bluuebunny/crossref_metadata_2025 huggingface dataset:
The Kyma system: A multiprocessor DSP controlled by a visual language.
Authors: Hebel Kurt J. , Scaletti Carla
Published: October 2005

The Kyma system is a flexible, open-ended environment for creating sound. It has both a hardware and a software component. The hardware, called the Capybara, is a multiprocessor DSP based on up to nine Motorola 56001s operating in parallel. The software is a visual language called Kyma. Kyma provides a set of more than 80 signal generators, signal processors, compositional algorithms, and ways to access the analog-to-digital converters, sample files stored on the hard disk, and MIDI input and output. These modules can be combined graphically to create new modules and new functionality. In Kyma, sounds are specified graphically as time-varying functions of zero, one, or an arbitrary number of other sounds. Sounds can be specified at any level of detail from an individual sample to an entire composition. A functional representation is uniform, allows for shared subfunctions, and can be segmented to allow for scheduling across multiple DSPs. The visual interface to the multiprocessor DSP enables the user to interactively specify and test complex sounds and compositions.
Signal processing techniques in music synthesis based on a physical model.
Authors: Smith Julius O.
Published: October 2005

Advances in signal processing technology have enabled whole new categories of music synthesis techniques. One of the most promising is ‘‘physical modeling synthesis’’ in which sound is synthesized by algorithms derived from the physics of real musical instruments. The most successful models to date are those corresponding to linear systems with simple excitations, such as plucked or struck strings. However, progress has also been made in synthesizing bowed strings, woodwinds, and brasses, coupling a linear resonating element (the ‘‘string,’’ ‘‘bore,’’ or ‘‘horn’’) with a nonlinear excitation mechanism (the ‘‘bow,’’ ‘‘reed,’’ or ‘‘lips’’) derived from the physical equations. It is possible to simulate several simplified string or woodwind models on a single DSP chip, such as the Motorola DSP56001. This presentation will discuss signal-processing aspects of physical models for music synthesis and describe recent results.
Control of time in musical performances using a computer sequencer program and a synthesizer.
Authors: Mathews Max V.
Published: October 2005

Timing is one of the most important expressive factors in musical performance. The conductor program is a species of sequencer program with which a performer can control a synthesizer to play music. The conductor program responds, in real time, to gestures made by the performer on two radio batons. Three types of time control can be used: (1) Specific tempos can be written into the sequence stored in the computer memory. (2) A sequence of trigger points can be written into the sequence. The program acts to put the sequence of trigger points into correspondence, in time, with the sequence of beats generated by gestures of a baton. A trigger can cause a note to start or to stop. The time between two successive triggers can set a tempo. (3) Continuous control of tempo can be done by making tempo proportional to a coordinate (x, y, or z) of the position of a baton. All three tempo control methods can be used in the same piece of music by writing suitable instructions into the sequence. The conductor program thus provides very flexible and expressive time control.
Experimental study of ellipsoidal focusing of spark-produced N waves in air.
Authors: Wright Wayne M. , Blackstock David T.
Published: October 2005

An experimental investigation of ellipsoidal focusing of intense pressure pulses in air is reported. Previous related experiments by Müller and Grönig, Cornet, and Coleman etal., are reviewed. The goal was to provide information of use in understanding lithotripter fields. An N wave was produced by a spark at the near focus of a concave ellipsoidal surface, which concentrated the reflected signal at the remote focus. A wideband condenser microphone was used to measure the pressure field along the axis of the reflector and also perpendicular to the axis in the focal plane. The N wave increased in amplitude as it approached the focus and changed into a U-shaped wave in the focal region. Beyond the focus the two legs of the U steepened to form shocks, and a new, nearly normal N wave developed. Nonlinear effects clearly played a strong role. For reference, a brief comparison with predictions based on Hamilton’s small-signal solution [see following paper] is presented. [Work supported by ONR, ARL:UT IR&D program, and Texas Advanced Research Program.]
An axial solution for the reflection of a spherical wave from a concave ellipsoidal mirror.
Authors: Hamilton Mark F.
Published: October 2005

In certain lithotripters, a spherical acoustic wave is produced at the near focus of a concave, axisymmetric, ellipsoidal mirror. The mirror causes the intensity of the reflected field to be maximized in the vicinity of the far focus, where the kidney stone is positioned. In the present paper, the Kirchhoff integral is used to derive a closed-form linear solution for the impulse response of the reflected field along the axis of symmetry in a lossless, homogeneous fluid. Results for arbitrary incident waveforms are obtained by convolution. Because geometrical acoustics is used to obtain the boundary condition on the mirror, the solution is restricted to wavelengths that are short compared with the minimum radius of curvature of the mirror. Otherwise, the eccentricity and depth of the mirror are arbitrary. Secondary reflections are ignored in the analysis. A simple expression is derived for the pressure at the far focus. For the special case of a spherical mirror, the solution obtained by Cornet and Blackstock [J. Acoust. Soc. Am. 52, 115 (1972)] is recovered. [Work supported by the Office of Naval Research, the David and Lucile Packard Foundation, and the National Science Foundation.]
The role of gallstone and bile properties in stone breakup in lithotripsy.
Authors: Vakil Nimish , Everbach E. Carr , Gracewski S. M.
Published: October 2005

