriately scaled for detonation depth and for charge weight. A universal model of the low frequency energy spectra of TNT and similar explosives, which can be scaled to different charge weights and different detonation depths, shows considerable promise.
Nearfield data from small and large explosions II: Spectral analysis
Authors: Gaspin Joel B. , Goertner Jean A.
Published: October 2005

The problem of the definition of the spectral source energy levels for the joint NSWC-NRL experiment, discussed in the previous paper, will be treated. Since surface reflections occur in the pressure records, the removal of surface reflection effects is necessary in order to obtain the source spectra. The spectral modulation pattern due to the surface reflections are different at the two sensor depths. This makes these data well suited for the examination of different techniques for demodulating the spectrum. Both time and frequency domain deconvolution and various smoothing techniques are examined, and their effects on the spectral energy in 13-octave bands are demonstrated. Certain computational difficulties arising in the frequency-domain deconvolution are avoided by use of time-domain deconvolution. In situations where the surface reflection is not acceptably coherent, or if cavitation occurs, simple deconvolution is not applicable, and special techniques must be used. The dependence of the source spectra on sensor type is also examined. The source energy levels derived from the 1000-lb-charge data are found in good agreement with the scaled-octave band source level curves of Christian [J. Acoust. Soc. Am. 42, 905(L) (1967)], which were derived from much smaller charges. [This work was funded by DNA.]
Flexural waves generated in glass fibers by fracture
Authors: Simpkins P. G.
Published: October 2005

In 1948 Davies noted that flexural waves could be created in a rod by an unsymmetrical lead applied to one end. Miklowitz applied this observation to the hypothesis that during tensile fracture a moment pulse is exerted on the specimen which generates a flexural wave. He went on to suggest that this flexural wave could augment the stress induced in the rod by a reflected longitudinal wave when the two were coincident; thereby causing the secondary failures observed in brittle materials. The nature of the failure in fused silica fibers qualitatively suggests that it alters as the strength increases. While the low-strength fibers simply fracture at one point, the high strength fibers (UTS > 2.8 GN/m2) appear to disintegrate into a particulate cloud. Recent photographic studies show that a flexural wave emanates from the primary fracture. When the strength is sufficient this flexural wave causes secondary explosive failures at other points along the fiber. These result in a particulate cloud of debris as the fiber appears to be progressively pulverized. Observations of the flexural wave amplitude show it to be a strong function of fiber strength. The observed group velocity of the flexural waves compares favorably with the classical time harmonic theories and the first-mode Timoshenko approximation.
Vibrations of two concentric cylindrical shells containing a viscous fluid
Authors: Yeh T. T. , Chen S. S.
Published: October 2005

This study was motivated by the need to design the thermal shield in reactor internals to avoid detrimental flow-induced vibrations. The system component is modeled as two coaxial shells separated by a viscous fluid. In the analysis, Flugge's shell equations of motion and linearized Navier-Stokes equation for viscous fluid are employed. First, a traveling-wave-type solution is taken for shells and fluid. Then, from the interface conditions between the shells and fluid, the solution for the fluid medium is expressed in terms of shell displacements. Finally, using the shell equations of motion given the frequency equation, from which the natural frequency mode shape, and modal damping ratio of coupled modes can be calculated. Some important conclusions are as follows: (1) There exist structural modes and acoustic modes. If the structural natural frequencies and mode shapes are of primary interest, the fluid may be considered inviscid and incompressible. (2) There exist out-of-phase and in-phase modes. The lowest natural frequency is always associated with the out-of-phase mode. (3) The lowest natural frequency of coupled modes are lower than the uncoupled modes. (4) The fluid viscosity contributes significantly to the modal damping.
On the forced responses of damped cylindrical shells filled with pressurized liquid
Authors: Lu Y. P.
Published: October 2005

An analytical formulation for the forced vibratory responses of a pressurized liquid-filled cylindrical shell having a number of mass segments adhered to it by a viscoelastic material is presented. The mass segments are discretely distributed around the outer circumference at an arbitrary section of the shell, and the excitation is a concentrated vibratory radial force located on the surface of the shell. The end conditions of the shell are assumed simply supported. The liquid is considered as a compressible and inviscid fluid. The viscoelastic material is assumed incompressible. The interaction between the shell and the attached mass segments, and the interaction between the shell and the enclosed liquid are taken into consideration. The driving point mechanical impedances at a location midway between two mass segments and half the distance along the length for a given damped system with or without pressurized water are given. These solutions are compared and discussed, respectively, with those of an undamped shell. The responses of a discontinuously constrained damped ring configuration without liquid enclosed and the frequencies of an undamped shell with pressurized water filled, which are the special cases derived from the analysis presented, are, respectively, compared very well with the available experimental data.
Response of nonclassically damped systems to random excitation
Authors: Michalopoulos C. D.
Published: October 2005

