bluuebunny/crossref_metadata_2025 huggingface dataset:
Extracting stable measures from unstable psychometric performance
Authors: Leek Marjorie R. , Hanna Thomas R. , Marshall Lynne
Published: October 2005

At the last meeting of the Acoustical Society of America, a new method of estimating slopes of psychometric functions when thresholds were varying over the course of an experiment was reported. Computer simulations were used to calculate the relationship between the true slope of the function and the variability across pairs of trials in two interleaved adaptive tracks. Further, the variability across all trials in the two tracks was shown to indicate the stability of the function over time. Here, the validation of that procedure in the measurement of human performance is reported. Listeners were asked to detect a 1000-Hz tone in a narrow band of noise, the level of which was either fixed or varied sinusoidally from trial to trial. This masker variation imposed systematic threshold shifts analogous to those in the simulations. The period of sinusoidal shift ranged from 16–128 trials per cycle, and the amplitude of shift was either 4 or 8 dB. Psychometric functions for individual listeners were constructed both by traditional methods of fitting the trial-by-trial data, and by using the functional relationship between slope and variability determined from the computer simulations. When the imposed threshold shift did not occur too rapidly, the newer method provided more stable slope estimates than did the traditional fitting methods. Under certain conditions the procedure was able to identify the unstable performance. These results show that this interleaved tracking technique can produce more stable slope estimates, and can alert experimenters to drifting in subject performance.
Increased CMR resulting from lowered levels of masker band
Authors: Cohen Marion F.
Published: October 2005

Comodulation masking release (CMR) for a 1000-Hz, 50-ms sinusoidal signal was measured in the usual manner, except that the overall level of the 100-Hz-wide masking noise band was varied from 18 to 78 dB SPL in 20-dB steps. The cue band was always 78 dB SPL. Presentation was either monotic or dichotic, with the cue band presented to the nonsignal ear only. Results show CMR values averaging as high as 7 dB when the cue band was centered at 8000 Hz, three octaves away from the signal, and 12 dB when it was two octaves away. CMR was greatly reduced when the two noise bands were at equal levels. These results were essentially replicated dichotically, except that CMR was somewhat smaller. These data relate to previous results showing poorer detectability of a noise-band signal when a synchronous noise band is introduced at a distant frequency [M. F. Cohen and E. D. Schubert, J. Acoust. Soc. Am. 81, 721–723 (1987); D. McFadden, J. Acoust. Soc. Am. 81, 1519–1527 (1987)]. [Work supported by AFOSR.]
Functional involvement of auditory cortex in fine frequency discrimination of biosonar signals in the bat
Authors: Riquimaroux H. , Gaioni S. J. , Suga N.
Published: October 2005

The auditory cortex of the mustached bat, Pteronotus parnellii, has a highly specialized region, the DSCF (Doppler-shift constant frequency) area, which overrepresents the frequencies of Doppler-shifted echoes (about 61 kHz), and, hence, may play a role in the fine frequency discrimination necessary for detailed velocity measurements (Suga, 1984). This function of the DSCF area was behaviorally tested using reversible ablation with muscimol, a potent GABA agonist (Hilosaka et al., 1985). Bats were trained on a discriminated shock avoidance task requiring a leg flexion response. The stimuli were trains of artificial pulse-echo pairs (tone bursts). For S +, the pulse and echo were the same frequency (e.g., 61.0 kHz), while, for S −, they were different frequencies (e.g., 61.0 and 61.1 kHz). Following baseline testing, 0.1–0.2 μℓ of muscimol (1 μg/μℓ saline) was bilaterally applied to the DSCF area. The bats failed on previously successful discriminations of small frequency differences (e.g., 50 Hz), but succeeded on discriminations of large frequency differences (e.g., 2 kHz). Discrimination performance returned to baseline levels within 24 h. These data indicate that the DSCF area is necessary for fine frequency discriminations involving biosonar signals. [Work supported by AFOSR Grant No. 8743250.]
Modal matched-field processing with small aperture vertical arrays
Authors: Smith G. B. , Frichter G. M. , Feuillade C. , Byrne C. L.
Published: October 2005

In the past years, much attention has been given to mode space matched-field processing in shallow-water, low-frequency environments. In this type of situation, mode space matched-field processing has several advantages over conventional matched-field processing such as: lower processing dimension, better sidelobe rejection, and the ability to discriminate signal against modal noise. However, extraction of modal amplitudes from data by the method pioneered by Shang [J. Acoust. Soc. Am. 77, 1413–1418 (1985)] is not very reliable for small aperture arrays. Attempts have been made to overcome this problem in order to make mode space matched-field techniques more applicable to actual physical environments [T. C. Yang, J. Acoust. Soc. Am. 82, 1736–1745 (1987)]. A fresh approach is brought to this problem here using computer simulations of matched-field processing in a Pekeris waveguide to demonstrate a new method for extracting modal amplitudes from data. These simulations demonstrate that this new method of modal amplitude extraction allows successful mode space matched-field detection and localization at array apertures significantly smaller than those allowed by previously used algorithms. Finally, the algorithm will be discussed in the general context of dimensional reduction techniques.
The transport coherence of acoustic fields as a criterion for range dependence in an ocean channel
Authors: Hurdle B. G.
Published: October 2005