The fragmentation mechanisms of gallstones exposed to both invitro and invivo lithotripsy with a Diasonics Therasonic piezoelectric litho- tripter were studied. Mechanical properties of stones, including air content, microhardness, acoustic attenuation, and shear and longitudinal wave speeds were measured and correlated with objective measures of fragmentation rates. These measures included the number of shocks to the development of the first crack, and the mass of the stone fragments larger than 3 mm in diameter remaining after 2000 shocks. The relationship of the volume of the surrounding liquid to the efficacy of litho- tripsy was studied for gallstones that were removed from the same patient. Finally, gallstones removed surgically from three patients who underwent lithotripsy and had incomplete fragmentation were examined under a scanning electron microscope for signs of damage. The results of these studies point to the combined roles of cavitation and stone properties as the principal determinants of gallstone breakup in lithotripsy.
Scattering from cylindrical shells due to membrane waves.
Authors: Rumerman M. L. , Schreppler Stephen C.
Published: October 2005

Measurements of acoustic backscattering from an ensonified thin cylindrical steel shell in water were displayed in a three-dimensional manner as a function of both frequency and angle of incidence by using color to represent the pressure levels. The display clearly showed two sets of frequency–angle loci which represent high levels of backscattering. One set is confined to an angular interval of 17 deg on either side of the shell normal, and the other set is confined to an interval of 29 deg on either side of the normal. Theoretical analysis indicated that these loci are associated with the two types of membrane waves in the shell that are generated by the incident wave, and represent frequency–angle combinations which yield a circumferential membrane wave number whose real part is an integer, and correspondingly high membrane wave vibration levels. Excellent agreement was obtained between the measured loci and those calculated theoretically.
Application of the generalized time-frequency representation cone-shaped kernel to structural acoustics data.
Authors: Tran Q. T. , Moore P. M.
Published: October 2005

In the field of target structural acoustics, the backscattered signal is determined by the structural properties of the object and acoustic interaction phenomena. Traditionally, information about the target is obtained by observing the impulse response (time domain) or the frequency response (transfer function). For a detailed study of the scatterer’s structure, a joint time-frequency representation can reveal some inherent features that do not stand out in a single domain representation. The generalized time-frequency representations (GTFR) can provide a good estimate of the spectral components in the signal, and resolve the starting time of these spectral peaks in the time-frequency plane. The GTFR is characterized by a kernel, and the properties of a particular representation are determined by the constraints imposed on its kernel. Here, the cone-shaped kernel is compared to the spectrogram and Wigner distribution, and shown to provide and improved tool for understanding the backscattering phenomena. The method is illustrated with data from a cylindrical shell in water.
Backward-wave model of the high-frequency enhancement or quasiresonance in the backscattering from spherical shells.
Authors: Marston P. L. , Sun N. H. , Zhang L.
Published: October 2005

For certain elastic plates in a vacuum, the group velocity near the mode threshold of a Lamb wave becomes negative and the average power flow is directed opposite the phase velocity. This is known as a backward wave. It is proposed that backward leaky Lamb waves on thin fluid loaded shells contribute to the high-frequency enhancement that has been noted recently by several researchers. Though the ka of that enhancement lies close to that of the longitudinal or thickness resonance, ray models of the internal reverberations do not adequately describe the structure in the form function where the group velocity of the s1 wave can be negative [S. G. Kargl and P. L. Marston, J. Acoust. Soc. Am. 89, 2545–2558 (1991)]. In the backward-ray model a leaky wave with a phase velocity in the clockwise direction propagates energy in the counterclockwise direction. As a result, energy is leaked off toward the source without having circumnavigated the backside of the shell, having propagated an angle of only 2θl on the shell where sin θl=c/cl. The contribution to the backscattering has the approximate magnitude ‖f0,l‖ ≊‖ Gl‖exp (−2θlβl) and only occurs where the radiation damping βl is large. Some discrepancies in previous ray models for thick and thin shells are partially accounted for and a prompt surface ray mechanism for tone burst scattering is evident. [Work supported by ONR.]
Beyond intelligibility testing: Development of an information theory based voice communication effectiveness test.
Authors: McKinley Richard L.
Published: October 2005

Speech intelligibility has been a popular measure of voice communication effectiveness. However, intelligibility measures do not account for some critical parameters in voice communication effectiveness. Task complexity, task criticality, language complexity, and time, all affect voice communication effectiveness but are not factors in speech intelligibility measures. Information theory gives the conceptual and mathematical basis for quantifying these parameters in terms of entropy, mutual information, channel rate, and channel capacity. The Voice Communications Effectiveness Test (VCET) has been developed to directly measure channel capacity of a communication ‘‘system’’ by varying these critical information theory parameters. VCET consists of a two-way interactive communication task. Scoring is based on correct responses and time. The basis for the development of VCET, its structure, application, and preliminary performance data will be presented.
Measurement and predictions of hearing handicap using an additive noise model.
Authors: Nilsson Michael J. , Sullivan Jean , Soli Sigfrid D.
Published: October 2005