The response to stationary random excitation of multidegree-of-freedom systems which do not possess classical normal modes is obtained using a matrix perturbation technique. It is assumed that the damping matrix [C] is such that the off-diagonal elements of the matrix [Φ]T[C][Φ], where [Φ] is the modal matrix associated with the undamped system, are small compared to its diagonal elements. General expressions for the correlation and spectral density matrices of the response are given. Examples of systems with two degrees of freedom are presented considering first and second orders of perturbation.
Identification of natural frequencies and modal damping ratios from response data
Authors: Michalopoulos C. D.
Published: October 2005

Response data are processed in the frequency domain to determine natural frequencies and modal damping ratios of a multidegree-of-freedom system. The excitation can be either deterministic or a random function of time. When the time history of the excitation is now known, the method requires its power spectral density to be flat. An iterative scheme is employed to minimize a mean-square error function between the theoretical and measured power spectral densities of the response. In processing random response data, special steps are taken to minimize the statistical error. [Work supported by NASA.]
Use of LPC synthesized speech in an instructional application
Authors: Sanders W. R. , Laddaga R.
Published: October 2005

Synthesized speech was recently introduced into Stanford University's “Introduction to Logic” course. For five years, the course has been taught entirely by computer using teletypes to individually present all lesson material and on which all exercises are worked. Last spring, 28 to 70 students took 30% of the course on CRT-display terminals with earphones. They heard recorded messages encoded using autocorrelation LPC techniques and synthesized by a computer built at the Institute. The computer has 48 output channels and can speak independently on 16 channels at a time [Sanders, Benbassat, and Smith, “Speech synthesis for computer assisted instruction: the MISS system and its applications,” ACM SIGGUE Topics 2, 200–211 (1976)]. The results of a questionaire indicated that more than two thirds of the audio students preferred the audio version. All of them felt the sound was clear; most preferred the speech presented faster than the original recording. Examples of the audio messages will be presented. [Supported by NSF Grant EPP-74-15016.]
How to put one person's tongue inside another person's mouth
Authors: Ladefoged Peter
Published: October 2005

An algorithm has been devided for recovering the shapes of the tongue involved in producing different vowels simply from a knowledge of the first three formant frequencies. The calculated tongue shapes may then be displayed inside a standard vocal tract shape. The algorithm has been tested by recovering the tongue shapes of a wide variety of vowels spoken by a number of different speakers and comparing them with the original x rays. The exact shapes of low vowels, such as [æ, α] are not recovered as accurately as those of other vowels, but the reconstructed tongue shapes of most men can be fitted very appropriately into a standardized diagram of the organs of speech based on x rays of the author (Peter Ladefoged). There is inevitably some distortion of the organs when a woman's tongue is put inside the author's mouth, because of the notable anatomical differences between men and women. The algorithm gives good approximations even when applied to speakers of other languages, such a Russian, Japanese, and Ngwe.
Articulatory compensation: A study of ambiguities in the acoustic-articulatory mapping
Authors: Atal B. , Chang J. J. , Mathews M. V. , Tukey J. W.
Published: October 2005

We present a method of constructing various vocal-tract shapes having identical acoustical characteristics. A 20-dimensional vocal tract model is used to compute the frequencies, amplitudes, and bandwidths of the first five formants. The dimensions consist of 20 uniformly spaced cross-sectional areas of the vocal tract. The model includes losses due to finite wall impedance, glottal leakage, friction, heat conduction, and radiation. Articulatory regions (fibers) are computed in which the frequencies and amplitudes of the first three formants remain constant. In general these fibers contain 14 dimensions, not all of which provide physiologically reasonable area profiles. A theory for selecting among the dimensions within a fiber according to maximum physical smoothness of the vocal tract is presented. Application of the theory to the 14-dimensional data indicate that 2 or 3 of the 14 dimensions correspond to physically realizable perturbations in the vocal tract and that along these dimensions changes in the vocal-tract shape may be made without changing the three formant frequencies and amplitudes. Almost no audible change in the sound is produced by these articulatory variations. Examples of synthetic speech illustrating these results will be played.
Numerical inversion of the transformation from articulatory to acoustic parameters in the vocal tract
Authors: Chang J. J. , Mathews M. V. , Atal B. , Tukey J. W.
Published: October 2005

