bluuebunny/crossref_metadata_2025 huggingface dataset:
Optimization of a low-frequency ultrasonic technique to monitor the change in physical states in viscoelastic media: Gelation process
Authors: Nassar G. , Nongaillard B. , Noël Y.
Published: July 2002

A new low-frequency ultrasonic device (50–100 kHz) in highly sharpened end sensors that behave as point sources were examined. The application of this new ultrasonic technique with two sensors coupled in the near field is to explore the relations between the physical properties measured through the evolution of the wave time of flight and structural changes during gel formation which is related to two factors: the ambient temperature and the mechanical resistance of the medium. The network evolution was interpreted by an approach based on the Flory model. The physical significance of this model was shown through a series of experiments using a low-frequency ultrasonic technique. Response curves demonstrate the different stages during gel formation.
Effective wide-band, low-frequency multiple resonator sound absorber
Authors: Shenoda Fathy B. , Abd-Elbasseer Mohammed
Published: July 2002

Resonator-type sound absorbers are incorporated to provide relatively large sound attenuation in narrow bands. They are applied in several environments and have great practical value for the attenuation of low frequencies. A detailed experimental study leads to the construction of a multiple resonator sound absorber. It is made of metal sheet and has a total volume of 0.0016 m3. The acoustic performance of the designed device, namely, the sound absorption coefficient and the normalized input acoustic impedance, were measured in 20-Hz steps at normal sound incidence in room temperature and under heating conditions up at 4000°C. The designed device provides good sound-absorption-frequency characteristics not only in the high values of the sound-absorption coefficient, but also in covering a wide frequency band (60 to 1000 Hz) without attenuation breaks. Also, its acoustic characteristics are changed very little under heating conditions. Applying earlier results obtained by Lord Rayleigh and simplified by Crandall for the propagation of a sound wave in a tube, and recently applied by Maa to characterize microperforated sound absorbers, the acoustic performance of the designed multiple resonator was predicted. The measured and predicted results are in good agreement.
Noise and vibration control for land mine detection
Authors: Costley R. Daniel , Sabatier James M. , Xiang Ning
Published: July 2002

Land mines can be detected by scanning the ground with a laser doppler vibrometer (LDV) to measure the motion (velocity) of the ground as it is being sonified. However, the system at times suffers from a low signal-to-noise ratio, especially when trying to detect deeply buried mines. If the signal-to-noise ratio could be improved, these mines could be more easily detected. One contribution to the noise is due to the acoustically induced vibration of the LDV. This noise is difficult to filter digitally or electronically since it occurs at the same frequency as the ground motion being measured. Passive noise and vibration control remedies have been incorporated with some success. These include enclosing the LDV within a box and mounting it on damped springs. However, it is impossible to make the LDV completely stationary at all frequencies. The translation of the LDV beam over anything other than a completely smooth surface will appear to the LDV as an out-of-plane motion of the surface. The noise and vibration control strategies will be presented and discussed, along with measurements showing both their effectiveness and limitations.
The effect of speech reading on masked detection thresholds for spoken sentences
Authors: Grant Ken W.
Published: July 2002

Detection thresholds for spoken sentences in steady-state noise are reduced by 1–3 dB when synchronized video images of movements of the lips and other surface features of the face are provided. In a previous report [K. W. Grant and P. F. Seitz, J. Acoust. Soc. Am. 103, 3018 (1998)], we showed that the amount of masked threshold reduction, or bimodal coherence masking protection (BCMP), depended on the degree of correlation between the rms amplitude envelope of the target sentence and the area of lip opening. In the present study, we extend these results by directly manipulating this cross-modality correlation through either bandpass filtering or amplitude adjustments of selected words contained in the target sentences. A control condition was also included in which visual orthography was provided to explicitly identify the target sentence prior to each test trial. Results showed that orthographic information reduced detection thresholds by about 0.5 dB for all target sentences. Preliminary results for filtered and amplitude-adjusted sentences suggest that the magnitude of the BCMP depends primarily on the cross-modality correlation between lip-area function and rms amplitude envelope computed over windows of approximately 300 ms. [Work supported by NIH and the Department of Clinical Investigation, Walter Reed Army Medical Center.]
Monaural phase effects: Timing versus level cues
Authors: Law Martin P. , Stellmack Mark A. , Viemeister Neal F.
Published: July 2002

As shown in previous studies, detection thresholds for a 500-Hz signal vary as a function of its phase relative to a 250-Hz masker. Due to differences in detectability, phase discrimination between suprathreshold 250- and 500-Hz components may be possible using the difference in effective stimulus level as a cue. Alternatively, phase discrimination may be based on changes in temporal fine-structure. If level were the cue, then the psychometric function [P(C) versus level of the 500-Hz component in the signal interval] should be V-shaped with a minimum at chance when the effective levels of the 500-Hz components in the signal and nonsignal intervals are equal. While the data did exhibit the predicted V-shaped function, performance dropped to chance only when the masker intensity level was below 70 dB, suggesting that fine-structure information can be used, but only above 70 dB. In another experiment, phase discrimination was assessed when the overall level of the two-tone complex was roved on each presentation. Small rove ranges significantly degraded performance, consistent with the use of a level cue. Overall, these data suggest that even at low frequencies monaural phase effects are not primarily based on fine-structure timing. [Work supported by NIDCD DC00683.]
Hearing discrete elements of a temporal sequence: Learning and generalization
Authors: Schlauch Robert S. , DiGiovanni Jeffrey J. , Ries Dennis T.
Published: July 2002