The ability to accurately, quickly, and reliably measure speech intelligibility in noise is an essential requirement for assessing the benefit of hearing aids to speech communication in noisy environments. Measurement error caused by variability of speech materials, content, speaking styles, talkers, and spectral characteristics of noise maskers often make accurate intelligibility measurements impractical. This paper will review measurements of speech intelligibility as applied to the hearing impaired. Emphasis will be placed on Dutch sentence tests developed by Plomp [e.g., R. Plomp, J. Acoust. Soc. Am. 63, 533–549 (1978)] and on recent English sentence tests we have developed, which control many of the sources of measurement error. Test development procedures will be described, and a summary of initial results will be used to demonstrate the reliability of the tests for both group and individual assessment of intelligibility in noise.
Comparison of three methods for the subjective measurement of speech quality.
Authors: Panzer Ira L. , Sharpley Alan D.
Published: October 2005

Three methods for the subjective measurement of speech quality, namely, the Diagnostic Acceptability Measure (DAM), the Absolute Category Rating method (ACR), and the Degradation Category Rating method (DCR), were used to evaluate a set of 24 systems conditions using a panel of 20 listeners. The DAM yields a Composite Acceptability Estimate (CAE), the ACR a Mean Opinion Score (MOS), and the DCR a Degradation Mean Opinion Score (DMOS). The 24 systems included three subsets: 12 modulated noise reference unit (MNRU) conditions, six speech to noise ratio (S/N) conditions, and six ‘‘n-bit quantization’’ conditions. All three methods provided scores that were monotonic over the range of values evaluated for each of the three subsets of system conditions. Analyses of variance, however, indicated differential resolving power for the three methods, across each subset of system conditions. Further analyses, including transformation of scores to a common scale, showed a linear relationship between CAE and MOS, but curvilinear relationships between CAE and DMOS and between MOS and DMOS.
Speech recognition in small room reverberation: Three experimental vignettes.
Authors: Johnson Carole E. , Kaylor Kelley R. , Robinson Tyson
Published: October 2005

This paper describes three experiments involving normal-hearing and hearing-impaired listeners’ speech recognition in small room reverberation. A consonant–vowel–consonant–vowel nonsense syllable test (NST) was recorded in a small room with a reverberation time of 0.8 s. Experiment I compared the monaural versus binaural performance of eight normal-hearing listeners in four listening conditions of (1) control (C), (2) noise-only (N) with a +10 dB S/N (speech-spectrum noise), (3) reverberation (R), and (4) reverberation and noise (N–T). Results indicated a binaural advantage and a ranking (highest to lowest) of scores for the listening conditions: C, N, T, and NT. Experiment II involved obtaining performance intensity functions at 25, 35, 45, and 55 dB SL (re: SRT) for consonant, vowel, and bisyllable scores for ten normal-hearing listeners. While consonant and bisyllable scores were significantly better at each successive SL, vowel scores were not significantly different across the SLs. Experiment III involved completing a Sequential Information Analysis, SINFA [M. Wang and R. C. Bilger, J. Acoust. Soc. Am. 54, 1244 (1973)] on five normal-hearing (NH) and five hearing-impaired (HI) (high-frequency) listeners’ consonant responses in four listening conditions similar to Experiment I, except the N condition had +12 dB S/N with multitalker babble. Results indicated that, by far, the least amount of information was transmitted in the N–T condition (NH−1.01 bits/HI 0.61 bits) with less than 50% transmitted for most features. High-frequency features (e.g., sibilancy) fared poorly in all conditions for HI listeners. Implications of the findings will be discussed. [Work supported by a Summer Academy Research Mentorship Grant, OUHSC, Higher Regents of the State of Oklahoma.]
Consonant confusions by human listeners for loud speech.
Authors: Martin Doug , Schiller Elaine
Published: October 2005

Loud speech stimuli were collected from three males and three females saying the 16 CV syllables of Miller and Nicely (1955). Headphone sidetone was mixed with 80 dBA pink noise. Talkers were instructed to speak loud enough to hear their voices over the masker. This resulted in average speech levels of 100 dBA at 1 ft. Only the loud speech was recorded. The intratalker speech levels were controlled within 2 dB. Seven subjects listened to the loud speech at a ‘‘comfortable’’ level, in quiet and mixed with multitalker babble. Subjects averaged 82% correct in quiet and 53% in babble. There were no significant performance differences for male and female talkers. Fricatives accounted for most errors with errors across all phonetic dimensions in both quiet and babble. In babble, subjects correctly identified voiceless stops 61% of the time but correctly classified stimuli as voiceless stops (place errors only) 92% of the time. Also in babble, nasals and voiced stops were confused 15% of the time; these errors never occurred in quiet.
Voice quality ratings: Theoretical considerations.
Authors: Kreiman Jody , Gerratt Bruce R.
Published: October 2005