A method is presented for determining the parameters of an articulatory model corresponding to a given set of formant frequencies. A four-parameter articulatory model of the vocal tract was used to define the vocal-tract shape. The articulatory parameters were (a) location of constriction, (b) area of constriction, (c) area of mouth opening, and (d) length of the vocal tract. The model included losses due to finite wall impedance, glottal leakage, friction, heat conduction and radiation. Frequencies, amplitudes, and bandwidths of five formants were computed for each of 30 720 vocal-tract shapes obtained by uniform sampling along each of the four articulatory dimensions. The data was sorted according to the first three formant frequencies. The articulator positions yielding given values of these formant frequencies can be looked up in the sorted data. One-dimensional fibers (curved lines) were computed along which the articulator positions can vary without producing changes in the three formant frequencies. These fibers span large distances in the articulatory space. Sounds produced by various articulations along a fiber were computed and were compared by informal listening tests.
A better LASS
Authors: Rice D. Lloyd
Published: October 2005

The first Line Analog Speech Synthesizer (LASS) system used at UCLA modeled a vocal tract consisting of 18 sections, each with an area and an acoustic loss factor that can be specified independently. For computational reasons each section had to be the same length, which was obviously a disadvantage. Another weakness was that the radiation impedance at the lips was modeled in a somewhat simplistic manner. Both these disadvantages have now been overcome, without losing the advantage of a model in which the losses can be specified at each point along the tract. A curve is interpolated between specified points along the tongue, converted to areas and smoothed to determine 18 evenly spaced sections. The losses at the lips are now modeled by a realistic resistive-inductive load. The formants calculated using this model have been compared with those calculated by the previous model, and with those calculated by a model described by Fant. There are still differences between calculated and observed formant frequencies. We do not know how to decide whether the difficulty in reproducing observed formant frequencies is due to inaccurate estimates of the acoustic losses along the tract or errors in the conversion from sagittal dimensions to cross-sectional areas, or some other factors.
Acoustic effects of longitudinal displacement in vocal-cord vibration
Authors: Ishizaka K. , Flanagan J. L.
Published: October 2005

Previously we modelled vocal-cord vibration by a nonlinear mechanical oscillator composed of two stiffness-coupled masses. The oscillator is driven by a function of intraglottal pressure and subglottal pressure, and the coupled masses are permitted one-dimensional displacement transverse to the direction of glottal flow. This formulation produces realistic self-oscillatory behavior, and duplicates characteristic phase-differences in cord motion and acoustic interactions with vocal tract and subglottal system. Through approximating calculations, we estimated that the longitudinal component of real cord motion (along the direction of glottal flow) has second-order influence upon glottal flow. In this paper we examine more rigorously this earlier estimate. We modify the two-mass model for an additional (longitudinal) degree of freedom in cord displacement. Oscillator and coupling parameters remain as before, but the oscillator is driven by functions of both transglottal pressure and intraglottal pressure. As a result of interaction with the subglottal system, we observe a microstructure in subglottal pressure variation. The microstructure is excited mainly by vocal-cord closure, and is consistent with measurements on human phonation. Also, we compute dynamic two-dimensional trajectories for the cords (masses), and find that they correspond with observations made on natural larynxes. We conclude that the longitudinal component of displacement is not essential for realistic self-oscillation of the cord model. Finally, we synthesize voiced sounds with the model, and find that the glottal flow component deriving from longitudinal displacement is physically and perceptually of second-order importance.
Airports and land use planning
Authors: Starley S. E.
Published: October 2005