The ability to hear discrete elements of a temporal sequence was assessed. Subjects judged whether the first element of a three- or four-burst sequence of 50% duty cycle noise that alternated in level from low to high (or high to low) began with a low-level (60 dB SPL) or a high-level (80 dB SPL) burst. The rate was varied from trial-to-trial [selected values between 5 to 40 pulses per second (pps)] to prevent absolute duration and loudness cues. Initially, listeners were at chance performance for rates above roughly 20 pps. However, after 6 h of practice with feedback performance was near 90% for the highest tested rate (45 pps). After training with the stimulus that alternated in level, subjects were presented with a new condition; the three- or four-pulse sequences began with either a high-level or a low-level burst of noise, as before, but the level of each subsequent burst in the sequence was selected randomly between 60 and 80 dB SPL. Performance remained high for this condition, suggesting that listeners did not learn patterns of sequences but, rather, were able to hear discretely the first element in the sequence. [Work supported by NIH.]
Extended perceptual spaces for pitched and percussive timbres
Authors: Lakatos Stephen , Beauchamp James
Published: July 2002

A series of listening tests attempted to better isolate higher-order perceptual dimensions of timbre by building on past findings showing that spectral centroid and rise time represent principal acoustic correlates of the primary timbral dimensions. Listeners with varying levels of musical training rated the timbral similarity of three sets of pitched and percussive instrument sounds that were equated for centroid and rise time using signal processing techniques. Multidimensional scaling analyses yielded two- and three-dimensional perceptual spaces for the three stimulus sets. Several aspects of the spectral fine structure of the timbres correlated with the dimensions of these spaces, but the nature of the acoustic correlate varied somewhat depending on the stimulus set in which the timbre was presented. The results suggest that additional perceptual dimensions of timbre exist, but that their precise acoustic correlates are context dependent and therefore less perceptually ‘‘primary’’ than centroid and rise time. [This research was funded by Air Force Office of Scientific Research Grant No. F49620-99-1-0293.]
Vibration power transmission coefficients for the coupling of circular cylindrical shells to flat plates
Authors: Willis Benjamin F. , Burroughs Courtney B.
Published: July 2002

Bending and in-plane waves are coupled by curvature of structures and at junctions between structures. To estimate the effects of wave coupling on the transmission of vibrational power through junctions between curved and flat elastic structures, an analytical model of circular cylindrical shells coupled to flat plates along a circumference of the shell was developed. Donnell–Mushtari shell equations are used to model the infinite and semi-infinite cylindrical shells. Classical theory is used to model both bending waves and in-plane longitudinal and shear waves in the plates. Models include plates inside and outside the shell as well as continuous through the shell. All plates are perpendicular to the shell surface. Plates coupled to both semi-infinite shells, where the shell terminates at the junction with the plate, and to infinite shells are modeled. Power transmission coefficients are presented as a function of circumferential mode number and frequency for different types of incident waves either in the shell or in the plate, and for different shell/plate configurations
A new algorithm for modal identification using mode isolation
Authors: Drexel Michael V. , Ginsberg Jerry H.
Published: July 2002

Multiple degree of freedom (MDOF) algorithms are the dominant methods for identifying modal properties from measured response data. Although there are numerous MDOF methods, all are based on the notion that because the response of a linear dynamic system is the sum of many modal contributions, the extraction technique must deal with all of the modal parameters in a simultaneous fashion. Thus, these methods do not exploit the fact that each mode has unique characteristics that can be used to isolate that mode from the contribution of the other remaining modes. The mode isolation method proposes an algorithm that extracts the modal parameters of each mode in a recursive search, and then refines the estimation of each mode by isolating its effect from the other modal contributions. The mode isolation method has an advantage over MDOF methods in that MDOF algorithms require an a priori guess of the number of significant modes, or degrees of freedom, of the system. An error in that guess can have serious consequences with respect to the accuracy of the extraction process. Several MDOF methods attempt to remedy this situation by utilizing peak counting to find the number of modes contained in the system. In contrast, the mode isolation method offers a simple and consistent modal extraction methodology that is inherently automatic in nature. A numerical example of the mode isolation method is presented for a system that was previously studied with the eigensystem realization algorithm (ERA) and enhanced ERA. Eigenvalues and mode shapes are compared for each algorithm. The results suggest that the mode isolation method is more robust in the treatment of noisy data.
Improved power input methods in SEA modeling
Authors: Haberman Robert C.
Published: July 2002