Research on voice quality perception has proceeded essentially atheoretically over the years. A thorough literature review suggests that most studies implicitly assume a model that partitions listeners’ ratings into two components: the true magnitude of the quality being rated (e.g., roughness) and an undifferentiated error component. However, recent studies suggest that several sources, including listener biases and context effects, contribute consistently to the variance in voice quality ratings. A model is proposed in which listeners judge vocal qualities against internal standards which are built up out of their experience with different populations of speakers. These standards are inherently unstable, particularly for less-experienced clinicians, and may be systematically influenced by factors other than the quality being judged. Research and clinical protocols and statistical treatments that control for regular sources of variance in voice quality ratings are described. These protocols may improve the reliability and validity of perceptual evaluations, both of which are questionable given current methods of assessment.
Voice quality ratings: Empirical evidence for a new model.
Authors: Gerratt Bruce R. , Kreiman Jody
Published: October 2005

This model of voice quality ratings suggests that context effects may cause drift in listeners’ ratings by influencing the internal standard against which judgments are made. To test this hypothesis, we asked 12 clinicians to judge the roughness of 22 synthetic stimuli using two scales: a traditional five-point equal-appearing interval (EAI) scale and a scale with explicit anchor stimuli (also synthetic) for each scale point. The stimulus set included a relatively large number of near-normal voices; we predicted that this would produce an increase in the perceived roughness of moderately rough stimuli over time for the EAI ratings (by influencing an unstable internal standard), but not for the explicitly anchored protocol. As predicted, ratings on the unanchored EAI scale, but not the anchored scale, drifted significantly within a listening session in the direction expected. Ratings using the anchored protocol were also significantly more reliable than those gathered using the unanchored paradigm. These results are consistent with our model, and suggest that explicitly anchored protocols for voice quality evaluation might improve both research and clinical practice.
The Heard Island acoustic global warming feasibility test.
Authors: Spindel Robert C.
Published: October 2005

In late January and early February, 1991, 57-Hz acoustic signals were transmitted underwater from Heard Island in the southern Indian Ocean to receiving sites worldwide. The experiment was designed to test the feasibility of using underwater acoustic signals to measure warming of the world’s oceans. The idea is based on the fact that the speed of sound in water increases with increasing temperature, about 1 m/s for a 4 °C change, so that the travel time of acoustic signals transmitted between fixed points will decrease if the ocean is warming. While the predicted warming at mid-depth (1000 m) where most sound energy is confined is low, about 4 m °C/yr, resulting in a sound-speed change of only 0.02 m/s/yr, the resulting travel time change over global paths is large and easily measured, 0.16 s for 18 000-km range. The goals of the feasibility experiment were threefold. First, to determine whether adequate received intensities can be obtained with electronic rather than explosive sources, second, to determine the minimum signal strength and bandwidth requirements for sufficiently accurate travel time measurements, and third, to learn if a long-term program can be conducted without harming marine life. It was estimated that it will take approximately a decade of monitoring to reveal an underlying warming trend in the presence of oceanic temperature changes that arise from sources other than global warming. This presentation is an overview of the experiment. The following presentations provide specific results.
Ocean signature of greenhouse warming.
Authors: Munk Walter
Published: October 2005

The Princeton GFDL coupled ocean–atmosphere global climate model (GCM) of greenhouse warming yields estimates of ocean temperature change from 1990–2000 as a function of position and depth. These estimates have served as a basis for computing changes in acoustic travel time between Heard Island and various receiver sites. (I am greatly indebted to S. Manabe for computing these changes.) The results differ from ocean basin to ocean basin and from gyre to gyre. A typical result is a reduction in travel time by 2 s per decade over a 10 000-km path. Mesoscale eddies produce a month-to-month variability of 0.5 s rms. Gyre and basin variability are more troublesome. The goal is to deploy a reasonable network of acoustic sources and receivers with gyre scale resolution and mesoscale suppression.
The Heard Island global program: Today Heard Island, tomorrow the world.
Authors: Briscoe M. G.
Published: October 2005

The objective of the Heard Island global program is to develop and implement a global acoustic network for the detection of climatic variability in the ocean, using the principle that sound travels faster in warmer water. The first phase was the Heard Island feasibility test, which showed in January 1991 that m-sequence coded acoustic signals transmitted 18 000 km could easily be heard and that their travel times could be measured to a few tens of milliseconds; this is sufficient to detect potential ocean warming against background oceanic fluctuations after about ten years of monitoring. The second phase of the global program (FY92–94) will prepare for the implementation of acoustic sources and receivers at sites around the world. The preparations consist of the development and testing of improved sources and receivers, of a network design that takes into account global models of greenhouse warming and ocean response, and of working with the marine mammal research and protection communities so that the acoustic programs can also provide much-needed information on the response of marine mammals to low-frequency sounds. The third and fourth phases of the global program are the actual implementation of the acoustic network (FY94–97) and the operation of the network for at least a decade (FY95–05). The implementation will first take place in a sparse network configuration for broad ocean monitoring that is global, starting in the Atlantic, the Indian, then the Pacific Ocean, followed by implementation of higher-resolution regional (mapping) networks.
HIFE signals and general results.
Authors: Metzger Kurt , Birdsall Theodore G. , Dzieciuch Matthew A.
Published: October 2005