A paper was presented [B. Hurdle, J. Acoust. Soc. Am. 55, 452 (1974)] on the nature of acoustic interference fields in a geometric dispersive ocean channel. The structure and multiple components of the fields as a function of the velocity of the source, the geometry of the boundaries, and the sound-speed structure of the channel were discussed. This paper reviews the nature of the fields and, based on the field characteristics, provides a criterion for determining the degree of range dependence of an ocean channel. This criterion is based on the transport coherence of the field and is measured in terms of the correlation coefficient of the transported field as a function of the separation distance between two receivers as the field is transported the distance between them. The difference in the transport coherence and the coherence in the field for beam forming will be discussed. Some other applications of this concept will also be discussed.
Context sensitivity in profile analysis by parallel distributed networks
Authors: Howard James H. , Miller Maxwell H. , Harpster Jeffrey L.
Published: October 2005

Spectral shape discrimination with uncertain signal frequency was simulated using simple, three-layer, feed-forward, parallel networks. As in previous studies, pairs of multitonal sounds were used—one with all components at a uniform, background level and the other with a tonal signal added in-phase to a single component [Howard et al., J. Acoust. Soc. Am. Suppl. 1 85, S37 (1989)]. Networks were trained to select the signal alternative in a 2 AFC paradigm using the generalized delta rule with error propagation and were then tested under various conditions. The signal frequency was either fixed or varied across trials, and stimulus pairs were presented either alone or with a cue that was correlated with the signal frequency. Network sensitivity was greater with the fixed signal frequency than with the varied signal frequency and this difference was reduced by the presence of a predictive cue. As in earlier experiments, these stimulation results mimic previous findings with human listeners under similar task conditions. [Work suported by ONR.]
A coarticulation model for continuous digit recognition
Authors: Pieraccini Roberto , Rosenberg Aaron E.
Published: October 2005

Between-word coarticulation is one of the major problems in continuous speech recognition since it modifies the acoustic characteristics at the boundaries of words. With large vocabulary speech recognition the problem can be solved by introducing the concept of interword units. If the vocabulary is small enough (e.g., digits) all possible coarticulations between all words of the vocabulary can be modeled. In this study every digit is represented by three segments, namely, a core segment that can be assumed reasonably insensitive to any coarticulation effect and head and tail segments that represent, respectively, the initial and the final part of every work spoken in isolation. In addition, a set of juncture segments is defined that represent the junction between every possible pair of words. The recognition process is driven by a regular grammar that represents all the allowed segments sequences, namely, digits spoken in isolation as well as sequences of digits spoken continuously or with pauses between words. Every segment is represented by a mixture density HMM. Since the vocabulary is composed of 11 words (digits 0 to 9 and oh) the overall number of models is 154 (11 cores, 11 heads, 11 tails, and 121 junctures). A number of experiments were carried out using the TI connected digits database, which consists of a set of digit strings. Recognition was performed with different experimental conditions, i.e., varying the number of mixture components, the number of models per segment, and the structure of the connection between words. String error rates of 3.27% and 1.97% were obtained for the best and the two best hypothesized strings. These results compare favorably with those obtained with whole word modeling.
An evaluation of a spectral moments metric with voiceless fricative obstruents
Authors: Tomiak Gail R.
Published: October 2005

The classification efficacy of a spectral moments metric was tested on a corpus of voiceless fricatives. The metric classified phonetic identity on the basis of mean, variance, skewness, and kurtosis values derived from cross-sectional spectra. The classification power of both linear- and Bark-based versions of the metric was tested using a corpus of 420 voiceless fricatives (/f/, /θ/, /s/, /∫/, /h/) obtained from multiple talkers and vowel environments. Discriminant function analyses performed on linear and Bark moment profiles of well-identified fricative tokens resulted in overall classification accuracies of 78% and 74%, respectively. Classification accuracies were substantially higher when the nonsibilant (/f/ and /θ/) data were excluded. In an attempt to improve metric classification of nonsibilant tokens, an experiment designed to identify the perceptually appropriate location of a fricative analysis window was performed. The results suggested that nonsibilant classification performance of the metric may be substantially improved when moment information is based upon the spectral information contained within the onset and/or offset (transition) portion of the frication. [Work supported by NINCDS Grant NS19653 and NIDCD Grant DC00219 to SUNY at Buffalo.]
Auditory word recognition and the locus of word frequency effects
Authors: Connine Cynthia M. , Titone Debra
Published: October 2005