The traditional method for determining SEA input power to structures subjected to point-force excitation is to consider the directly driven structural element as infinite. This allows for a relatively simple calculation of input power. For example, the power input to three-dimensional frame structures is to treat the driven beam as infinite and use the corresponding real part of the infinite beam mobility along with the mean-square force. Similarly, the power input to a structure consisting of interconnected plates is to likewise consider the driven plate as infinite and use the corresponding real part of the infinite plate mobility and force. Errors are introduced by this method when the drive point approaches the boundaries (in terms of flexural wavelengths) of interconnected structural elements. At these locations, the power is influenced by near-field effects and the transmission of power into the connected structure. To reduce these errors, a method based on modeling the driven member and directly connected structural elements is investigated. All the members are modeled as semi-infinite. Examples are given of interconnected beam and plate structures, and guidelines are introduced for improved SEA power input calculations
Probabilistic and possibilistic models of uncertainty in structural dynamics
Authors: Johnson Dexter F. , Langley Robin S.
Published: July 2002

The prediction of structural reliability under static loading has received much attention in the field of civil and structural engineering, and well-established methods such as FORM and SORM are available to estimate the failure probability. If the quality and quantity of the available data do not justify a probabilistic analysis, then the reliability can alternatively be assessed by using less precise ‘‘possibilistic’’ methods such as interval analysis or convex modeling. None of these probabilistic or possibilistic methods has as yet been applied extensively to noise and vibration problems, and the present work considers this issue. Initially it is shown that the probabilistic and possibilistic algorithms can all be expressed in the form of a constrained optimization problem, and the methods are then applied to a vibration problem involving a beam with an uncertain mass distribution. The beam is modeled both experimentally and theoretically to highlight various types of uncertainty that can arise: in addition to ‘‘controlled’’ uncertainties involving adjustable lumped mass positions, there are ‘‘uncontrolled’’ uncertainties arising from items such as boundary conditions and the support stiffnesses. The relative merits of the various reliability assessment techniques are discussed in light of this example.
Auditory–visual context effects in a speech and nonspeech condition
Authors: Norrix Linda W. , Oved Iris , Rosenblum Lawrence D.
Published: July 2002

Perception of an auditory continuum can be changed by a preceding visual context [Norrix and Green, in Proc. AVSP’99, Santa Cruz, CA, August, 1999]. This study asked whether this perceptual shift requires that listeners experience the stimuli as speech. A sine-wave continuum (/ara–ala/) was presented to three groups of trained participants for identification (auditory-only condition). Tokens from the continuum were also paired with a point-light display of a talker saying /aba/ (auditory–visual condition) and presented to the same listeners for identification. Observers in group one (speech) identified the sounds as containing an /r/ or /l/ and reported after the experiment that they heard the sounds as speech. Group two, also instructed to identify the speech sounds as containing /r/ or /l/, reported they did not hear the sounds as speech. Group three (nonspeech) was instructed to identify the environmental sounds as most similar to the ‘‘first’’ or ‘‘last’’ sound of the continuum. Results indicated a reliable shift for the auditory–visual compared to the auditory-only identification function only in the speech group, suggesting that auditory–visual context effects might depend on observers interpreting the stimuli as speech. [Work supported by NSF Grant ♯SBR9809013 awarded to Kerry P. Green (deceased).]
Development of a facility for simultaneous recordings of acoustic, optical (3-D motion and video), and physiological speech data
Authors: Bernstein Lynne E. , Auer Edward T. , Chaney Brian , Alwan Abeer , Keating Patricia A.
Published: July 2002

A multidisciplinary, multilaboratory project is underway whose focus is optical and acoustic phonetic signals and their relationships to each other in speech production and perception. The goals are to quantitatively characterize optical speech signals, examine how optical phonetic characteristics relate to acoustic and to physiologic speech production characteristics, study what affects the intelligibility of optical speech signals, and apply obtained knowledge to optical speech synthesis and automatic speech recognition. We describe the data acquisition facility, which includes recording of video, acoustic, electromagnetic midsaggital articulography, and 3-D motion capture data simultaneously and in synchrony. We also outline the design of the database of recordings being obtained and show examples from the database. [Work supported by NSF 9996088.]
On the visual distinctiveness of English words
Authors: MacEachern Margaret
Published: July 2002