HIFE transmitted for 28 full hours and 8 half hours, repeating the same signal continuously for each transmission. HIFE had a repertoire of cw, Pentaline, and four different phase-coded time-resolving signals based on m sequences. The several purposes of each signal will be explained. Receptions were processed on line at 14 sites at ranges of 200 to 18 000 km; to date, the digital tapes from 8 sites have been processed at CSPL. The feasibility questions concerning power, stability, time resolution, and dispersion were answered, and a thorough study of the receptions begun. Examples of the on-line processing and selected frequency domain and travel-time-domain time series will be presented and discussed. An extraordinary look at the phase stability of the propagation was revealed through comparison of (1) the diagonal ship surge deviation from a uniform path measured at the transmitter, and (2) the integrated autocorrelation phase (IAP1) after constant Doppler correction measured at a 9000-km distant receiver.
Observations on the abundance and behavior of marine mammals exposed to the Heard Island source transmissions.
Authors: Bowles Ann E. , Smultea Mari , Würsig Bernd , DeMaster Douglas P. , Palka Debra
Published: October 2005

Marine mammal density and behavior were measured in a 70-×70-km square centered on the site of the Heard Island feasibility test 4 days before and throughout the transmissions. Observers were stationed on two large vessels; they conducted line-transect surveys, and monitored marine mammal behavior during transmissions. Observers surveyed 590 nm of tracklines before and 598 nm during transmissions; they made 123 h of observations before and 149 h during. Transmission source characteristics are described elsewhere. Forty schools of cetaceans and 18 pinnipeds were sighted before the transmissions; 40 schools and 25 pinnipeds were sighted after. Sightings of endangered cetaceans were equally common before and during transmissions (3 schools each), but the number of hourglass dolphin (Lagenorhynchuscruciger) schools increased and schools of midsized whales, chiefly southern bottlenose whales (Hyperoodonplanifrons) and minke whales (Balaenaacutorostrata), decreased. These changes could be explained by the transmissions, the survey vessel, or unknown natural factors. Cetacean school size may have changed during transmissions, but there were no consistent changes in direction of travel or calling behavior. Endangered whales showed changes in respiration rate and rate of reorientation. However, these whales were able to aggregate and probably feed during transmissions.
Sound production and reception by insects: Variety is the spice of life.
Authors: Forrest T. G.
Published: October 2005

Insects have a vast array of mechanisms for producing acoustic signals and receiving sound stimuli. Because of their small size, insects have special physical problems in producing efficient signals and localizing sound sources (signaling insects). Communicative sounds may be airborne, substrate borne vibration, or surface waves and are produced by movement of air, percussion, or stridulation. Reception may involve detection of particle displacement or pressure. These different modes of sound production and reception are discussed. Important mechanisms that have evolved in response to the insects’ physical problems of sound production and reception are shown.
Acoustic method for identification of geographical types of honeybees.
Authors: Kerr Howard T. , Collins Anita M.
Published: October 2005

Acoustic signals generated during flight by different geographical types of honeybees (Apismellifera) were collected and analyzed to determine the feasibility of an acoustic based method for type identification. Flight sounds were collected in the United States for European origin honeybees common to North America and at locations in Venezuela and Mexico for both the European types and the African origin types which have spread throughout South and Central America during the last three decades. Equipment and data collection procedures were developed so that individual honeybees providing the acoustic signals could be collected for subsequent identification by the reference morphometric method used by the USDA. The acoustic signals were analyzed and significant characteristics were identified. These characteristics combined with the morphometric identifications provided an identification algorithm: European origin honeybees have wingbeat frequencies generally less than 250 Hz; African origin honeybees have frequencies more than 250 Hz. This work was motivated by the need for a simple, nonintrusive, and inexpensive field survey method for identifying honeybees suspected of being Africanized. An electronic device called ‘‘Buzz Buster’’ is now being marketed.
A neural network architecture for recognizing dolphin acoustic response time and response type.
Authors: Root William , Ridgway S. H.
Published: October 2005