The EPA has developed an Airport Noise Regulatory concept, including an impact analysis methodology and a readily understandable format for the presentation of results. The regulatory process provides a means of reducing and/or preventing aviation noise impacts via a balanced mix of aviation actions and lead use controls. The analysis methodology which EPA has developed is based upon a comparison of aviation noise and the background levels which are due to all sources other than aviation. Hence, the noise impacts of multi modal systems can be evaluated. Noise criteria have been developed for the classification of airports, by severity of impact, and for goals necessary to protect the public health and welfare. The regulatory process was designed to insure participation of the airport proprietor, local governments, and the general public. Each airport noise abatement plan will be subjected to public scrutiny, in an open hearing, as well as to review by the Federal Aviation Administration, with consulation by the EPA. The process results in an airport operating pier for noise abatement which is basic to land use planning.
Call discrimination as a function of playback intensity in the green treefrog (Hyla cinerea)
Authors: Gerhardt H. Carl
Published: October 2005

The intensity at which female green treefrogs (Hyla cinerea) detected spectral and fine-temporal changes in synthetic vocal signals varied. Given a simultaneous choice between a standard call consisting of three components (0.9 + 2.7 + 3.0 kHz) of equal relative amplitude and a call in which the two high-frequency components were attenuated by 15 dB, females consistently chose the standard call at 65, 75, and 85 dB but failed to discriminate at 60 dB. The same results were obtained when females were given a choice between the standard call and one in which the 300/sec beating was lacking (i.e., 0.9 + 3.0 kHz). Females discriminated in favor of the standard call at 60, 65, and 75 dB when the low-frequency component of the alternative stimulus was attenuated by 10-dB relative to the two high-frequency components. These results will be discussed in relation to response properties of the peripheral auditory system and to the ways in which intensity and spectral properties of natural and synthetic calls change as a function of distance, habitat, and elevation of the sound source. [Work supported by NSF.]
Critical period for the development and maintenance of binaural interaction
Authors: Silverman M. S. , Clopton B. M.
Published: October 2005

A determination of the critical period for the development and maintenance of binaural interaction was made by unit recording at the inferior colliculus of neonatal and sound deprived rats. Collicular units are activated by clicks delivered to the contralateral ear. An indication of binaural interaction in normal adult rats is the suppression of this activity by ipsilateral clicks that are of equal or greater intensity. Such binaural interaction was evident in the unit activity of young rats during the first week after the onset of hearing. Monaural deprivation beginning at 10 days after birth (DAB) and continued for three to five months abolished binaural interaction. The same duration of deprivation had much less effect if begun at 30 days and no effect if initiated at 60 days. The result indicate that the essential connections for binaural interaction are genetically predetermined and present at the onset of hearing at 10 DAB. The critical period for experiential validation of binaural interaction begins at the onset of hearing and extends up to 60 DAB. These conclusions should be considered specific to the rat. [Work supported by NIH Grant NS-13052.]
Establishing noise criteria for residential living in areas surrounding commercial aviation airports
Authors: Mabry J. E.
Published: October 2005

Results are provided which contribute to establishment of noise levels that are compatible with residential living activities. Community-noise simulation systems were placed in the homes of 24 families that were not impacted by airport noise. Each family was exposed to four different airport noise conditions of one week duration. Interference with daily living activities and annoynace responses to the four conditions were obtained. Some results and conclusions are as follows: (1) Noise from other sources within their homes bothered or annoyed the participants as much or more than the flyover noises. (2) Behavioral awakening did not occur for many participants exposed to night flyovers peaking at approximately 70 dBA. It was concluded that the 10 dB penalty applied to night flyovers for NEF and Ldn is too large. (3) For the mix and number of aircraft utilized, it was concluded that a NEF of 32 to 33 is compatible with indoor residential living. [FAA-RD supported.]
The annoyance and loudness of magenta noise
Authors: Burns E. M. , Turner C. W. , Ward W. D.
Published: October 2005

The relative “annoyance” and “loudness” of magenta noise (a broadband noise whose octave-band spectrum falls off at the rate of 5 dB per octave) and truncated-magenta noise (the high-frequency portion of the magenta noise) were determined using a 2AFC procedure. For moderate intensities (90 dBA, magenta) the relative annoyance and loudness were qualitatively in the direction predicted by the PNdB and Sone scales, respectively. That is, the magenta noise was judged to be significantly more annoying and louder than the truncated-magenta noise, for the condition in which the energy in the frequency region common to both noises was equal. However, for the same condition at higher intensities (110 dBA, magenta) the truncated-magenta noise was judged significantly more annoying, while the magenta was judged louder. Thus at high intensities we have the paradoxical situation in which the deletion of a major portion of the energy from the spectrum results in a considerably more annoying noise, a result which is not predicted by PNdB. Possible explanations for these results, along with their relevance to aircraft noise, are discussed.
Noise-induced threshold shifts in the Mongolian gerbil
Authors: Ryan Allen
Published: October 2005