Lipreading is facilitated when words are visually distinct. It has been established [Auer and Bernstein, J. Acoust. Soc. Am. 102, 3704–3710 (1997)] that words in the English lexicon manifest a low degree of visual similarity: most words remain visually unique, even when distinctions are collapsed across phoneme classes, as is characteristic of visually perceived speech. But, is the high degree of visual uniqueness a distinctive (nonaccidental) property of the English lexicon? Perhaps not: the lexical space of English is large, both in crosslinguistic terms and in absolute terms. In other words, many lexical slots exist; a fairly low percentage of them are filled by actual words [Hockett, 288–290 (1958)]. Therefore, the high degree of visual distinctiveness might be an ‘‘accidental’’ property of the lexicon, deriving from the segment inventory and phonotactics of the language. This study will explore this question by comparing the visual similarity properties of words in simulated (hypothetical) lexicons to those of words in the actual English lexicon. Each novel lexicon will be generated on the basis of the phonotactic patterns of the actual lexicon. The modeled lexicon is PhLex, a 30000-word phonologically transformable lexicon of American English [Seitz, Bernstein, and Auer (1995)].
The effects of source motion on the performance of matched field processors
Authors: Abawi Ahmad T. , Booth Newell O. , Schey Phil , Hodgkiss W. S.
Published: July 2002

To obtain a reliable and accurate estimate of the cross-spectral matrix, it is necessary to average individual matrices over a number of snapshots. However, an upper limit in the number of snapshots that can be used to average the cross-spectral matrix is imposed by source and interferer motion. Averaging while the target moves through resolution cells introduces signal gain degradation, spreading the beam-former peak response through the resolutions cells occupied by the source. Averaging while interferences move smears sidelobes, increasing the number of eigenvalues in the cross-spectral matrix and reducing the effectiveness of adaptive noise cancellation processes. During the May 1996 Shallow Water evaluation cell Experiment (SwellEx-96), which was performed in shallow water over a relatively flat bottom off the coast of San Diego, data was obtained with towed sources and interfering surface ships moving at many different rates. Data were recorded on several arrays including two 118-m 64-phone vertical array, one tilted by 45 degrees. This paper presents an analysis of source and interferer motion effects using SwellEx-96 data. Signal gain degradation, target and side-lobe smearing, and adaptive noise cancellation effectiveness are examined as a function of integration time, and number of elements used in the processing.
An objective method for determining soundboard material quality
Authors: French Mark , Cherng John G.
Published: July 2002

For a stringed instrument of the guitar or violin family, the dynamic characteristics of the soundboard are extremely important in determining the sound quality of the finished instrument. Since an exceptional instrument can be worth several tens of thousands of dollars, the importance of getting such a basic step correct are obvious. This paper describes our efforts to describe the desirable characteristics of a guitar soundboard blank using what is currently some of the highest quality test equipment available. Our goal is to provide objective standards that can be implemented in an industrial guitar-making process to help ensure the highest quality possible sound quality in the finished instruments. Formal modal tests on a range of soundboard blanks ranging from inexpensive ones intended for use by those just learning their craft to master grade blanks suitable for the highest quality instruments have been performed. The subjective ratings used to grade these blanks with objective test results are correlated.
Electric arc source for high-frequency seismic measurement
Authors: Martin James S. , Larson Gregg D. , Rogers Peter H. , Scott Waymond R.
Published: July 2002

An electric arc source was designed to study high-frequency seismic surface wave propagation. The noncontact nature of this source made it feasible for use in synthetic aperture transmit arrays. The transmit signal, which was not linearly controllable, was found to be predominantly in the 1- to 4-kHz band with a Gaussian spectrum. This is an octave below the simultaneously generated air acoustic pulse. The source was used to create a synthetic line array in conjunction with a stationary receiver. The experiment was conducted in a sand-filled tank. Surface wave speeds in the range of 80 m/s were measured on the resulting seismograms. Significant dispersion occurred in the propagating waveform at distances both near and far from the source. Surface wave arrivals were discernable over 1 m from the source and compressional head waves could be observed within 30 cm. The data were in good agreement with lower frequency measurements made by other techniques. The surface wave generation was studied and found to be a combination of the surface interaction of the arc and the air acoustic interaction. Strong hysteresis was observed in the first arcing event. Later, the signal was smaller but sufficiently stable for averaging. [Work supported by ARO.]
Quantifying a new form of acoustic creep in rocks and concrete
Authors: Smith Eric , TenCate James A. , Guyer Robert A.
Published: July 2002

A new form of nonlinear elastic response, which we call ‘‘slow dynamics,’’ has recently been found to scale universally in all sandstones, limestone, and concretes we have tested. Slow dynamics is unique among known creep processes; the elastic modulus and inverse loss tangent drop in response to harmonic acoustic stress, so the response violates the symmetry of the source. We show here that the slow dynamic recovery rate is a function of temperature, consistent with a thermal activation model for creep requiring a spectrum of energy barriers. We also show that short-term modulus drop, with slow dynamic recovery, accompanies temperature change of either sign, again violating the source symmetry. These two modes of activation are compared quantitatively, and suggest a common origin for slow dynamics in disequilibrated internal strain fields. [Work supported by the Department of Energy: Office of Basic Energy Sciences and IGPP, the Institute for Geophysics and Planetary Physics at Los Alamos.]
Syllable-boundary magnitudes from jaw movement patterns
Authors: Pardo Bryan , Fujimura Osamu
Published: July 2002