The locus of lexical frequency influences in auditory word recognition was investigated using a speech identification paradigm. Subjects were required to identify the initial phoneme of tokens from voicing speech continua. Here 44 continua were constructed such that the voiced and voiceless endpoint of each pair contrasted in lexical frequency. Identification responses for ambiguous stimuli tended to be the higher-frequency member. Reaction times for unambiguous stimuli showed an advantage for the high-frequency responses; ambiguous stimuli showed no such reaction time advantage. The patterns of reaction time for word frequency were similar to influences of monetary payoff [Connine and Clifton, J. Exp. Psychol.: Hum. Percept. Perform. 13, 219–299 (1987)] and support a post-perceptual influence of word frequency. [Work supported by NIH.]
A mixture modeling approach to text-independent speaker ID
Authors: Reynolds Douglas A. , Rose Richard C. , Smith Mark J. T.
Published: October 2005

Automatic speaker identification (ASI) systems generally fall into two classes: text dependent and text independent. High recognition rates for short utterances (<1 s) are more common for text-dependent systems since the process benefits from the a priori knowledge of the underlying acoustic-phonetic stream. Without this added information, text-independent ASI usually requires long test utterances for averaging out the unknown phonetic variations. In this paper a new technique is introduced to bridge the gap between text-dependent and text-independent ASI, which allows for high recognition rates using short utterances in a text-independent ASI system. Speakers are parametrically represented by a Gaussian mixture probability density function, where the parameters are maximum likelihood estimates obtained from a form of the iterative estimate-maximize (EM) algorithm [G. J. McLachlan, Mixture Models (Dekker, New York, 1988)]. The components in the mixture model can be considered to represent “hidden” acoustic classes so that during testing an utterance is automatically segmented into subword units for comparison to a speaker's subword models, as in text-dependent ASI, allowing for reliable scoring over short utterances. An application of this technique is for speaker spotting in conversational utterances over long distance telephone connections, where speaker identity may change frequently in time. A preliminary study [R. C. Rose and D. A. Reynolds, “Text independent speaker identification using automatic acoustic segmentation,” ICASSP-90] on a clean conversational database has shown great promise, producing an 89% recognition rate for 1-s test utterances. This new ASI technique and various channel compensation methods will be applied to a telephone database and results will be presented at the conference.
A study of acoustical enhancement systems, leading to the use of time-variant synthetic reverberation—Part I
Authors: Griesinger David , Tanner Robert H.
Published: October 2005

The rehabilitation of the Elgin Theater in Toronto, built in 1913, necessitated the installation of an acoustical enhancement system. This led to a study of the various existing systems, most of which involve complex system design, high expense, marginal stability, and sound coloration. This paper develops some simple mathematical techniques for analyzing such systems by examining a few examples. After outlining the advantages and disadvantages of each, it then suggests a new system using time-variant synthetic reverberation that appears to solve many of the previous problems.
Measuring the vertical directional spectra caused by sea surface sound
Authors: Kennedy Robert M.
Published: October 2005

The objective of the measurements discussed here is the development of a database of directional spectra obtained in an acoustically isolated area (Tongue of the Ocean, the Bahamas) where the acoustic ambient is principally controlled by the local (observable) sea surface conditions. The purpose of the database is the identification of a model of the space-time statistics of sea surface sound. A wideband (40- to 4000-Hz) vertical acoustic linear antenna system was used. Data logging took place in a remote autonomously operating subsurface buoy system. Recording restrictions limited each of the seven octavely nested apertures of the sensor system to four wavelengths. However, quite adequate spatial resolution of the directional spectral estimates was obtained using a physically based parametric model of the random process. The results of the initial 2-month deployment (wind speed range of 2–16 m/s) are discussed here. The measured directional spectra are displayed as a function of frequency, elevation angle, and surface friction velocity. The sea surface sound source structure is discussed.
When is a stop aspirated
Authors: Lisker Leigh
Published: October 2005