Mongolian gerbils were trained in a shuttlebox avoidance task and their auditory thresholds determined. Groups of six subjects each were exposed to a two-octave (1414–5656 Hz) band of noise for one hour at intensities of 100, 110, and 120 dB SPL. Thresholds between 0.1 and 16.0 kHz were determined 0.5, 3.0, 6.0, and 12.0 hours after exposure, and daily for 27 days. Final threshold determinations were made at least two months postexposure. A temporary threshold shift (TTS) was observed in all groups, the extent of which increased with increasing intensity of exposure. Recovery of thresholds followed an exponential course, and for extensive TTS lasted throughout the 28-day observation period. In some cases additional recovery was noted upon final threshold determination. No significant permanent threshold shift (PTS) was seen in the 100-dB exposure group, but PTS did occur in the 110-and 120-dB groups. PTS was limited to a relatively narrow band of frequencies, and exhibited a half-octave shift toward higher frequencies. In terms of both TTS and PTS, the gerbil appears to be less sensitive to noise than chinchillas exposed to the same stimulus. [Work supported by NINCDS and the Medical Research Service of the Verterans Administration.]
Discrimination of pure-tone intensities by the California sea lion
Authors: Moore P. W. B. , Schusterman R. J.
Published: October 2005

The ability of the California sea lion (Zalophus californianus) to detect intensity differences of 16-kHz pure-tone pulses was determined in a yes-no signal detection task with trial by trial feedback, using a modified method of limits similar to the “staircase method.” Additionally, three prethreshold experiments were performed to determine the effect of stimulus presentation strategy on discrimination ability. Results of the prethreshold experiments suggest that a “collapse” method (i.e., symmetrically reducing the intensity of the louder tone pulse in a pair and increasing the intensity of the softer) resulted in smaller threshold estimates than either an ascending or descending staircase method. Under the collapse method, a difference threshold estimate of 3.19 dB was found for 16-kHz pure-tone pulses. These results support a previous hypothesis based of behavioral underwater sound localization experiments and sound-skull measurements by Moore and Au [J. Acoust. Soc. Am. 58, 721–727 (1975)]. These experiments also indicated that a sea lion which persistently biased its responses by making a low rate of false alarms in an absolute threshold task [Schusterman, J. Acoust. Soc. Am. 55, 845–848 (1974)] did not bias its responses in a difference threshold task.
Perception of conspecific communication signals by Japanese monkeys (Macaca fuscata)
Authors: Beecher M. D. , Zoloth S. R. , Petersen M. R. , Moody D. B. , Stebbins W. C.
Published: October 2005

A method for assessing perception by monkeys of their own acoustic commmunication signals is described. Japanese macaques were trained with a positive-reenforcement conditioning procedure to discriminate between two classes of conspecific vocalizations, “smooth early highs” (SEH) and “smooth late highs” (SLH) [classification of Green, Primate Behavior 4, 1–101 (1975)]. Subjects contacted a response disk to initiate a series of stimuli from the SLH class. A trial was defined as an insertion of a SEH stimulus into the series in place of a SLH. A key release on such a trial (correct detection) was reenforced with food. With this method we are examining a variety of perceptual questions regarding the animal's acoustic communication system. For example, do the animals in the laboratory task classify these calls as they do in the field? Do they perceive acoustic intermediates as true intermediates or as members of one or the other signal class in the manner that humans categorize their own speech sounds. [Supported by NSF Research Grants BMS 74-20050 and 5-27092, NINCDS Program Project Grant NSO5785, and NIGMS Training Grant GM-01789.]
Discrimination of synthetic prevoiced labial stops by infants and adults
Authors: Eilers Rebecca E. , Wilson Wesley R. , Moore John M.
Published: October 2005