Bottlenose dolphins (genus Tursiops) were trained in ‘‘choice reaction time’’ experiments to respond to each of two different types of acoustic stimuli by producing the corresponding one of two different vocalizations. An acoustic time history for each stimulus–response pair was obtained by digitizing the voltage output of a nearby hydrophone. To determine response time required that the onset of the response vocalization within the digitized waveform be identified. To distinguish ‘‘correct’’ from ‘‘incorrect’’ responses required that each response waveform be categorized as ‘‘whistle’’ or ‘‘pulsatile.’’ Both determinations were complicated by the presence of background noise and by variability in the characteristics of response vocalizations among the several dolphins. A four-layer feed-forward neural network (using neurons with identity activation functions and unit-gain sigmoidal transfer functions) was developed which achieved a mean error of <1.9 ms in estimating response time and a response-type recognition rate of ≳98% on a test set of one thousand 100-ms vocalization samples from three dolphins, when trained by backpropagation using a training set of two hundred 100-ms vocalization samples from the same dolphins. The performance of this neural network architecture (which can operate at real-time rates) surpasses that previously achieved using offline discriminant analysis (BMDP/PC-90) applied to vocalization samples of much greater (e.g., 700-ms) duration.
Comparison of sonar discrimination by an echolocating dolphin and a counterpropagation neural network.
Authors: Au Whitlow W. L.
Published: October 2005

The capability of an echolocating dolphin to discriminate −0.23- and +0.27-mm differences in the wall thickness of cylinders (3.81-cm o.d. and 12.7-cm length) was reported previously [W. W. L. Au and D. A. Pawloski, J. Acoust. Soc. Am. Suppl. 1 88, S4 (1990)]. Echoes from the same cylinders were digitized using a broadband simulated dolphin sonar signal with the transducer mounted on the dolphin’s pen. Features of the echoes were determined by passing them through a constant-Q filter bank, and calculating the output energy of each filter. The upper frequency limit of the filter bank was set at 150 kHz, corresponding to the upper frequency threshold of the dolphin. The lower frequency limit, depending on the Q value, varied between 65 and 71 kHz. Echo features of the standard and each comparison target were analyzed in pairs by a counterpropagation neural network having two output nodes. Twenty echoes per target were used in the training set and 30 additional echoes per target were used in the test set. For the noise-free condition, the network performed at a comparable level to the dolphin for Q values between 4 and 5. In the presence of noise, Q values between 7 and 8 were needed before the network could perform at a comparable level to the dolphin for echo energy-to-noise ratios of 10 and 15 dB.
Relationship between psychoacoustics and ‘‘say–stay’’ identification in hearing-impaired listeners.
Authors: Nelson Peggy B. , Norton Susan J.
Published: October 2005

The relationship between gap detection, glide discrimination, and identification of synthesized speech stimuli as more like ‘‘say’’ or more like ‘‘stay’’ was investigated in 18 listeners with varying degrees of hearing impairment. Speech stimuli were patterned after Best, Morrongiello, and Robson’s (1981) ‘‘say–stay’’ identification task, which involved the listeners’ use of silent gap (between signal-to-noise and vocalic portions of the signals) and first formant onset frequency, both indicating the presence or absence of stop consonants. A significant effect of hearing impairment was found on the gap detection and glide discrimination tasks, and on the identification functions of the ‘‘say–stay’’ stimuli. However, the speech identification functions were not correlated with the psychoacoustic results. In fact, the identification performance suggested listeners with mild to moderate hearing impairments were using weaker spectral cues than the normal-hearing listeners, which is contradictory to what is predicted by reduced spectral and temporal resolution abilities.
Low-frequency acoustic biasing and its effect on middle and inner ear pathology.
Authors: Hoehmann Dirk H.
Published: October 2005

It has been shown that a continuous low-frequency sound can modulate threshold detection in psychoacoustic measurements and that the low-frequency acoustic biasing technique enables one to differentiate between a hydropic and a normal cochlea. The purpose of the present study was to establish differences in CAP, CM, and SP modulation for the normal cochlea, the cochlea with endolymphatic hydrops, and sensorineural and conductive hearing loss in the guinea-pig model. Six guinea pigs were operated on according to the method of Kimura and Schuknecht by obliterating the endolymphatic duct and destroying the endolymphatic sac. In addition six animals were treated with gentamycin (75 mg/kg) for 4 weeks and in six guinea pigs a conductive hearing loss was created by disconnecting the ossicular chain. Each group has a control collective of equal size. The degree of hearing loss was monitored using ECoG recording on the round window. A low-frequency masker of 52 Hz (90–110 dB SPL) was used to modulate CAP, CM, and SP elicited by tone bursts (1–8 kHz, 80–100 dB SPL). Following the measurements all temporal bones were histologically processed. In a normal cochlea, the CAP amplitude was modulated according to the change of phase of the low-frequency masker. In addition a pronounced modulation was registered for the SP amplitude, whereas the CM responses were suppressed. The CAP amplitude modulation of the hydropic cochlea and an ear with a conductive hearing loss was less biased, whereas a cochlea with gentamycine intoxication was found in between. For the SP modulation there was a significant phase shift in the hydropic cochlea compared to other pathologists and the normal group. A valid criteria to differentiate between a hydropic cochlea and other pathologists was the range of CAP modulation. A CAP modulation of less than 50 μV throughout the measurements was pathognomonique for endolymphatic hydrops. Frequency specific changes and the effect of the biasing technique on cochlea micromechanics will be discussed in detail. [Work supported by DFG Grant No. HE-477/8-2.]
The effect of impulse noise on the chinchilla.
Authors: Ward W. Dixon , Zhou Zhiyi
Published: October 2005

