bluuebunny/crossref_metadata_2025 huggingface dataset:
Ice canopy roughness inversion and Arctic propagation loss estimates via elastic perturbation theory
Authors: LePage Kevin , Schmidt Henrik
Published: October 2005

Low-frequency (f≪100 Hz) boundary interaction loss in the central Arctic has been observed to exceed the losses that may be predicted by free surface Kirchhoff or MSP scattering theories, 2-D boss scattering models, or large elastic feature scattering arguments. In addition, the measured frequency dependence of the scattering loss in the Arctic (∝f1.5) is lower than that which is predicted with either the boss or the large elastic feature scattering models. Here a self-consistent elastic perturbation theory is used to estimate coherent boundary interaction loss for the Arctic scenario. Using this theory it is found that the elasticity of the ice canopy can account for a substantial decrease in the local coherent reflection coefficient in comparison with a rough free surface interaction, while the frequency dependence of the loss remains proportional to f2. Using acoustics measurements obtained by a horizontal array under the Arctic ice, a ML inversion for the ice canopy roughness statistics is obtained using the perturbation theory, and these parameters are used to estimate the loss per interaction. This loss, when combined with a ray cycle distance taken from the relevant literature, yields an estimate of scattering propagation loss (dB/km), which closely matches the data in the low-frequency regime. When the canonical roughness parameters and cycle distances from the literature are used in the theory, the estimated loss is lower, but improvements are still obtained over the free surface results. [Work supported by ONR.]
Spatial and source level distributions of ice cracking in the Arctic Ocean
Authors: Greening Michael V. , Zakarauskas Pierre
Published: October 2005

This paper reports measurements of the spatial and source intensity level distributions of ice cracking events occurring in the rough Arctic pack ice. These distributions are computed from a total of 916 detected events covering approximately 2 h of data distributed over several days during April 1988. Measurements were obtained over the frequency band 4–200 Hz on a 22-element vertical array and a 7-element horizontal array deployed below the Arctic pack ice in 420 m of water. The observed spatial distribution of detected events is consistent with the case where ice cracking events are uniformly distributed. Source levels from 110 to 180 dB//μPa at 1 m were measured. The calculated source level distribution for all events approximates a linearly decreasing function on a log-dB scale of the number of events versus source level. The median, mean, and maximum numbers of events occurring per square kilometer per minute over a 1-dB band of source levels centered at 110 dB//μPa at 1 m are approximately 4, 8, and 40, respectively.
Auditory supplements to speechreading: Combining amplitude envelope cues from different spectral regions of speech
Authors: Grant Ken W. , Braida Louis D. , Renn Rebecca J.
Published: October 2005

Many listeners with severe-to-profound hearing losses perceive only a narrow band of low-frequency sounds and must rely on speechreading to supplement the impoverished auditory signal in speech recognition. Previous research with normal-hearing subjects [Grant etal., J. Exp. Psychol. 43A, 621–645 (1991)] demonstrated that speechreading was significantly improved when supplemented by amplitude-envelope cues that were extracted from different spectral regions of speech and presented as amplitude modulations of carriers with frequencies at or below the speech band from which the envelope was derived. This experiment assessed the benefit to speechreading derived from pairs of such envelope cues presented simultaneously. In general, greater improvements in speechreading scores were observed for pairs than for single envelopes when the carrier signals were chosen appropriately. However, lowpass-filtered speech provided at least as much benefit to speechreading as any combination of envelope signals tested that had the same overall bandwidth. Suggestions for improving the efficacy of frequency-lowered envelope cues for hearing-impaired listeners are discussed. [Work supported by NIH Grants DC 00010 and DC 00117 to Massachusetts Institute of Technology, and by NIH Grant DC 00792 to Walter Reed Army Medical Center.] a)Present address: Walter Reed Med. Ctr., Army Audiol. and Speech Ctr., Washington, DC 20307-5001.
Syllabic perception: Effects of absolute and relative temporal cues
Authors: Harsin Charles A. , Porter Robert J.
Published: October 2005

To investigate how syllabic differences might be conveyed by the manifestations of articulator gestures in the absolute and relative timing of acoustic cues, speech signals corresponding to the V+C+V sequence /ipi/ were created. The durations of various acoustic cues were independently manipulated, and subjects were asked to judge whether particular signals had the structure /i♯pi/ or /ip♯i/. Subjects were also asked to discriminate among the same signals. A clear, ‘‘categorical’’ relationship was found between bursts’ relative temporal positions in the intervocalic interval and syllabic categorization. On the other hand, discrimination measures revealed sensitivity to small absolute differences in segment durations regardless of judged syllable type. Thus, a comparison of syllable judgment and discrimination results suggests that both absolute and relative temporal cues are available for perceptual decisions but that relative cues are selected for syllable judgments, perhaps because they map more directly onto the relative timing of articulatory gestures. [This research was supported in part by University of New Orleans Chancellor’s Fellowship.]
Visual influence on heard speech syllables with a reduced visual image
Authors: Saldaña Helena M. , Rosenblum Lawrence D. , Osinga Theresa
Published: October 2005