A simple answer to the question is this: A stop is aspirated when a linguist/phonetician judges it to be so. This property, often referred to in descriptions of the stop consonants of English and many other languages, has been variously defined in physical terms, aerodynamically, articulatorily, and acoustically. Linguists seem generally to agree that in English aspiration is a predictable, i.e., context determined, and hence a nondistinctive attribute of/p t k/in certain contexts. In some other languages, e.g., Hindi, aspiration is a distinctive feature of certain stop phonemes, so that [t] contrasts with [th]. Hindi-English bilinguals, however, when speaking English, regularly use Hindi /t/ and not/th/, where a “predictably” aspirated English/t/would be expected. For native English speakers, including linguists, this choice seems erroneous. Consistent with this choice by Hindi speakers, is the general opinion of many Hindi-speaking linguists that, contrary to the opinion of American linguists, English /p t k/are not aspirated. Labeling tests of edited naturally produced nonsense CV syllables yielded data showing that Indian and American linguists differ considerably in locating an inaspirate-aspirate boundary along the VOT dimension.
Medial voicing distinctions in English trochees
Authors: Abramson Arthur S. , Lisker Leigh , Koenig Laura
Published: October 2005

Phonetic studies of distinctive consonantal voicing in English have focused on utterance-initial and final positions. Much less attention has been given to medial stops following stressed and preceding unstressed vowels, as in pairs like tugging/tucking. The high frequency of occurrence of unaspirated voiceless stops in this context is commonly overlooked because of the usual emphasis on the aspiration of/p t k/in stressed syllables. The voiced stops in this context are likely to have glottal pulsing through much or all of the closure, unlike their utterance-initial counterparts, which may not show glottal pulsing until the moment of release. For this study, a number of such word pairs were recorded in carrier sentences at a normal rate by three native speakers of American English: Three acoustic features were measured: voice timing, closure duration, and amplitude of release burst. Values of the first two features were clearly different for the two categories. With respect to the third feature, burst amplitude, the two categories differed much less clearly. To generate sets of stimuli for perception testing, the three features were covaried incrementally by waveform editing. Labeling data collected from a jury of native American English listeners will be presented to show the relative efficacy of these features for perception.
The effect of lung volume on voice onset time
Authors: Solomon Nancy Pearl , Hoit Jeannette D. , Hixon Thomas J.
Published: October 2005

The mechanical linkage between the respiratory apparatus and the larynx may have consequences on the acoustic speech signal. For example, at high lung volumes, the diaphragm usually flattens and pulls the trachea and larynx caudally, exerting a force that abducts the vocal folds. This abductory force may delay the onset of vocal fold vibration that could be realized acoustically as a delay in voice onset time (VOT). To examine this possibility, five young men produced repetitions of a six-syllable phrase containing two stressed /pi/ syllables from the total lung capacity to the residual volume. Lung volume was determined with magnetometry and VOT for each /pi/ syllable was measured from spectrograms. VOT generally was longer at higher lung volumes than at lower lung volumes in four of the five subjects. This finding supports our hypothesis that the mechanical linkage between the respiratory apparatus and larynx can influence voicing onset. Further, it suggests the need to take lung volume into account when using VOT as an indicator of laryngeal behavior in individuals with speech and voice disorders. [Work supported by NIH.]
A two-microphone method for source strength measurement
Authors: Moritz C. , Lamancusa J. S.
Published: October 2005

In studies of outdoor sound propagation, it is common practice to measure the source strength by placing a single reference microphone in close proximity to the source (typically 1–3 m). In this method, the ground impedance is measured at some time and is assumed to remain constant during the measurement period. If measurements are recorded over long periods of time (several minutes or several hours), this assumption may nol hold. Utilizing this single reference microphone technique, apparent sizable fluctuations in source strength have been recorded under conditions when the source strength is known to be constant. Therefore, a two-microphone technique to simultaneously determine source strength and ground impedance has been developed. This technique is an extension of the two-microphone technique for ground impedance measurement developed by Glaretas et al. [J. Acoust. Soc. Am. Suppl. 1 67, S73 (1980)]. This method takes into account fluctuations in ground impedance so that consistent source strength measurements can be obtained.
STI approach for predicting the effect of fluctuating interference on speech intelligibility
Authors: Bronkhorst Adelbert W. , Houtgast Tammo
Published: October 2005

The modulation transfer function (MTF) and the derived speech transmission index (STI) have been successfully applied to predict the effects of interfering noise and time-domain distortions on speech intelligibility. Results of experiments using fluctuating interference (e.g., chopped noise) show, however, that the STI can grossly underestimate performance in such conditions. It was found that STI predictions for fluctuating interference can be considerably improved by modifying the calculation scheme. The normal approach is based on calculation of the modulation reduction m according to the simple relation m = 1/ (I + IN/IS), where the signal and noise intensities IN, and IN are averaged over a certain time interval. In the modified approach, m instead of noise intensity is averaged, and, additionally, a low-pass filter is applied to the intensity envelope function to account for the limited auditory temporal resolution. Predictions based on this method are generally quite good, except for an underestimation of the masking efficiency of slowly fluctuating signals (with modulation frequencies below 4 Hz).
Moving load on a Timoshenko beam
Authors: Mackertich Seroj
Published: October 2005