Individual data were obtained from eight normal 6-month-old infants on discrimination of eight synthetic labial stop pairs differentiable on the basis of voice onset time. All infants were presented with the pairs pha (+40) versus pha (+70), pha (+40) versus pa (+10), ba (+10) versus ba (−20), ba (−20) versus ba (−50), ba (−20) versus ba (−60), ba (−30) versus ba (−60), pa (0) versus ba (−30) and pa (+10) versus ba (−60). Discrimination was assessed for each infant on each pair using the VRISD paradigm (Visually Reinforced Infant Speech Discrimination). Adult discrimination was assessed for the same pairs in the same manner. With few exceptions, infants presented no evidence of discrimination of pairs other than the pair pha (+40) versus pa (+10) despite the fact that these infants were successful in the same paradigm with a variety of subtle natural speech contrasts. Contrary to the hypothesis that age restricts discrimination capacities, adults were far more successful than infants in discriminating the non-English contrasts. Individual data will be presented to illustrate these points and the use of synthetic speech stimuli in infant research will be discussed in light of these data. [Works supported by NIH-NICHD, HD-3-2793.]
Vowel duration as a cue for consonant voicing?
Authors: Fruin C. W. , Bischoff D. M.
Published: October 2005

Several studies state that the duration of the vowel preceding the consonant is a significant cue to the voicing characteristic of that consonant. The CVC's used in these studies have generally been prepared on the Pattern Playback. The present study was designed to examine vowel duration as a consonant voicing cue in naturally produced speech. Several CVC's were recorded by both authors as stressed citation forms. Vowel durations were measured and parts were removed from: the endmost section of the vowel, the middle of the vowel, or the end and the middle of the vowel. At least for these stimuli produced by these speakers, no clear relation is seen between the length of the preceding vowel and the perception of the voicing characteristic of the final consonant.
Perceptual adaptation to the duration of vowels preceding stop consonants
Authors: Williams Paul D. , Sharf Donald J.
Published: October 2005

The affect of selective adaptation on the ability to identify stop consonants as being either voiced or voiceless solely on the basis of preceding vowel duration was tested. Stimuli consisted of computer-modified, real-speech sounds edited from a single utterance of /æ d/ which varied in duration in 25-msec steps. Each subject classified recorded, random orderings of the stimuli as either AT or AD before and after periods of repetitive listening to long and short adapting stimuli. Calculated phonetic boundaries showed a mean shift toward the category of the adapting stimulus in both conditions, indicating a reduction in sensitivity for that category. The findings of independent, adaptable feature detectors for the voiced and voiceless categories and a greater resistance to adaptation for the voiced feature were comparable to results reported for adaptation studies employing voice onset time. [P.D. Eimas and J.D. Corbit, Cognitive Psychology 4, 99–109 (1973); P.D. Eimas et al., Perception and Psychophysics 13, 247–253 (1973)].
Formation of auditory descriptions
Authors: Bregman A. S.
Published: October 2005

In natural environments, sounds from many sources are mixed at the ear. The auditory system must parse the input to recover separate descriptions of the original sources. In addition to mechanisms for sound localization, it employs “compositional heuristics” for segregating and combining acoustic components to create a perceptual description of simultaneously active acoustic sources. The clues in the acoustic signal that indicate that components should be assigned to a single source include spectral similarity of successive components, slow and continuous changes, information for “temporary occlusion,” and, finally, synchronous and parallel changes of simultaneous components. Descriptions of multiple sources are built up simultaneously, and these descriptions compete for acoustic elements until elements are assigned as belonging to specific sources. The perception of certain qualitative features, such as timbre, depend on the perceptual grouping of components. Other perceptual consequences, such as the number of sources perceived, and the ability to judge the temporal order of elements, or to recognize patterns, also depend upon the preliminary parsing of the signal as controlled by the “compositional heuristics.” To demonstrate these effects, experiments will be described and recorded stimuli will be played.
Synthetic aperture arrays in sonar and seismic profiling
Authors: Baggeroer Arthur B.
Published: October 2005

The impressive results of applying the synthetic aperture concept in radard has stimulated interest in employing it for mapping the seabed and the seismic structure beneath it. In sonar, several studies have proposed modifications of side scan systems to implement this concept; and in seismics, it is embodied in the operation of migration. These two applications have evolved independently; nevertheless, it appears that each could benefit from the other in solving related problems, particularly with regard to the signal processing. In this presentation we indicate areas of mutual interest in the signal processing of these two applications of the synthetic aperture concept by exploring the similarities and differences from a general array processing perspective.
Analysis of passive synthetic-aperture arrays for nearfield measurement of large moving noise sources
Authors: Rudgers Anthony J.
Published: October 2005