Visual information of a speaker’s articulations can influence heard speech syllables [H. McGurk and J. McDonald, Nature 264, 746–748 (1976)]. The strength of this so-called McGurk effect was tested using a highly reduced visual image. A point-light technique was adopted whereby an actor’s face was darkened and reflective dots were arranged on various parts of the actor’s lips, teeth, tongue, and jaw. The actor was videotaped producing syllables in the dark. These reduced visual stimuli were dubbed onto discrepant auditory syllables in order to test their visual influence. Although subjects could not identify a frozen frame of these stimuli as a face, dynamic presentations resulted in a significant visual influence on syllable identifications. These results suggest that ‘‘pictorial’’ facial features are not necessary for audiovisual integration in speech perception. The results will be discussed in terms of the ecological approach, the fuzzy logical model, and the motor theory of speech perception.
Comparison of human and monkey sensitivity to the English liquid /ra–la/ contrast
Authors: Sinnott Joan M. , Barnett David M.
Published: October 2005

This study compared human and monkey sensitivity to the English liquid /ra–la/ contrast using a low-uncertainty repeating-standard discrimination procedure. A synthetic continuum was constructed by varying both a spectral cue in the third formant (F3 onset frequency) and a temporal cue in the first formant (steady-state period before the F1 transition into the vowel). In experiment 1, differential sensitivity was measured using both the /ra/ and /la/ ends of the continuum as standards. Humans were less sensitive than monkeys to variation at the /ra/ end, but equally sensitive at the /la/ end. In experiment 2, subjects were presented with pairwise comparisons from the continuum to discriminate. Human discrimination was best at the identified phoneme boundary near the center of the continuum, while monkey discrimination was best inside the /ra/ category. These results indicate qualitative differences between human and monkey discrimination of this continuum: categorical-like for humans, and noncategorical for monkeys. [Work supported by NIDCD.]
Acoustical analysis of false starts in spontaneous speech
Authors: O’Shaughnessy Douglas
Published: October 2005

In spontaneous speech many false starts occur, where a speaker interrupts the flow of speech to restart the utterance. The acoustic aspects of such restarts in a common database were examined for duration and fundamental frequency. Automatically identifying the type of restart could improve speech recognition performance, by eliminating one version of any repeated words (or parts), and in the case of changed words, suppressing the unwanted words, so that the recognizer operates on only desired words. In virtually all current recognizers, words in a restart either simply pass to the textual component of the recognizer or cause difficulties in having a proper interpretation in the language-model component (since the language model is invariably trained only on fluent text). The spoken data consists of 42 speakers, each speaking about 30 different utterances (median length of about 12 words). There were 60 occasions with simple repeated words (or portions), 30 cases of inserted words, and 25 occurrences of new words substituted. When a word was simply repeated in a restart, it had virtually the same prosodics usually, but occasionally the repeated word had less stress. With a substitution or insertion in the restart, the modified word was virtually always more stressed. For restarts where the speaker stopped in the middle of a word and simply ‘‘backed up,’’ the pause lasted 100–400 ms in 85% of the examples. About 75% of the interrrupted words did not have a completion of the vowel in the intended word’s first syllable. In virtually all examples, the speaker completed at least 100 ms of the word, however, before pausing for at least 100 ms.

When the pause occurred at a word boundary, the words repeated after the pause were either a straight repetition with little prosodic change or a repetition where the repeated words shortened up to 50%. As for the repeated words (after the pause) prior to the inserted word, function words showed little or no shortening, but usually had lower F0; content words here exhibited significant shortening and lower F0. Such prosodic change only applied to nonprepausal words, because words immediately prior to a pause were often subject to significant prepausal lengthening. [Work supported by NSERC.]
High-frequency scattering in the Arctic
Authors: Sammelmann Gary Steven
Published: October 2005

There currently exists no high-frequency model of propagation in the Arctic. This article describes a simple, fast model for computing high-frequency propagation in the Arctic. This model is a one-parameter model of scattering in the Arctic, where the free parameter of this model is the average rms slope of the small-scale surface roughness of the ice canopy. This model couples ray theory to a scattering strength formalism of scattering from a rough surface. This coupling is achieved by associating an effective area of ensonification with each ray trajectory from the source to the receiver. The average intensity at the receiver is obtained by performing an incoherent summation over all possible ray trajectories from the source to the receiver, including nonspecularly scattered rays. A comparison between model predictions and experimental measurements of forward scattering is made. The model predicts the overall shape and amplitude of the scattered signal within 5 dB depending upon the range of the source to the receiver.
Comparison of the contributions of surface pressure and normal displacement in the Kirchhoff–Helmholtz integral representation of scattering by an elastic structure in water
Authors: Hickling Robert , Petrovic Mirjana , Ball James F.
Published: October 2005