This paper presents the dynamic response of a simply supported beam excited by moving loads based on the Timoshenko beam theory with corrections for the shear-deformation and rotatory inertia effects. Numerical results for various velocities of moving loads are given, and results are compared with the response of a Bernoulli-Euler beam theory.
Modeling and analysis of chain drive systems
Authors: Liu Shyi-Ping , Wang K. W. , Hayek Sabih I.
Published: October 2005

Chain drive noise caused by meshing between the chain link and the sprocket has been recognized to be an important problem in the design of automotive powertrains. This paper presents a new dynamic model that integrates the local meshing mechanics with the global chain vibration characteristics. The chain is modeled as an axially moving, heavy string under tension. The upstream boundary condition is considered to be fixed, while the downstream boundary condition depends on the impact load between the sprocket tooth and the meshing chain link. The chain/sprocket impact load is derived based on the Lagrange's impact equation, and is a function of the system's dynamic characteristics. A parametric study is performed based on this model. The relationships between the impact load and the important system parameters, such as chain tension, chain speed, chain inertia, chain length, and pitch, are quantified. The chain response under the meshing effect is analyzed. The results of this fundamental research are useful in optimizing such mechanical systems. [Work supported by General Motors Corporation.]
Investigation of headphones suitable for psychophysical experiments
Authors: Hirahara Tatsuya , Ueda Kazuo
Published: October 2005

In order to find the appropriate headphone to use in psychophysical experiments, the frequency responses of 12 headphones were measured by three physical methods: on an IEC coupler (B&K 4134), on a C coupler attached to a head and torso simulator (Kohken SAMRAI) (Okabe et al., J. Acoust. Soc. Jpn. (E) 5, 95–104), and by using a probe microphone in real ears. The results showed that very few electrostatic circumaural headphones (e.g., STAX SR-Lambda Pro.) have relatively flat frequency characteristics, with excellent invariance among the measuring methods. In contrast, many dynamic supraaural headphones (e.g., Rion AD02, Beyer DT48, Elega DR831, etc.) have poor frequency characteristics, especially at lower frequencies, with many differences occurring between the three measuring methods. For these headphones, the energy leakage at the lower frequency region is inevitable, since the headphone pad fitting to the pinna is usually incomplete, and acoustic impedance of the diaphragm is very high. These undesirable characteristics might affect the results obtained in psychophysical experiments. As an example, results were severely affected by headphone differences in the vowel identification test using several synthesized vowels where the F0 component amplitude was manipulated. The physically determined frequency responses were compared with those from a psychophysical loudness matching procedure [K. Ueda and T. Hirahara, J. Acoust. Soc. Am. Suppl. 1 87, S142 (1990)].
Vowel nasalization: An acoustic and perceptual study of natural and synthetic vowels
Authors: Gilbert H. R.
Published: October 2005

The purpose of the present study was to determine the acoustic cues used by speakers to nasalize vowels. These cues were then used to synthesize nasal vowels. The synthetic vowels, along with their naturally produced counterparts, were presented to a group of judges to rate vowel identity, naturalness, and the presence or absence of nasality. Speech samples were obtained from ten speakers who produced the vowels /i, u, æ, a, ɔ, ʌ/in a variety of non-nasal and nasal contexts. DFT analyses were performed on the utterances. Differences between the non-nasal and nasal vowels were studied by comparing pole frequency, amplitude and band-width values, zero frequency values, F0 values, first-harmonic amplitude in relation to first-formant amplitude (H1-A1) values, spectral tilt, and open-quotient values. The 1988 version of the Klatt synthesizer was used to synthesize nasality in vowels based upon the cues identified. Accuracy of the judges perceptions of the [+ nasal] feature in the synthesized vowels was dependent on the presence of a nasal pole-zero pair in the vowel spectrum. Frequency of the nasal pole-zero pair varied with the specific vowel being synthesized.
Mapping the organization of vowel sequences into words
Authors: Chalikia Magdalene H. , Warren Richard M.
Published: October 2005

Previous research [Warren et al., Percept. Psychophys. (in press)] has shown that when listeners hear recycled sequences of steady-state vowels, they do not perceive them as a succession of vowels. Illusory phonemes are introduced and real or nonsense words are heard. Often the sequence of vowels is split perceptually in two simultaneous words differing in both quality and phonemic content. The present study employed sequences of eight 80-ms vowels, and mapped the perceptual phonemes to acoustic phoneroes by terminating the repeated sequence at various positions and determining the last sound heard in the perceived word for each position. When two simultaneous words were heard, they both were mapped. Relations between the acoustic phonemes and the perceived phonemes will be described, and implications concerning the perceptual organization of speech will be discussed. [Work supported by NIH and AFOSR.]
Wave propagation in plate structures
Authors: Yong Y. , Mani G.
Published: October 2005