Passive synthetic-aperture arrays can be used to measure, in the nearfield, the farfield characteristics of large moving noise-radiating sources. A planar synthetic-aperture array is created when such a source moves at uniform speed past a line receiving array. Trott-array shading is applied to the receivers comprising the line array. The time-varying signals, which are generated by the source at each receiver, are likewise processed using Trott-array shading functions. This results in a virtual Trott receiving array many times the length of the moving source. The characteristics of the virtual array are equivalent to those of a physical array of the same size, with the noise source fixed within its plane-wave region.
Numerical investigation of an acoustic slow waveguide
Authors: King Benson Jay
Published: October 2005

The effect of a cylinder of material whose sound speed is less than that of water on the radiation beam pattern of an omnidirectional sound source is investigated. Specifically a point source is placed near one end of the cylinder. This cylinder serves as an acoustic waveguide and enhances the radiation from the source in the direction of the waveguide. The waveguide is mathematically modeled by use of the surface Helmholtz integral equation together with the requirement of continuity of acoustic pressure and normal velocity on the surface common to the waveguide and surrounding water as reported by P.H. Rogers and W.J. Trot: [J. Acoust. Soc. Am. 56, 1111–1117 (1974)]. The integral equations are solved numerically. Beam patterns and directivity gains for waveguides of various lengths and diameters are calculated. Included are results for composite waveguides consisting of an inner core of one material surrounded by an outer shell of another material. [Work supported by Naval Electronic Systems Command, Code 320.]
Lecture room in Hunt Hill (formerly the Fogg Art Museum), Harvard University—An historical review
Authors: Wetherill Ewart A.
Published: October 2005

At the 86th meeting of the Acoustical Society of America, the author reviewed the history of Hunt Hall—the building where W.C. Sabine began his studies of architectural acoustics—and reported his acoustic measurements of the lecture room immediately before its demolition in June 1973. Subsequent investigation has answered some unresolved questions and has yielded further information, summarized here.
Image method for efficiently simulating small-room acoustics
Authors: Alien J. B. , Berkley D. A.
Published: October 2005

Image methods have been successfully used in the analysis of the acoustic properties of enclosures for the purpose of architectural design. In this paper we disuss use of image techniques for simulating the impulse response between two points in a small room. The resulting response is then convolved with any desired input signal and the reverberated output used for signal processing or psychoacoustic studies. The entire process is carried out using a digital computer so that bide ranges of parameters ran be simulated with perfect control over experimental conditions.
Sound transmission between absorbing parallel planes
Authors: Hurst C. J. , Mitchell L. D.
Published: October 2005

Partially absorbing parallel surfaces can be dominant acoustical feature of many rooms. The sound fields in such rooms are not diffuse, which causes difficulty in predicting sound pressure levels. A method is developed for predicted sound pressure levels in these rooms. It rests on the prediction of sound pressure levels caused by a nondirectional source of known sound power radiating between absorbing parallel planes. The development proceeds from a geometrical acoustics viewpoint. Good correlation has been found between predicted and measured levels in existing rooms having simple geometries. [Work supported by NSF.]
Acoustic environment for audio teleconferencing
Authors: Betres R.
Published: October 2005

In contrast to handset telephony, the acoustic environment plays an important role in handsfree (loudspeaking) telephony and more significantly in audio teleconferencing. With increasing distance between talker and telephone terminal, the effects of room resonances, reverberence and background noise on the quality of speech telecommunication became more and more critical. The range of variation of these parameters in executive offices and conference rooms was investigated in this survey. The results indicate a common shortcoming in most of these locations: a sound-absorbing ceiling and hard, parallel walls. The effectiveness of directional microphones in improving the situation was subjectively tested in simulated environments. The results give an indication of the limits of acceptability of the acoustic environment. A weighting system for ranking the audio quality of a conference room is proposed, taking into consideration the above mentioned factors and results. [Work supported by Bell Canada.]
Acoustic design of a multipurpose hall with variable reverberation units and a movable partition unit
Authors: Furukawa S. , Yamamoto T.
Published: October 2005