The C/D model [O. Fujimura, J. Acoust. Soc. Jpn. (E) 13, 39–48 (1992)] describes rhythmic patterns of utterances by amplitude-controlled pulse trains, each pulse representing a syllable or a boundary. The syllable magnitude (pulse height) controls mandibular movement. Assuming a direct proportional relation between the two variables, the syllable magnitude for each occurrence of five, nine, and Pine under various prominence conditions was inferred using Erickson’s Pine Street dialogue data [Erickson, Fujimura, and Pardo, Lang. Speech 41, 395–413 (1998)], which contained repeated corrections of the street address. A computational procedure has been devised to process semi-automatically microbeam data produced by two male and two female speakers. The timing of each syllable pulse was determined using the iceberg method [O. Fujimura, J. Acoust. Soc. Am. 99]. Syllable durations were determined in proportion to the syllable magnitudes. Boundary magnitudes were represented by the gaps among syllable durations. The syllable and boundary durations thus determined reasonably agreed with acoustic durations. [This work was supported in part by NSF Grant Nos. SBR-951199B, SBR-9809046, and BCS-9977018 (O. Fujimura) and by ATR/MIC, Japan.]
Measuring vocal quality with speech synthesis
Authors: Gerratt Bruce R. , Kreiman Jody
Published: July 2002

Previous research has demonstrated that listeners do not agree well when using traditional rating scales to measure pathological voice quality. Although these findings may indicate that listeners are inherently unable to agree in their perception of such complex auditory stimuli, an alternate explanation implicates the particular measurement method—rating scale judgments—as the culprit. A novel method of assessing quality—listener-mediated analysis by synthesis—was devised to assess this possibility. In this new approach, listeners explicitly compare synthetic and natural voice samples, and change speech synthesizer parameters to create acceptable auditory matches to voice stimuli. Thus this method replaces unstable internal standards for qualities like breathiness and roughness with externally presented, infinitely variable stimuli, thus overcoming one major hypothetical source of disagreement in rating scale judgments. To assess the reliability of this method, listeners were asked to adjust the signal-to-noise ratio for 13 pathological voices so that the resulting synthetic stimuli matched the original voices as well as possible. In a separate experiment, listeners judged the noisiness of the same stimuli using a traditional visual-analog rating scale. The relative reliability and validity of the two methods of quality assessment will be discussed, as will implications for models of voice quality perception.
Pathologies not included: Survey of premillennium speech recognition software
Authors: Tull Renetta
Published: July 2002

An October 1999 survey of several leading speech recognition software manufacturers revealed that the speech technology industry is unprepared to serve users with vocal pathologies in 2000. Companies surveyed had neither prepared for consumers with vocal pathologies nor had immediate plans to alter systems to accommodate the pathological voice by 2002. Examples of vocal pathologies include leukoplakia, carcinoma, hyperfunction, polyps, and nodules. While the aforementioned pathologies involve direct impairment to the vocal mechanism (e.g., swollen pharyngeal mucous membranes, vocal fold polyps), neurological disorders such as Parkinsons disease also cause vocal pathologies (e.g., dysarthrias, spasmodic dysphonia). Some companies are beginning to incorporate natural language processing for consumers with aphasia; however, many of the top producers of speech recognition engines have not considered the pathological voice within their disability market. Acknowledging that the speech technology market is still new, speech recognition software developers note that lack of time and manpower prohibits concentration on groups with special needs. This research examines the feasibility of altering linguistic grammars of templates from healthy, prototypical voices to accommodate the phonetic changes that are observed in the pathological voice. Backus-Naur form and Chomskys syntactic structures are used to modify existing grammar templates.
Improved voice activity detection of noisy speech
Authors: Koh Min-sung , Mortz Margaret
Published: July 2002

This paper reports on the performance of a new algorithm for detecting voice activity in the presence of additive Gaussian noise. The proposed algorithm enhances the noisy speech before determining voice activity. In the speech enhancement, two noise reduction stages are used. The first stage uses Donoho’s wavelet-based noise-reduction method. The second stage uses nonlinear spectral magnitude enhancement in the Fourier domain. After signal conversion back to the time domain and renormalization, simple energy thresholds are used to detect the presence of voice activity. Experimental results show that the new method can achieve good performance with preprocessing a speech signal even though a simple energy-based voice-activity detector is used. A new algorithm shows significantly better performance when it is compared with the voice-activity detector specified in the ITU standard G.729 Annex B and other methods proposed recently.
Speech processing using the average localized synchrony detection
Authors: Ali Ahmed M. A. , Van der Spiegel Jan , Mueller Paul
Published: July 2002