This study is concerned with dynamic response of complex plate structures, composed of rigidly connected thin plates, subjected to point or line loadings. The finite dement method is not recommended for this analysis since a very large number of meshes are needed to obtain accurate results in the high-frequency range. Although the finite strip method only requires strip refinement in one direction, the computational efforts may be still tremendous if the structure is large or contains a large number of thin plates. However, the computational efforts can be greatly reduced when the finite strip method is combined with a new method developed by Yong and Lin [Y. Yong and Y. K. Lin, J. Sound Vib. 129(2), 99–118 (1989)]. In this approach, each strip with various boundary conditions is treated as a waveguide capable of transmitting different wave motions. A scattering matrix is defined to characterize wave motions at boundaries, locations where two plates are joined, and where the type of waveguides are changed. Thus a response at any location can be derived in terms of the scattering matrices. Three main advantages can be found in this analysis. First, the numerical computation is stable since it always proceeds in the directions of wave motions. Second, a strip refinement will not result in an increase in computer time. Third, the response is exact except for the modeling by finite strip approximation. Numerical examples will be given in light of the new analytical approach.
Propagation of elastic waves on thin cylindrical shells at frequencies below the ring frequency
Authors: Kouzoupis Spiro
Published: October 2005

Recent experimental results (to appear) by Williams and colleagues at NRL suggest that wave propagation concepts apply to vibrations caused by point excitations on circular shells even at frequencies substantially below the ring frequency. A previous analysis by Pierce and Kil [ASMEJ. Vib. Acoust. (1990)] has given general dispersion relations for such waves, but with detailed analysis only for frequencies far above the ring frequency. The present paper examines the propagation in the opposite limit. In this limit two types of propagating waves are excited, which in the axial direction correspond to shear waves in bulk media [phase velocity equal to (G/p)1/2] and to longitudinal waves in rods [phase velocity equal to (E/p)1/2]. At low frequencies these waves are nondispersive with phase velocities that do not vary with frequency. The propagation, however, is anisotropic and the mode that nominally coresponds to shear waves has zero phase velocity for propagation in the circumferential direction. Hence wave fronts for the two types of waves are not circular; the “shear wave's” lines of constant phase resemble figure-eight's aligned in the axial direction; those for the mode that nominally corresponds to the longitudinal (thin rod) mode resemble ellipses elongated in the axial direction. Numerical examples are also presented for amplitudes and radiation patterns associated with point excitation. [Work supported by ONR and by the William E. Leonhard endowment to Pennsylvania State University. The author acknowledges the advice of A. D. Pierce.]
Active structural acoustic control with shape memory alloy hybrid composites
Authors: Rogers Craig A.
Published: October 2005

Active structural acoustic control using shape memory alloys (SMA) hybrid composites has been demonstrated both theoretically (Rogers et al., AIAA-89-1389) and experimentally (Saunders et al., AIAA-90-1090). Theoretical models involving fundamental constitutive relations of SMAs, micromechanical analysis of the hybrid material system, macromechanical constitutive relations, as well as dynamic and structural acoustic radiation have been developed and exercised to evaluate the potential of active control of beam, plate, and shell structures using active strain energy tuning. Frequency placement control of a graphite-epoxy beam with 15% nitinol volume fraction was experimentally demonstrated to minimize the structurally radiated noise from a clamped-clamped beam. The reduction in the sound-pressure level for the beam excited at 35 Hz (n = 1 mode) and 145 Hz (n = 4 mode) was 25–30 dB. These results, and others, have shown the validity and potential of new material/structure paradigms to minimize structural radiation using adaptive and distributed fibers embedded in hybrid composite material systems. [Work supported by ONR Grants N00014-88-0566 and N00014-88-0721.]
Piezoelectric and electrostrictive transducers, sensors, actuators, and smart materials
Authors: Cross L. Eric , Pugh Evan
Published: October 2005

This talk will discuss recent developments in large area acoustic sensors that become possible using composite piezoelectric transducers. The design rules for the composites will be briefly reviewed and the immense improvement in figure of merit that can be encompassed will be delineated. The development of materials for agile switching transducer arrays will be discussed and electrostrictive materials described that can be controlled for both phase and amplitude in transmit and receive modes. Polarization control in the electrostrictive family of electroceramics permits both direct dc field control and a pulsed charge control mode with memory capability so that large arrays can be rapidly switched by a limited number of supplies. In actuator systems, the development of multilayer actuator materials will be traced and new possibilities for ultrahigh-strain materials discussed. The theme will again be electrical polarization control to exploit the large dimension changes that occur in domain and phase change antiferroelectrics and ferroelectrics. In the composite systems, a logical extension of current work is to explore possible systems that incorporate both sensor and responder in the same material coupled together by active solid-state electronics. Systems that seek to control elastic compliance in this manner will be discussed and preliminary measurements of the controlled response presented. Extension of this concept to alternative systems will be considered.
Performance characterization of piezoelectric sensors in smart composite structures
Authors: Hahn H. T. , Kim I. , Granlund E. R.
Published: October 2005