The Orange Hall of the Kochi Prefectural Culture Center is a two-floor-type multipurpose auditorium of 10 620 m3 in size, provided with 1500 seats. This hall is designed so that the main and the second floors can be divided with a movable partition unit, which is housed in the ceiling when not in use, as required by the type of presentation. It can be used in three ways: as the full-size hall, as a medium-size hall using only the main-floor auditorium with 1000 seats, and as a meeting room using only the second floor with about 350 seats. In order to furnish the hall with room acoustic conditions suitable for diversified use, suspended ceilings have been used in the auditorium as variable reverberation units. When these are lowered, the increase in the absorption power of the hall is about 150 m2 at 500 Hz, and the reverberation time is reduced to about 0.3 s. According to the results obtained, the reverberation time at 500 Hz in the empty hall is 1.7 s when the movable stage sound reflectors are used, 1.45 s when the suspended ceilings are lowered, and 1.15 s with the stage curtains lowered. Other characteristics have been realized as expected in the design.
Design considerations for high fidelity hearing aids
Authors: Killion Mead
Published: October 2005

Guidelines for high-fidelity hearing aid design can be deduced from known objective and subjective factors. These factors include head diffraction and external-ear resonance, the bandwidth and peak levels required to reproduce live (e.g., orchestral) performances, the nonlinear distortion characteristics of earphones and ears, the masking effect of background noise levels, and the effect of different types of hearing impairments. A simple set of preliminary guidelines results from the assumption that the hearing aid user has only a mild sensorineural hearing loss cochlear origin. Two laboratory hearing aids have been constructed in accordance with such guidelines; one is a practical Over-the-Ear aid with 8-kHz bandwidth, while the other is an In-the-Ear aid with 16-kHz bandwidth. The objective and subjective evaluation of these aids will be discussed briefly. If time permits, a portion of the prerecorded ABA listening test used to obtain fidelity ratings will be demonstated.
A consideration on a generalized loudspeaker system
Authors: Nomura Y.
Published: October 2005

A new design method, which uses the nonlinear optimization method, for a generalized loudspeaker system is presented. The frequency of the system is controlled by a system parameter vector in the 16-dimensional euclidean space, whose components are composed of the elements of the mechanical equivalent circuit. In order to realize the flat response and at the time the extension of the low frequency response, an evaluation function is defined. The value of the function decreases as the frequency response approaches to the target frequency response. By minimization (optimization) of the function, it is possible to obtain a vector which realizes the flat response and the extension of the low frequency response. This design method is rich in flexibility of design as compared with the former design method.
Signal processing technique for reducing tracing distortion in two-dimensional recording systems
Authors: Date Hikaru , Fukudome Kimitoshi
Published: October 2005

Because of high density recording and very short access time (less than 0.1 s) for 104−105 audio program sources of duration 10 s or so, a microfiche system which records audio signals as two-dimensional patterns after suitable signal processing is now being developed. Simulations are carried out on several candidates of signal processing methods in order to overcome serious distortion produced by mistracing of the two-dimensionally transformed audio patterns. Among them, recoding of pitch or noise and impulse response of vocal tract extracted by PARCOR method is wonderfully excellent and allows oblique mistracing over up to 16 lines of recorded two-dimensional signals without distortion for speech. Studies on general audio signals other than speech are continuing.
Binaural earphone for out-of-head localization
Authors: Sakamoto N. , Aoi T. , Kajitani M. , Atoji N.
Published: October 2005

This paper describes a binaural sound reproduction system which converts a monaural electric signal to a sound wave by an electroacoustic transducer and transmits the converted sound wave through acoustic transmission paths such as tubes to the left and the right ears of listener. This system has a reverberation unit comprising a pure mechanoacoustic element which produces two indirect sounds. These indirect sounds have a time delay relative to the direct sound and a phase difference of 180° from each other, and are mixed with the direct sound for each ear. These mixed sounds are transmitted through the acoustic transmission paths to the left and the right ears so that a sound image is localized externally of the head of the listener. As mentioned above, this paper presents the construction of the binaural earphone for out-of-head localization.
DUMAND: Deep Underwater Muon And Neutrino Detection, opening a “nu” window on the universe
Authors: Stenger Victor F. , Peteron Vincent Z.
Published: October 2005

A tutorial introduction to the neutrino and muon interactions, especially at ultra-high energies several orders of magnitude greater than those available from earth-bound accelerators, is followed by a description of the detection problems which have led to the proposal to instrument 10 km3 of ocean water near the sea bed for the study of a host of extremely intriguing physical and astrophysical events.