A new auditory-based speech-processing system based on the biologically rooted property of average localized synchrony detection (ALSD) is proposed. The system is a modification to the generalized synchrony detector (GSD) [S. Seneff, J. Phonetics 16, 55–76 (1988)]. It generates a pseudospectrogram of the speech signal by detecting periodicity, while reducing the response to the individual harmonics of the fundamental frequency and the sensitivity to implementation mismatches. This is achieved without sacrificing the frequency resolution. Hence, it presents a more consistent and robust representation of the formants. The system is evaluated for its formant extraction ability while reducing spurious peaks. It is compared with other auditory-based front-end processing systems in the tasks of vowel, stop and fricative classification on clean speech from the TIMIT database, and in the presence of noise. The results illustrate the advantage of the ALSD system in extracting the formants and reducing the spurious peaks, while preserving the frequency resolution. They also indicate the superiority of the synchrony measures over the mean rate in the presence of noise. [Work supported by Catalyst Foundation.]
Relative contributions of passband and filter skirts to intelligibility of bandpass speech: Some effects of amplitude and context
Authors: Bashford James A. , Warren Richard M. , Lenz Peter W.
Published: July 2002

Recently, Warren and Bashford [J. Acoust. Soc. Am. 106, L47–L52 (1999)] reported over 90% intelligibility for everyday sentences reduced to a 1/3-octave band (center frequency 1500 Hz, slopes 100 dB/octave, peak levels 75 dB). Surprisingly, when the bandpass sentences were partitioned using 2000-order FIR filtering, the rectangular 1/3-octave passbands had only 24% intelligibility, while the filter skirts separated by a 1/3-octave notch had an intelligibility of 83%, despite their severe spectral tilts. The present study found that when monosyllabic word lists were substituted for the sentences, the whole-band intelligibility was 26%, the passband 4%, and the filter skirts 16%. Intelligibility also was measured for 1/3-octave sentences with peak levels ranging from 85 down to 35 dB: The whole band intelligibilities ranged from 90% to 68%, and the filter skirt pairs each had three to four times the passbands’ intelligibility. These findings demonstrate the ability to compensate for extreme distortions of the spectral profile of speech. Hence, it is necessary to consider the contribution of frequencies outside the nominal passband as defined by the 3 dB downpoints along the filter slopes when dealing with filtered speech. [Work supported by NIH.]
Matched field processing with Doppler and the degrees of freedom in an acoustic channel
Authors: Baggeroer Arthur B.
Published: July 2002

Matched field processing which incorporates source motion, or Doppler, has been examined by several investigators. There are two motivations: (i) long integration times improve signal-to-noise ratios, but this leads to signal gain degradation if the source transits a single resolution cell, and (ii) mainlobes interfere constructively and sidelobes destructively. The MFP Doppler processing can be separated into incoherent and incoherent methods. In the former a series of short-term MFP outputs are incoherently combined while in the latter a time-varying replica is used. In both methods the number of degrees of freedom (DOF), i.e., the dimension of the representation basis set, in the acoustic signal is a critical parameter since it determines both MFP resolution and sidelobe properties. The DOF are limited by the minimum between number of sensors and the number of modes propagating in the channel. We examine the concept of degrees of freedom in an acoustic channel and the performance of Doppler-based MFP methods. [Work supported by ONR, Code 321US, Undersea Signal Processing.]
Shock formation in the presence of entropy gradients
Authors: Lin Hao , Szeri Andrew J.
Published: July 2002

The steepening of a normal compression wave into a shock in a homentropic flow field is well understood through the method of characteristics. In a non-homentropic flow field, however, shock formation from a compression wave is more complex. The effects of entropy (or sound speed) gradients on shock formation from a compression wave are determined using a wave front expansion in Cartesian and in spherical polar coordinates. The latter problem has application to the intense energy focusing of sonoluminescence, particularly when applied to a spherically collapsing gas. The principal result is an analytical criterion for the time and place of shock formation, for a wave propagating into a field of smoothly varying entropy.
Critical speech bands: Spectral resolution of contrasting temporal speech patterns
Authors: Healy Eric W. , Bacon Sid P.
Published: July 2002

The high intelligibility of narrow-band sentences is attributable to contrasting temporal patterns in different frequency regions within the narrow band [E. W. Healy, Dissertation, University of Wisconsin–Milwaukee (1998)]. In the current study, the spectral resolution for contrasting temporal speech patterns was measured by partitioning the narrow-band spectrum. Sentences from the HINT were filtered to a minimum overall bandwidth, centered at 1500 Hz, which allowed relatively high intelligibility of component words. The overall bandwidth was 2/3, 1, or 3/2 octaves. Each overall band was digitally filtered (several thousand dB/octave) into 1, 2, 4, 6, or 10 partitions. The envelope from each partition was used to amplitude-modulate a corresponding band of low-noise noise, and the modulated carriers were combined and presented to normal-hearing listeners. Intelligibility increased and reached an asymptote as the number of partitions increased. The number of partitions necessary for asymptotic performance was 4, 6, and 10 for the 2/3-, 1-, and 3/2-octave overall bandwidths. For each, this corresponds to a partition bandwidth of approximately 1/6 octave. These results indicate that, in the region surrounding 1500 Hz, resolution of individual temporal speech patterns approximates the critical band measured using traditional psychoacoustic methods and nonspeech stimuli. [Work supported by NIDCD.]
Effects of signal frequency and masker level on the Schroeder phase effect
Authors: Gifford Rene H. , Bacon Sid P.
Published: July 2002