The pressure in the scattered sound field of an elastic structure in water can be represented in the standard form of the Kirchhoff–Helmholtz integral with the free-space Green’s function [see, for example, A. D. Pierce, Acoustics: AnIntroductiontoitsPhysicalPrinciplesandApplications (ASA, New York, 1989), p. 182]. This integral, over the surface of the elastic scatterer, involves terms in both the normal component of displacement and the acoustical pressure at the surface. These quantities are of course not independent of each other, the surface pressure being derived from the surface displacement by means of an integral equation. It is of interest, however, to examine the relative contribution of the pressure terms and the displacement terms to the backscattered sound in the far field. This is computed for an aluminum solid sphere and spherical shell.
Modeling of echoes from elastic spherical and cylindrical shells in a convergence zone
Authors: Baus T. A. , Chang W.
Published: October 2005

The monostatic echoes from an arbitrary incident signal in a deep ocean environment are computed for aluminum, steel, and plastic spherical and cylindrical shells. The target frequency response is calculated using thin shell theory. The medium response is obtained using the frequency-based generic sonar model (GSM). The combination of the target and medium transfer functions gives the full frequency response for the target-medium system. The product of the total transfer function with the frequency spectrum of the incident signal yields the quantity whose inverse FFT is the time series or echo. This return signal incorporates the effects of both the medium and target on the incident signal. [Work supported by ONT Code 231.]
Evaluation and modeling of channel independence and uncertainty processing in cochlear implant subjects
Authors: Collins Leslie M. , Wakefield Gregory H.
Published: October 2005

Results from previous experiments [Collins and Wakefield, Midwinter Research Meeting, Association for Research in Otolaryngology (February 1992)] have shown that performance in temporal pattern discrimination tasks is an efficient and valued psychophysical predictor of speech recognition. In the present paper, these measures are refined by extending the discrimination task to include temporal patterns across two electrode ‘‘channels.’’ Data from implanted subjects are presented and related to both speech and discrimination performance in the single channel case. A model that accounts for the discrimination data is proposed and used to evaluate channel independence. This model utilizes an alternative optimal statistic to that suggested by Sorkin [Sorkin, J. Acoust. Soc. Am. 87, 1695–1701 (1990)] for determining the performance of an ideal receiver. While this alternative model provides a better fit to the single channel data from Sorkins’s experiment, both the alternative model and Sorkins’s model fail to account for the worst performance of the subjects. Additional modifications to the model that reflect the impaired ability of some subjects to handle signal uncertainty are described and evaluated. [Research supported by NIH NIDCD NS21440.]
Prediction of sonic boom overpressures near lateral cutoff
Authors: Downing J. Micah , Lee Robert A.
Published: October 2005

Two methods of sonic boom predictions are used to compare predicted overpressures to measured sonic booms to quantify the effects of weather and flight variability on the lateral propagation of the booms. The sonic boom measurements were collected by the US Air Force at Edwards AFB, CA using a variety of supersonic aircraft. The testing included 43 flights from eight different aircraft at several Mach number and altitude combinations. The flights were planned for steady and level operation but some maneuvering of the aircraft caused focusing of the propagated booms at several measurement locations. The comparison of measured to predicted peak overpressures uses both Carlson and Full ray-tracing algorithms. The comparison examines the lateral spread of the boom and the effects on local weather profiles and variability of the flight profile on the propagated boom. This analysis demonstrates that, in general, predictions are in good agreement with the measured values but tend to overestimate the peak overpressures near the lateral cutoff.
Sonogram descriptors of propagation/scattering events
Authors: McCoy John J. , Chen Xingfu
Published: October 2005

At the ASA meeting in Salt Lake City, the windowed Fourier transform (WFT) was presented as providing the bases for phase-space representations for acoustic signals. For pulsed signal propagation in a one-dimensional waveguide, the sonogram−a description of the signal in a combined time/frequency space−was shown to be a representation of the signal that can be ‘‘propagated.’’ That is, this description of the acoustic signal as input to a propagation/scattering event provides the necessary information for determining this description at the completion of the event. It was further shown that propagation/scattering events are described when using this representation, by operators that might be termed, ‘‘propagator-filters.’’ Thus, the framework is an alternative, to both an ‘‘impulse-response function’’ and a ‘‘transfer function,’’ as quantitative descriptors of propagation/scattering events. The issue of the convenience of this descriptions is to be addressed. Using a moderately thick rod theory due to Mindlin and McNiven, this issue is addressed in the context of two propagation/scattering events for moderately broadband signals. One is the propagation of these signals through dispersive waveguides. The second in the phenomenon referred to as an end resonance. In our numerical experiment, this phenomenon applies when a broadband forcing is applied to the end of the rod. It manifests itself in a strongly frequency-dependent input of significant acoustic energy to a vibration confined to the end of the rod, a consequence of higher-order propagation modes below cut-off, with the subsequent ‘‘leaking’’ of this energy to the lowest order mode, which has no cut-off. The efficacy of the phase-space description is clearly evident.
Further report on the propagation of spark-produced N waves through turbulence
Authors: Lipkens Bart , Blackstock David T.
Published: October 2005