The increased use of fiber-reinforced composites in primary structural components has created a need for effectively monitoring the structural integrity of these composite components. Piezoelectric sensors may provide one means of ensuring composite integrity. This paper examines the results of a research effort to characterize the performance of piezoelectric sensors in smart composite structures subjected to low-velocity impact. A series of low-velocity impact tests was made on composite specimens equipped with attached and embedded piezoelectric sensors. The tests were designed to help establish the relationship of sensor response to impact location and energy for a variety of sensor and composite material configurations. Impact wave characteristics within the composite were studied by theoretical modeling (indentation method, 3-D F.E M., theory of plates), in order to better understand composite behavior at the sensor site. These results combined with sensor material properties (coupling coefficients, etc.) were used to predict sensor response to impact. Last, a comparison was made of experimental and theoretical sensor responses in order to develop impact detection methodology.
Experimental verification of models of piezoelectrically actuated structures
Authors: Cudney H. H.
Published: October 2005

Piezoelectrically controlled structures are currently being investigated for noise control, rotating machinery vibration control, and large space structure control. A model of piezoelectrically controlled structures has been developed that takes into account structures with distributed parameters, Timoshenko deformation assumptions, multiple piezoelectric layers, energy dissipation mechanisms in the structure, and the electrical boundary conditions on the piezoelectric layers [H. H. Cudney, Ph.D. dissertation, SUNY. Buffalo (1989)]. The objective of this research is to verify by experiment the PDE model developed for multiple layers of piezoelectric material attached to a beam. The experiments investigate the accuracy of both shear-lag and ideal models of strain transfer between the structure and the piezoelectric material. The effect of piezoelectric polymer film and piezoelectric ceramics on the damping ratios and natural frequencies of the structure is measured. The effect of varying the electrical boundary conditions on the dynamic response of the structure is demonstrated. The results of using several thin layers of piezoelectric material to control beam vibrations are compared to using only a single thick layer of piezoelectric material. [Work supported by DARPA.]
Representation of pitch input to neural network models of music perception
Authors: Laden Bernice
Published: October 2005

Neural networks are a general class of computational models that can be used to model a variety of music perceptual tasks. An important issue in designing a neural network is the representation of input. The choice of representation can influence the network's trainability, its plausibility as a perceptual model, and its ability to generalize to other musical tasks. Networks that have been trained either to classify musical chords or to identify musical pitch are described. Four approaches to representation are examined. The simplest is a tone-chroma notation in which there are 12 possible input nodes, one for each tone of the Western chromatic scale. Two approaches, a harmonic and subharmonic template, are motivated by theories of complex pitch perception. Input nodes are quantized into pitch-class categories of the Western chromatic scale, and incorporate the notion of pitch height as well as tone chroma. In the fourth approach, each input node represents a frequency bin with a one-third semitone bandwidth. This enables coding of input frequencies that are mistuned with respect to the standard tuning of Western music.
Music composition and model of cortex
Authors: Leng Xiaodan , Shaw Gordon L. , Wright Eric
Published: October 2005

The trion model [PNAS 82, 2364 (1985)] for the firing activity of groups of neurons was developed using the Mountcastle cortical principle of columnar organization; an essential feature is that it is highly structured in time and in spatial connections. The trion model has been shown to be useful for understanding aspects of memory and pattern recognition. To study higher cognitive processes, spatiotemporal firing patterns of time evolutions of the model were mapped onto pitches to produce music. The preliminary results yield the following: (1) The trion music is very pleasant to listen to; (2) one set of connections among the trions produces music sounding like a waltz, and another set gives a minuet; (3) styles of specific periods of composition can be identified. Furthermore. the model demonstrated the ability to learn a musical theme using a Hebbian algorithm. The model provides a window into examining higher brain function.
Asymptotic analysis of nonlinear bounded Gaussian sound beams
Authors: Coulouvrat F.
Published: October 2005