Positive-Schroeder (m+) and negative-Schroeder (m−) harmonic complexes have identical amplitude spectra but different phase spectra. The corresponding waveforms are mirror images of one another. The m+ complexes usually produce less masking, revealing a so-called Schroeder phase effect (SPE). Many have hypothesized that the SPE results from a combination of the m+ complexes producing a more peaked response along the basilar membrane along with the influence of fast-acting compression. The present study examined the effects of signal frequency and masker level on the SPE in a simultaneous-masking paradigm. The signal frequency was 600, 1200, 2400, or 4800 Hz. The masker covered a spectral range from about one octave below to one octave above the signal frequency. Fundamental frequencies were chosen individually based on comparing data for m+ and sine-phase maskers. The overall level of the masker ranged from 50 to 95 dB SPL. The SPE generally increased with increasing signal frequency. For all signal frequencies, the SPE increased as the masker level increased from its lowest level. At the higher frequencies, however, the SPE decreased significantly with level. The results are consistent with a greater degree of compression at high frequencies and moderate levels. [Work supported by NIDCD.]
Temporal resolution of frequency-modulated and virtual frequency signals
Authors: Iyer Nandini , Feth Lawrence L. , Sridharan Ramkumar
Published: July 2002

Discrimination of ‘‘glide’’ versus ‘‘step’’ transition was investigated for two classes of signals: frequency modulated (FM) tones and virtual frequency signals [Lublinskaya (1996); Anantharaman et al. (1997)]. A ‘‘virtual frequency signal’’ refers to the frequency transition perceived in an amplitude-modulated two-tone complex when the amplitude of one of the tones increases with duration, while that of the other decreases. In the FM tone experiment, the glide signal traverses a certain frequency range in a linear trajectory, whereas the step signal traverses the same frequency range in a series of discrete steps. In the virtual frequency signal experiment, the glide signal has a linear amplitude transition whereas the step signal has a staircase-like amplitude transition. Listeners heard these signals in an adaptive 2Q, 2AFC paradigm. The experimental parameters for both classes of signals were center frequency (500, 1000, and 4000 Hz) and number of steps. The signal duration was retained constant at 250 ms. The smallest ERB separation for which the step signal could be distinguished from the glide signal was determined. Results from these experiments will be discussed along with implications for temporal resolution and spectral integration of virtual frequency signals. [Work supported by a grant from The Ohio State University College of Social and Behavioral Sciences.]
Temporal envelope processing in developmental dyslexics
Authors: Lorenzi Christian , Fullgrabe Christian , Dumont Annie
Published: July 2002

This study evaluates temporal-envelope processing in six dyslexic children (mean age: 10.10 years), six normal control children (11.6 years), and six normal control adults (24.8 years) by measuring: (1) temporal modulation transfer functions (TMTFs), i.e., the detection thresholds of sinusoidal amplitude-modulation applied to a white noise carrier, as a function of modulation frequency (4, 16, 64, 256, or 1024 Hz), (2) recognition performance for vowel-consonant-vowel stimuli over five sessions; speech stimuli were either unprocessed or processed to degrade the spectral information. Modulation thresholds were similar in normal children and adults; for both groups, TMTFs were low pass in shape. TMTFs measured in dyslexics were bandpass in four children, low pass and flat in the two others. Overall, modulation thresholds were higher in dyslexics than in normal children at 4 and 1024 Hz. Unprocessed-speech recognition performance was perfect in normal children and adults, and impaired in dyslexics. Processed-speech recognition performance was poorer in normal and dyslexic children compared to normal adults. Performance improved across sessions in normal children and adults, but remained constant in dyslexics. These results support the hypothesis that some dyslexics show abnormal auditory temporal-envelope processing. This may, in turn, explain their difficulties with speech perception. [Work supported by the Cognitique program.]
Production and perception of English vowels
Authors: Yang Byunggon
Published: July 2002

This study examined the relation between production of nine English vowels and perception of the synthesized vowels by 14 American male and female speakers. Fant’s bandwidth equations were employed to dynamically tune to the varied formant frequency values. A remarkable r2 value was obtained from the regressional analysis between the center formant frequency values at which the subjects perceived the same vowel quality from the discriminatory test and the formant values of the synthesis models. Males and females perceived the synthesized vowels in the same way with converging center formant values and similar ranges of the same vowel quality. There was a strong link between production and perception of male and female speakers. The average r2 value was very high, which suggests a very lawful relation between production and perception. From the individual analyses we found that listeners adjusted the criteria for vowel discrimination in relation to their own vocal tracts. This result suggests that speaker normalization can be pursued by finding an individual regression equation between the reference and observed formant patterns. Also, the present data proved that human listeners possess a very accurate perceptual mechanism to extract invariant features from complex acoustic stimuli.
Voice onset time of infant-directed speech in Korean
Authors: Lee Soyoung , Davis Barbara L.
Published: July 2002