A laboratory model experiment to study the effect of turbulence on waveform and rise time of sonic booms has previously been reported [B. Lipkens and D. T. Blackstock, J. Acoust. Soc. Am. 90, 2307(A) (1991)]. Electric sparks produce the N waves, which then propagate through turbulence generated by a plane jet. Downscaling of the model turbulence from atmospheric turbulence was designed to match downscaling of the model N wave (in terms of duration and rise time) from the sonic boom. Reported here are hot wire anemometry measurements of the jet, which confirm that the turbulence characteristics are indeed those of a classical plane jet. The measured values of the integral length scale, Taylor microscale, and Kolmogorov microscale show that the desired downscaling from atmospheric turbulence was achieved. Also reported are measurements that show how rise time and wave shape distortion vary with propagation distance through the jet, turbulence intensity, and peak amplitude of the N wave. [Work supported by NASA.]
NASA’s high-speed research program
Authors: Darden Christine M.
Published: October 2005

In response to a study by the Office of Science and Technology of the Office of the President that identified technology development to support a long-range supersonic transport as one of the national aeronautical goals, NASA awarded contracts for market and technology feasibility studies of a high-speed civil transport (HSCT) to Boeing Commercial Airplanes and Douglas Aircraft Company in October 1986. Areas to be emphasized in these feasibility studies included market, economics, range, Mach number, fuels, payload, and technology needs. As a result of these studies, NASA initiated a program in 1990 to identify and develop technical and economically feasible solutions to environmental concerns surrounding this type of airplane. A second phase of the program to enhance the economic competitiveness of the vehicle is scheduled to begin in 1994. Results of the early studies showed that the economic viability of a supersonic transport would be tremendously reduced by environmental impact associated with sonic boom which limits overland supersonic flight. Research to develop technologies that would allow supersonic overland flight have been an important part of Phase I of NASA’s high speed research (HSR) program. Results of the feasibility studies that support the current HSR research will be covered in a 10-min overview.
Overview of current knowledge and activity in sonic boom research
Authors: Maglieri Domenic J.
Published: October 2005

This presentation will introduce the session on sonic booms by providing a brief overview of current knowledge and activity relative to their generation, propagation, and effects. A large percentage of the sonic boom flight test database was acquired in the decade of the 1960s in connection with overland flight of the proposed U.S.-SST. Interestingly, on every one of the measurements that included over 20 different vehicles, the sonic boom signature was basically N wave or sawtooth in character. Sonic boom minimization schemes were addressed but were considered, at the time, to be very costly and questionable. Recent renewed interest in high-speed civil transports (HSCT) stimulated another concerted effort to address the overland sonic boom issue. Analytical and experimental studies have been directed toward vehicle designs that will produce minimized sonic boom signatures, which still allows for a viable airplane. A key parameter in shaping the boom signature for reduced response is to increase the rise time. Such modified waveforms have been shown to be ‘‘do-able’’ on wind tunnel experiments out to about 50 body lengths. It is generally believed that these modified (non-N-wave) signatures will persist for very large distances (300 body lengths) in a real atmosphere; however, this has yet to be demonstrated. The influence of temperature, pressure, wind gradients, and turbulence on sonic boom N waves propagating down through the atmospheres has already been established. Molecular relaxation of O2 and N2 and relative humidity has also been shown to influence boom signatures, especially rise time. The effect of these parameters on modified waveforms are being established. Finally, some unique features relative to sonic booms produced by vehicles flying at very high altitudes at hypersonic speeds are also presented. [Work supported by NASA and USAF.]
Time delay estimation of noisy signals using spatial amplitude distribution analysis
Authors: Rog Andre M. , Losh Marc W. , Riemer Terry E. , Trahan Russell E.
Published: October 2005

The estimation of time delay in signals in the presence of noise has been an historically stubborn problem. A discrete method, known as spatial amplitude distribution (SAD) analysis, is shown here to be an effective procedure for dealing with time delay estimation issues. This technique employs a window that is shifted along the data point set containing the signal (if any) and noise. The amplitudes of this windowed signal segment and the original reference signal are paired point for point, forming a two-dimensional distribution. As the window is shifted along the noisy signal a set of distributions is produced that may be analyzed independently of time. SAD map simulations involving both sinusoidal and Gaussian noise signals contaminated with Gaussian noise of varying SNR produced results indicating substantially improved performance over conventional cross-correlation techniques.
A comparison between ray theory and normal mode theory to very high frequencies
Authors: White E. , Werby M. F.
Published: October 2005