Five chinchillas were given exposures of increasing severity (10, 32, or 100 pulses at peak levels of 131 to 155 dB SPL) to simulated gunfire (pulses from a high-intensity loudspeaker, one pulse each 4 s) separated by 10-day intervals, in an attempt to determine the ‘‘critical’’ exposure. Little evidence of such a critical dose could be seen; permanent threshold shifts developed gradually and very slowly. The results are consistent with the hypothesis that a given single exposure will produce only a fixed amount of damage; when that damage has already occurred, a subsequent exposure will produce additional damage only if the dose is greater. [Work supported by NINCDS Grant No. NS12125.]
Condition monitoring and diagnostics of systems and its relationship to some aspects of artificial intelligence.
Authors: Muster Douglas
Published: October 2005

Artificial intelligence and condition monitoring and diagnostics of systems are emerging, computer-based transdisciplines. The origins of the theoretical kernels of both transdisciplines can be traced to the last century, but the richly structured theories in the body of knowledge that supports the present work in the cognitive sciences and our recently achieved ability to make real-time decisions related to the operation and maintenance of modern, large-scale, complex mechanical systems are due, in large part, to the explosion of synergistic activity in systems thinking and computers which has occurred in the past three decades. In artificial intelligence one strives to make machines (usually, digital computers) do the sorts of things that the human mind does, including making decisions and problem solving, which is an essential element in condition monitoring and diagnostics as well. Thus there is a strong link between the two transdisciplines. A generic definition of artificialintelligence is given above; however, there is no universally accepted definition of the term conditionmonitoringanddiagnostics, either in its general form or in one of its specific forms, say, the condition monitoring and diagnostics of machines. Nevertheless, it is believed that most practitioners of the condition monitoring and diagnostics art would agree that the activities encompassed by the term can be characterized as a two-phase sequence of processes designed to monitor the condition of an operating, functioning system for the purpose of diagnosing faults in the system.

Monitoring is a passive, system-specific, information and data gathering process. Diagnostics is an interactive decision-making process in which the unique, system-specific information and data obtained in the monitoring phase are used by a human or computer-based decision maker to decide if faults exist in the system, where they are located, their criticality and what advisory and/or corrective action should be taken. In the diagnostic phase, computer-based decision making (read ‘‘artificial intelligence’’) can be used to support decisions made by a human or to make and implement decisions automatically. Both cases are discussed here, using the condition monitoring and diagnostics of a gas turbine as an illustrative example.
Ultrasonic monitoring of bearing wear in
situ.
Authors: Hazony Dov
Published: October 2005

Of concern are wear measurements of surfaces of bearing seals and valves. It will be seen that these may be implemented in terms of ultrasonic transducer arrays, constructed so as to be an integral part of the wear surface, monitoring insitu the amount and rate of the material removed. The associated data are assembled and manipulated in real time allowing for computer control and prediction. Technical aspects of the device will be discussed including journal bearing oil film measurements and the displacement and orbit of the rotating shaft. [Work supported by TecSonics, Inc. of Twinsburg, Ohio, Ohio Edison, and EPRI.]
Acoustic emissions as diagnostic tools for complex systems.
Authors: Unal Aynur , Bush Jeff J. , Wu Chwan-Hwa ‘‘John’’ , Vaidya U. K. , Crocker Malcolm
Published: October 2005

In mechanical systems, there is a need to develop improved techniques for quick verification of the ‘‘integrity’’ and ‘‘fitness’’ of key plant components such as nuclear reactor cooling pumps, steam boilers, etc. Acoustic emissions (AE) provide a quick overall picture of the system at a much lower cost than some other nondestructive evaluation methods such as ultrasound or radiography. Even though AE are faster and cheaper, they suffer from lack of quantitative diagnostic features. This paper is a demonstration of the fact that the artificial neural network can classify the nature of the failure modes in real time using the peak amplitudes as the discriminating feature.
On-line machine condition monitoring using artificial neural networks.
Authors: Wasserman Philip D.
Published: October 2005

Modern predictive maintenance strategies often require continuous monitoring of rotating machines condition. By detecting incipient failures, expensive unscheduled downtime can be eliminated, unnecessary periodic maintenance procedures avoided, and potentially dangerous machine failures prevented. Vibration signatures have long been recognized as an effective indicator of machine condition. By observing vibration wave forms in the time, frequency, and cepstral domains, human experts are able to assess accurately the condition of bearings, gears, and other machine components. The expense and limited availability of human experts make it highly desirable to automate the vibration interpretation process. This has proven difficult; human pattern recognition ability has exceeded the most sophisticated computer programs. This paper describes a study where artificial neural networks were trained to emulate the ability of a human expert to detect defective rolling element bearings from their vibration signatures. A cascade of probabilistic and backpropagation neural networks learned to agree with the human expert on 100% of the bearing vibration samples, where these were drawn from data outside of the training set. [Work supported by the Electrical Power Research Institute.]
Present status of the diagnostic acceptability measure (DAM).
Authors: Voiers William D.
Published: October 2005