Voice onset time (VOT) of infant-directed speech (IDS) was compared to adult-directed speech (ADS) at the sentence level in Korean. Previous research has not investigated VOT at the sentence level. Korean mothers were asked to describe pictures to their infants and to a Korean adult using subject-object-verb structure. Each picture showed an object noun and a verb containing stop consonants in prevocalic position. VOT of lenis, fortis, and aspirated stop consonants for nouns and verbs in IDS and ADS was measured. In general, Korean mothers used two different strategies in IDS compared to ADS: longer VOT for aspirated stop consonants and shorter VOT for lenis/fortis stop consonants. These strategies may make a clear distinction between aspirated and lenis/fortis conditions. However, VOT of aspirated verbs was shorter than for nouns in IDS. When mothers show exaggerated pitch accent at the end of sentences in IDS, noun and verb phrases are likely to be combined together. Verbs are no longer at the phrase initial position, so VOT of aspirated verbs becomes shorter. These results indicate that Korean mothers do not consistently provide an acoustically prominent phonetic environment for important syntactic structures in IDS to their infants.
Laboratory measurements of low-frequency acoustic scattering from a buried canonical target
Authors: Simpson Harry J. , Houston Brian H. , Frederickson Carl K.
Published: July 2002

Finding and identifying proud and buried objects in littoral waters is of growing interest. Current acoustic-based systems can readily detect objects sitting proud; however, identification is considerably more difficult. For a buried target, even detection may prove difficult especially using higher-frequency systems. These higher-frequency systems are limited in range and sound penetration, which makes imaging buried targets extremely difficult. The research presented here is concerned with the use of relatively low-frequency acoustics (1 to 20 kHz) to overcome transmission and penetration limitations. This research includes quantifying the acoustical interaction with a buried target for ensonification angles above and below the critical angle. The target used for these measurements was an aluminum cylinder that was 20 cm long and 30 cm in diameter. A right-angle cone was machined in one end of the cylinder to enhance backscattering. Free-field measurements and completely buried measurements have been conducted. Of particular interest is the scattering from the buried canonical target for ensonification angles below the critical angle. These measurements are used to quantify the evanescent field interactions and the roughened interface enhanced scattering from the buried canonical target. [Work supported by ONR.]
Influences of temperature and pressure on the material and acoustic scattering properties of zooplankton
Authors: Chu Dezhang , Wiebe Peter H. , Stanton Timothy K. , Hammar Terence R. , Doherty Kenneth W. , Zhang Jack , Redeer Benjamin D. , Benfield Mark C. , Copley Nancy J.
Published: July 2002

It is well known that small variations (a few percent) in sound speed and density contrasts of the fluidlike weakly scattering zooplankton result in a significant variation in the target strength of zooplankton and ultimately in acoustically derived estimates of zooplankton biomass. Since pressure and temperature often vary considerably as a function of depth, the material properties of zooplankton are expected to change accordingly. To accurately determine zooplankton biomass from the acoustic scattering data, knowledge of the relationship between their material properties and these oceanographic parameters is required. In situ measurements of sound-speed contrast of the zooplankton species, Calanus finmarchicus, were made with the Acoustic Properties Of zooPlankton (APOP) system in Wilkinson Basin (Gulf of Maine) in August 1999. The sound speed of this animal in the mixed layer above the thermocline (0–18 m) was significantly smaller than that right below the thermocline. Below the thermocline, temperature became nearly constant with depth so that only pressure was changing. The measured sound-speed contrast below the thermocline (60–180 m) increased slowly with increasing pressure. Models of the material properties of weakly scattering zooplankton that take into account the temperature and pressure dependence are proposed and numerical examples are presented. [Work supported by NSF, NOAA/NURP, and ONR.]
Reconstruction of flight and acoustic behavior of echolocating bats
Authors: Eastman Kyler M. , Simmons James A. , O’Farrell Michael J.
Published: July 2002

A method for localizing sounds was applied to simultaneously reconstruct the emission pattern and flight path of echolocating bats. An infrared camera and an array of three microphones were used to record simultaneous audio signals and video images of groups of bats while foraging. The bats studied (Eptesicus, Myotis, Pipistrellus) use frequency sweeping ‘‘chirps’’ in the 25–45 kHz to 100–120 kHz band for echolocation. Cross correlation of the bats’ chirps was used to find delays in chirp arrival time between microphones in the array. These delays were then used to locate the chirps’ direction. A time-series of these coordinates was found, then calibrated to pixel points of the corresponding infared video recording to reconstruct the flight path and chirp pattern for each of the bats over 15-s intervals. By providing simultaneous emission and position information throughout time, this method addresses questions about dynamic adaptation of the emission patterns of bats to the acoustic environment. Simultaneous adaptation between bats flying in groups is particularly interesting. An accompanying abstract discusses the implications of these findings for the current paradigm of information processing in the FM bat’s auditory pathway.