A new normal mode method capable of predictions to very high frequencies is developed. The objective is to compare predictions of ray theory with the more exact normal mode theory to determine how well and at what point ray theory agrees with normal mode theory. The approach is to compare transmission loss curves computed by both methods. Discussion of the discrepancies are made and comments will be made on how improvements can be implemented. In strong ducting problems it is shown that there is a strong association between the lower order modes and the ray trace results.
Does ray chaos in range-dependent environments disappear when higher-order approximations are made?
Authors: Mazur Martin A.
Published: October 2005

Recent work has shown that when range dependence, in either the index of refraction or the boundary conditions, is introduced into the infinite frequency (ray) approximation to the wave equation, ray chaos results. This is because the resulting conservative system of ordinary differential equations is nonlinear and nonintegrable. Attempts have been made to reconcile chaotic ray behavior with the fact that chaotic solutions of the full linear wave equation cannot exist. When the terms neglected in making the eikonal approximation are kept, the resulting system of equations is generally no longer conservative, and hence chaos is not a necessary result of the higher-order approximation. [Work supported by ONR.]
Travel time tomography simulations for the test ocean constructed for the 124th Meeting of the Acoustical Society of America
Authors: Cornuelle Bruce D.
Published: October 2005

Geometric rays were traced for a variety of source–receiver configurations in the test ocean sound-speed field constructed for the 124th Meeting of the Acoustical Society of America. A stochastic inverse incorporating the statistical information distributed with the test ocean was performed using only one-way travel times as data. Realistic errors in travel times and source–receiver positions were allowed for. The travel times were assumed to be contaminated with internal-wave-induced travel time fluctuations. The fluctuation variance increased linearly with range, with a magnitude of 10 ms rms at 1000 km. This internal wave noise can be reduced by a factor of N−1/2 by assuming that the mesoscale sound-speed structure remains static over N internal-wave-induced decorrelation times. Decorrelation times shorter than 10 min have been observed in the North Pacific at 1000-km range, so a daily average can yield a net contamination below 1 ms, neglecting the effects of longer period fluctuations like internal tides. The simulation results are sensitive to the size of the assumed error, and using values below 1 ms can lead to unrealistically good performance. The size of the source–receiver position errors can vary dramatically depending on the navigational techniques used and can have significant effects on the array design problem. These sensitivities will be explored in the process of choosing a source–receiver array adequate to resolve 96% of the sound-speed variance at 700-m depth in the test ocean.
Relative arrival tomographic reconstruction using drifting receivers and a ship-board source
Authors: Lynch J. , Pawlowicz R. , Duda T. , Newhall A.
Published: October 2005

Rather than testing ocean imaging with accurately located and timed acoustic transmission equipment, a study of the quality of images produced with sonobuoy-type drifting receivers and ship-lowered sources is presented. The imprecise location and timing information means that the only information available is the relative timing of multipath (ray-type) arrivals. Accuracy of relative arrival time measurement is presumed to be 5 ms and positioning is to 1 km. The forward model is a ray trace through the background, and the basis set has various vertical forms. Three array sizes are considered, 100, 350, and 1000 km. Deployments on these different scales will have different oceanographic goals. The distribution of successfully mapped variance at all scales smaller than the array dimension will be shown. The appropriate array dimension, relative to the sound-speed field correlation length, will be discussed. Comparisons with more standard quick-survey methods (e.g., AXBT drops) will be considered.
Temporal integration of noise band signals measured in a profile analysis task
Authors: Formby C. , Heinz M. G. , Luna C. E. , Elbert S. C.
Published: October 2005

Of interest in this study was whether temporal integration (TI) of brief noise band signals in a broadband masker reflects (1) a successive comparison between the intensities of the signal and nonsignal intervals of a trial or (2) a simultaneous comparison of the signal and background intensities within the signal interval (i.e., spectral shape discrimination). This issue was studied in a profile analysis paradigm in which the overall levels of the signal and background (masker) were randomized together from interval to interval of each 2IFC trial. Random level variation rendered successive comparison of intensities between intervals useless and forced the listener to discriminate changes in spectral shape. Accordingly, the hypothesis was that if TI depended on successive comparison of levels, then randomized level would disrupt performance; whereas if TI were independent of randomized level, then the listener must be discriminating spectral shape. Detection thresholds were measured for signal bandwidths from 125 to 4000 Hz, both with random level (35 to 50 dB) and fixed level, at six standard durations from 10 to 320 ms. The TI functions were independent of randomized level. These results are consistent with spectral shape judgments in detecting brief noise band signals in a broadband noise background.
Directional masking of a complex signal by speech spectrum noise
Authors: Abouchacra Kim , Letowski Tomasz
Published: October 2005