The DAM has been used to evaluate the acceptability of processed and otherwise degraded speech for more than a decade. It has undergone several modifications, but the principles on which it is based remain the same. In the latest version of the DAM, listeners evaluate experimentally processed speech in terms of the detectability of its effects on overall acceptability and on a diversity of elementary and complex perceptual qualities of the signal and background. Data thus obtained yield ‘‘isometric,’’ ‘‘metametric,’’ and ‘‘parametric’’ estimates of acceptability, which are compared in various ways to provide several types of diagnostic information and combined to provide a single, composite acceptability estimate (CAE). The DAM has recently undergone extensive validation and refinement, leading to, among other things, the identification of additional elementary perceptual qualities (EPQs) of the signal and background, which qualities determine overall acceptability and provide the basis of one means of its estimation. An evaluation of 120 selected types of experimentally degraded and system-processed speech demonstrated the qualitative and quantitative sensitivity of the DAM to all types of degradation involved. [Work supported by DOD.]
An independent replication of the diagnostic acceptability measure (DAM).
Authors: Tardelli J. D. , Kreamer E. W.
Published: October 2005

For the first time, an independent replication of the DAM [W. D. Voiers, IEEE Proc. ICASSP 204–207 (1977)], a Dynastat proprietary test, was conducted as part of a three-phase effort to facilitate secondary sources for the DAM. Dynastat conducted phases one and two of this effort. In the first phase, a description of the DAM and the procedures to conduct the test were developed. In the second phase, an extensive set of speech distortions was evaluated with the DAM at Dynastat. For this study, the results of the initial phase were used to set up a test facility and to recruit, screen, train, and calibrate a crew of listeners. This independent facility and listener crew then used the DAM for evaluation of the same set of speech distortions tested at Dynastat during phase two of the overall effort. The distortion set included various types of masking noise, filtering, amplitude distortions, interrupts, and signal compression algorithms. Most distortion types were tested at six levels designed to cover the measurement range of the DAM. There were a total of 180 conditions tested for six speakers each. Initial comparative analysis between the DAM results [composite acceptability measure (CAE) averaged over speakers and listeners] for Dynastat and this study demonstrated a strong linear relationship (r=0.97) and a standard error of the estimate equal to 3.3 DAM points. Outliers to this regression analysis were grouped by some distortion types. In order to study intralistener variability of the two crews, the isometric and parametric components of the composite measure, and distortion type variability, further analyses were conducted. A modified significant difference method was used on the overall and component measures for the individual conditions. These analyses led to recommendations for additions to the phase one DAM description and to the DAM listener calibration process. [Work supported by USAF Rome Laboratory RL/EEV.]
Thermo-viscous effects on finite amplitude sound propagation in a rectangular waveguide
Authors: Anderson M. J. , Vaidya P. G.
Published: October 2005

The role that thermo-viscous effects play in the propagation of finite level sound in a waveguide has been reexamined from a fundamental perspective. In the past, nonlinear acoustic interactions have been described by energy conserving modulation of spectral amplitudes as wave packets travel axially down the waveguide. To account for thermo-viscous effects in this modulation, investigators have included without formal justification into the modulation equations dissipative terms with a magnitude corresponding to the Kirchhoff rate of attenuation encountered in linear theory. In this investigation, the problem of the propagation of finite magnitude plane waves is analyzed in a different manner. As opposed to previous investigations, all three modes (acoustic, vorticity, and entropy) are considered from the outset. The boundary conditions are extended to include vanishing normal and tangential fluid velocity, as well as vanishing fluid temperature perturbations. A new solution at second order is presented (second order being the first correction due to nonlinearity), which is uniform in the spatial variables. As a consequence, it is shown that the thermo-viscous effects are incorporated into the spectral amplitude modulation equations through one of the boundary conditions. These modulation equations apply to both plane and higher-order modes, including the region arbitrarily near the cutoff frequency for the higher-order modes. It is shown that the small parameter 1/(N)1/2, where N=ρ0Dc/μ (the acoustic Reynolds number), is a special scale for analysis of nonlinear interactions in a waveguide. In particular, the relative magnitude of the sound source and 1/(N)1/2 is a determining factor that predicts whether nonlinear interactions will be significant.
Wave propagation in anisotropic liquid-saturated porous solids
Authors: Sharma M. D. , Gogna M. L.
Published: October 2005

Biot’s theory is employed to study the propagation of plane-harmonic seismic waves in a transversely isotropic liquid-saturated porous solid. Along with SH wave, the existence of three quasiwaves is discussed and analytical expressions for their velocities of propagation have been obtained. It has been observed that velocities of existing waves vary with the direction of propagation. Frequency equation for the propagation of Rayleigh-type surface waves at the free surface of transversely isotropic liquid-saturated porous solids has been obtained. Possibilities of existence of body waves and surface waves have been discussed numerically by assuming different sets of values of elastic constants. Dependence of the velocities of propagation on the direction of propagation has been exhibited.