An experiment was conducted to investigate the effects of spatial separation of speakers on the detectability of a complex stimulus in speech spectrum noise (65 dB SPL). Twenty normal hearing subjects were randomly assigned into groups of 10 subjects each. Detection thresholds were obtained using the Békésy up-and-down tracking technique and an ascending technique for the first and second group, respectively. At five different combinations of azimuth of the signal and noise sources, six thresholds were obtained for each subject. The results showed that detectability of a signal in noise depends on the relative spatial location of both the signal and noise source. A secondary objective of the experiment was to determine if knowledge of the source locations improves a listener’s detection threshold. A light display was used to indicate the source locations. No difference in detection thresholds was seen for either group in any sound source combination when compared to conditions without a light display.
Tempo sensitivity in auditory sequences
Authors: Drake Carolyn , Botte Marie-Claire , Baruch Clarisse
Published: October 2005

Differential thresholds for 11 tempi (ranging from 100 to 1500 ms between successive onsets) were measured for four subjects using a 2AFC paradigm. In a first experiment, the number of events in the sequence was varied to test whether sensitivity is greater in regularsequences than in simple duration discrimination tasks (only two events). Relative jnd were: (1) optimal at intermediate tempi (as low as 1.5% in the range between 300–800 ms), and (2) decreased as the number of events increased (2 events=6%, 3 events=4%, 5 events=3.2%, 7 events=3%). A second experiment tested whether this higher sensitivity was due to the fact that the sequences were regular or not by measuring differential tempo thresholds for irregular sequences of five events. Globally, sensitivity for these irregular sequences was of an intermediate level between that of the simple duration task and the sequences with five regular events. The results are discussed in terms of the hypothesized ‘‘regularity detectors.’’
Identification of multidimensional complex sounds having parallel dimension structure
Authors: Christopherson Laurel A. , Humes Larry E. , Watson Charles S. , Kidd Gary R.
Published: October 2005

Ten listeners classified complex sound pulses that differed on three independent and discriminable dimensions. The stimuli were 100-ms sound pulses synthesized using five simultaneous sinusoids used previously by Kidd and Watson [J. Acoust. Soc. Am. Suppl. 1 81, S33 (1987)]. The three dimensions of the stimuli were harmonicity, spectral shape, and amplitude envelope. Each dimension could take on one of two possible values, target or nontarget. Eight stimuli were synthesized using all possible combinations of the dimension values. Subjects were trained to label two stimuli, the two having either all-target or all nontarget values, as ‘‘+’’ and ‘‘0,’’ respectively. Following this training, subjects were asked to classify all eight stimuli as either ‘‘+’’ or ‘‘0.’’ Results indicated that listeners preferred one dimension when classifying the stimuli. However, the preferred dimension was not the same for all of the listeners. In addition, it was demonstrated that listeners could be trained to use a dimension that they previously were not using to classify the sound pulses. [Work supported by NIA and AFOSR.]
Intensity changes in two-tone complexes and model predictions
Authors: Versfeld Niek J.
Published: October 2005

Thresholds for changes in the amplitudes of two-tone complexes have been measured as a function of frequency separation. Thresholds have been measured both with and without a roving intensity level. Thresholds for changes in the spectral shape of two-tone complexes were generally lower than those for changes in the overall intensity. Thresholds for the former changes are lowest at a frequency separation of one semitone. Thresholds for the latter decrease as a function of frequency separation. Two models have been applied to the data. For narrow-bandwidth signals, the EWAIF model [Feth, Percept. Psychophys. 15, 375–378 (1974)] can partly account for the obtained data. For (partially) resolved signals, the two-channel model [Durlach etal., J. Acoust. Soc. Am. 80, 63–72 (1986)] can account for the data. The obtained values of the model parameters show that the auditory system does not process the changes optimally, in the sense that amplitude changes in one component contribute more in the detection process than those in the other component. [Work supported by the Netherlands Organization for the Advancement of Pure Research (NWO) through the PSY-CHON Foundation, Grant No. 560-262-034.]
Effects of spectral and temporal uncertainty on the detection of increments in the level of individual tonal components of ‘‘profile’’ stimuli
Authors: Watson Charles S. , Li Xiaofeng
Published: October 2005

In a modification of the previous profile discrimination experiments [summarized in D. M. Green, ProfileAnalysis (Oxford U.P., New York, 1988)], intensity increments were introduced at one of ten temporal positions during the overall duration of 11-tone profiles. When both the frequency and the temporal position of intensity increments are varied randomly from trial to trial, listeners show considerably lower thresholds for increments occuring at higher frequencies, and later in time. These results are, in some cases, quite similar to the distribution of selective attention to individual tonal components of multi-tone patterns [e.g., Watson etal., J. Acoust. Soc. Am. 60, 1176 (1976)]. When the same increments are presented under minimal-uncertainty conditions (i.e., at a single frequency and temporal position within each block of adaptive-tracking trials), thresholds are reduced by as much as 6–8 dB. These results suggest a common source of the distribution of selective attention to the components of sequences of individual tones, and to the spectral-temporal range of the spectrally richer and more representative profile stimuli. [Work supported by NIH and AFOSR.]
Teaching about turbulence noise sources in the vocal tract
Authors: Stevens Kenneth N.
Published: October 2005

Several classes of speech sounds are produced by causing fluctuations in airflow in the vocal tract. This turbulence usually occurs in the vicinity of a constriction at the glottis or in the vocal tract, and results in the generation of noise sources. Teaching about these turbulence noise sources begins by introducing some basic ideas about airflow through orifices and the nature of turbulent flow, using analogies with other familiar situations in which turbulent flow occurs. Different mechanisms of sound-source generation as a consequence of turbulent flow are then considered, including fluctuating flow within an orifice, at a boundary, and in the absence of boundaries. Finally, the coupling of these sources with the vocal-tract modes is discussed, to indicate how fricatives, bursts, and other noise-like sounds are produced. Simple tubes and orifices with airflow are used to demonstrate the different mechanisms of turbulence noise generation and of noise excitation of resonators.
The role of resonators in voiced speech production
Authors: Strong William J.
Published: October 2005

A resonator is something that responds to and enhances some frequencies with respect to other frequencies. In the simplest case of voiced speech production, an air-filled tract acts as a resonator that selectively enhances certain frequencies of the periodic glottal airflow as it passes from one end to the other end of the tract. As a simple example for demonstrating the effect of a resonator, the tract is represented by a cylindrical tube closed at the ‘‘glottal’’ end and open at the ‘‘mouth’’ end. The resonance frequencies of the tube−those frequencies of the airflow that will be enhanced as the air passes through the tube−can be determined easily from the tube’s length and end conditions. The resonance frequencies−formant frequencies in the terminology of speech science−are verified by clapping a cylindrical tube to produce a tube spectrum. The tube is then driven with an artificial larynx as a source to produce a ‘‘voiced’’ sound. The spectrum of the sound is seen to be the sum of the source spectrum and the tube spectrum (when expressed in terms of sound levels). The production of ‘‘vowel’’ sounds by this method will be demonstrated.
Teaching how the vocal tract resonates
Authors: Ladefoged Peter
Published: October 2005

Students of speech communication need to know why a particular shape of the vocal tract has the resonances (and antiresonances) that it does. They also need to know what will happen to the formants if the vocal tract shape changes in a certain way. It is easiest to begin by studying the resonances of a neutral vocal tract. Having shown why the first three resonances of such a tube are at approximately 500, 1500, and 2500 Hz, the next step is to demonstrate the existence of nodes and antinodes in the pressure and flow waveforms. It is then possible to establish the relation between added constrictions in the vocal tract and changes in the formant frequencies, the easiest to explain being the effect of lip rounding. Another important concept is the way in which part of the vocal tract may act as a Helmholtz resonator, producing the low frequencies often associated with the first formant. All these concepts may be used in an explanation of diagrams showing the relation between formant frequencies and vocal tract shapes such as Fant’s nomograms. Finally, the notion of an antiresonance may be used in an explanation of the effects of nasal coupling.
The mu+ system for speech database analysis
Authors: Harrington Jonathan , McVeigh Andrew
Published: October 2005

The mu+ system has been developed in response to the increasing availability of phonetically and prosodically segmented speech databases. Using an on-line dictionary and pronunciation rules, mu+ maps waves derived label files containing phonetic segments onto phonemes that are in turn parsed into words. Foot and syllable structure are built automatically between word and phoneme levels and a grid is provided to enter (manually) intonational transcriptions. The different kinds of labels can be identified both hierarchically and sequentially. A typical search and display pattern would be: ‘‘display as ellipses in the formant plane F1 and F2 at the vowel target of all accented front vowels that follow syllable-initial oral stops in trisyllabic content words that are phrase-final and utterance-medial.’’ Analogous search patterns and displays can also be made for segmented speech production (movement, EMG) data and any combination of segments can be D/A converted. The mu+ system has applications not only in speech recognition and text-to-speech, but also in many classic speech research areas such as coarticulation, vowel undershoot, and speech timing.
An acoustic-phonetic hypercard stack for undergraduates: A first report
Authors: Ralston James V. , Schwartz John , Van Son L. George
Published: October 2005

An interactive hypercard stack is being developed for pedagogy in several departments at a liberal arts college. The stack is intended for use as a supplement to introductory level courses in physics, psychology, speech pathology, audiology, and linguistics. Therefore, the stack is designed for students with no background in acoustic-phonetics, mathematics, or physics. At present, the stack is primarily concerned with vowel sounds. The three main content domains of the stack are speech production, auditory/speech perception, and the physics of sound. In addition, there is a hypertext-oriented glossary and a corpus of speech sounds with associated articulatory and acoustic information.
A profile analysis model
Authors: Berg Bruce G.
Published: October 2005

Recently developed analytic techniques, aptly described as molecular psychophysics, are used to provide a detailed, quantitative description of performance in auditory profile tasks. These techniques consist of a trial-by-trial analysis of the relationship between the stimulus and a listener’s response. One application is used to estimate weights associated with each spectral component of a complex sound (e.g., spectral profile). Another application is used to estimate the inherent variability, or internal noise, of the listener. Estimates of spectral weights and internal noise are consolidated in a model that is completely determined by the data (no free parameters). Results are consistently within 2 dB of listeners’ observed thresholds. The model is applied to the data from several profile discrimination experiments, including the use of narrow band, wideband, and rippled spectra. [Work supported by ONR.]
Spectral-shape discrimination of 3-component complexes as a function of the center frequency
Authors: Dai Huanping , Green David M.
Published: October 2005

Thresholds for profile detection using 3-component complexes were measured at different frequency regions. The frequency ratio of adjacent components was fixed at 1.37. In the standard spectrum, all components were equal in level. In the standard-plus-signal spectrum, the center component was increased in level. The overall level of the standard was randomly selected from the range 45–65 dB SPL on each presentation. Thresholds were estimated using a maximum-likelihood procedure in a two-alternative, forced-choice task. The center frequency of the complex was varied from 250 to 4000 Hz. The results from three observers show that as the center frequency of the complex increases it is more difficult to detect the signal−the threshold at 4000 Hz was nearly 20 dB above that at 250 Hz. Possible explanation for these results are discussed. [Research supported by NIH and AFOSR.]
Using piezoelectric film and resonant ultrasound to determine the elastic tensor of small, fragile samples
Authors: McKenna M. J. , Spoor P. S. , Maynard J. D.
Published: October 2005

The ultrasonic and elastic properties of materials are conventionally measured using quartz or lithium niobate transducers in a pulse-echo technique where the transducer is driven at resonance. Some problems include transducer ringing, transducer bonding, parallelism of the sample faces, beam diffraction, and the necessity of remounting transducers in order to measure all the elastic constants. Nearly all these problems disappear if a resonance technique is used, and all the elastic constants may be determined with a single measurement. For broadband response, minimum loading by the transducer required for a resonance measurement, and to gently contact a small fragile sample, polyvinylidene flouride (PVDF) piezoelectric film as thin as 9 μm is ideally suitable. Small active areas and leads are produced with metalization patterns on each side of the PVDF film. For resonance measurements, electrical crosstalk across the small sample is processed by frequency modulating the drive and using phase sensitive detection. Small fragile samples such as aerogels or biological crystals with dimensions of only a few hundred microns may be measured with large signal-to-noise ratios. Recent measurements on the elastic anisotropy of a single grain Al–Cu–Li quasicrystal are two orders of magnitude more sensitive than conventional pulse-echo techniques and should help to resolve some of the questions over the structure of actual quasicrystalline materials. [Work supported by NSF Grant No. DMR-9000549 and by the Office of Naval Research.]
Filter order in maximum entropy echo identification
Authors: Roemer Louis E.
Published: October 2005

The use of maximum entropy methods to detect ultrasonic echoes through phase information requires one to choose a filter order for signal processing. Choice of too low an order filter will cause the observer to miss multiple echoes. Choice of too high an order filter order will result in spurious echoes. Criteria for selection of filter order, based on noise power, are useful and well known. However, additional insight can be gained from observing the echo pattern as the filter order is changed. The effect of filter order selection is demonstrated for thin metal sheet targets, in which the echo return overlaps the original transmitted signal. Nonspurious echoes are shown to vary less with filter order than spurious echoes, thereby allowing identification.
Frequency and time domain modeling of acoustic liner boundary conditions
Authors: Bliss Donald B. , Peretti Linda F.
Published: October 2005

A general model of the acoustic boundary condition for an absorptive liner, including bulk reaction effects, is developed. The boundary condition is an extension of the traditional impedance boundary condition modified to include the effect of gradients along the absorptive surface. Using a bulk-reacting porous material as an example, a boundary condition of this type can be constructed that is applicable over the entire frequency range, provided the liner damping is sufficiently high. The boundary condition contains frequency-dependent coefficients, in addition to the normal incidence impedance. These coefficients can be calculated theoretically or measured experimentally. A measurement procedure is described. The accuracy of the boundary condition is demonstrated by comparison with the exact solution for some simple cases. The boundary condition is converted from frequency domain to time domain by calculating the appropriate impulse response behavior. The various impulse response functions are derived by Fourier transform methods, and their behavior is discussed. The impulse functions exhibit behavior associated with viscous relaxation and wave reflection. Once the impulse response functions are known, a response to an arbitrary pressure field can be constructed. Future extensions and applications of the bulk reacting boundary condition are described briefly. [Work supported by NASA.]