The dimensional analysis of the propagation of large bounded sound beams leads to the nonlinear parabolic Khokhlov-Zabolotskaya equation. It involves a parameter N measuring the magnitude order of the beam widening compared to nonlinear effects. A quasilinear solution of that equation is valid only for low-intensity sound sources (N≫1). In order to obtain explicit solutions valid for more intense sound sources, first, a straightforward expansion in powers of 1/N is sought up to the third order for Gaussian beams of low intensity (N≫1). Then, the secularity appearing in that expansion for smaller N is removed by using the renormalization technique. The main point is the introduction of a phase shift between the linear and nonlinear characteristic variables, which enables us to extend the validity of the asymptotic solution beyond shock for moderate intensities (N>0.5). The computation of the renormalized solution displays the nonlinear attenuation of the fundamental, the widening of the beamwidth, and the asymmetrical distorsion of the wave profile.
Evaluation of transient responses of sound beams using a modified version of the NPE
Authors: Too Gee-Pinn James , Ginsberg Jerry H.
Published: October 2005

The nonlinear progressive wave equation (NPE) computer code developed by McDonald and Kuperman [Comp. Math. Appl. 11, 843–851 (1985)] was used to evaluate a time domain solution in acoustic waveguides. In an earlier work [G. P. Too and J. H. Ginsberg, J. Acoust. Soc. Am. Suppl. 1 86, S63 (1989)], a modification of NPE was described to evaluate the linear steady-state signal generated by a piston in an infinite baffle. Since pulses are usually generated in experiments on sound beams, it is worthwhile to explore the ability of the modified version of NPE to describe transient pulse propagation in the case of a baffled circular piston. In order to evaluate this application, the transient version of the Rayleigh integral [A. D. Pierce, Acoustics, pp. 227–231 (1981)] is used to generate the acoustic field when the pulse is close to the projector. That field is used as the input to NPE, which then propagates the signal. The signal predicted in this manner at long range is then compared to the direct prediction of the Rayleigh integral. [Work supported by ONR, Code 1125-OA.]
Parametric reception near a reflection surface
Authors: Darvennes C. M. , Hamilton M. F. , Tjøtta J. Naze
Published: October 2005

A quasilinear solution of the Khokhlov-Zabolotskaya-Kuznetsov equation is derived for a parametric receiving array (parray) that operates near a reflecting plane (rigid or pressure release). The theoretical model is based on solutions for the sum and difference frequency sound generated by two Gaussian beams that intersect at a small angle. The solutions are combined using the method of images to account for the presence of the reflector. Consideration is given to the parametric measurement of sound radiated by directive underwater sources whose acoustic axes are parallel to an air-water interface. The pump is located on the source and radiates a directive high-intensity beam through the sound field. Cases are shown for which the sum and difference frequency sound detected by a hydrophone on the axis of the pump beam provides reasonable free-field source directivites in the presence of multipath components. Parameter domains (relative radius and frequency of the pump, and distance to the hydrophone) for optimal performance of the parray are discussed. [Work supported by ONR.]
High-efficiency acoustic chamber
Authors: Allen J. L. , Wang T. G.
Published: October 2005

A method for improving the coupling between the sound sources (acoustic drivers) and resonance chamber of an acoustic levitation system has been developed and tested experimentally. By appropriate configuration of the system, a matched driver pair can produce a fourfold increase in the acoustic energy transferred to the chamber as compared to that produced by a single driver. This coupling improvement can be extended even further by adding more driver pairs. The actual improvement of one drive pair as compared to a single driver was determined by measuring sound-pressure levels and acoustic radiation forces on rigid spheres in a rectangular resonance chamber. The dependence of the acoustic pressure and force on sample position and size as related to the coupling improvement will be discussed. Sample rotation and higher harmonics effects will also be discussed. [Work supported by NASA.]
The use of acoustical levitation to estimate air content in coal particles
Authors: Tian Yuren , Apfel Robert E.
Published: October 2005

Since the air content in coal particles is an important parameter for coal utilization, a simple method to estimate it using acoustical levitation is sought. A levitation cell that can easily trap a solid particle at its center is built and used in this study. The corresponding formula for the acoustical force on this kind of particle is also derived. Considering that a coal particle is composed of a solid component and some air, mixture laws for its density and compressibility can be assumed. Based on these relations, the air content in a coal particle can be calculated by measuring acoustical pressure in the levitation cell where it is trapped. Different coal samples are tested. The results show that the ratio of air content in the coal particle varies with its class and size.
Microstructure of the auditory threshold: Reliability, morphology, and relationship to evoked otoacoustic emissions
Authors: Montgomery Allen A. , Cooper William A.
Published: October 2005

The threshold of hearing has been reported to vary as much as 15 dB within a narrow frequency range for normal-hearing subjects. The pattern of such variation across frequency is known as the microstructure of the threshold. In the present study, thresholds were obtained repeatedly from ten trained normal-hearing subjects in the ranges of 800–1000 and 3800–4000 Hz using adaptive threshold techniques. Results indicated that (a) the repeatability of the microstructure varied considerably across subjects, and (b) reliable, large shifts within a range of 100 Hz did not occur. Relationships between the microstructure and spectral/temporal characteristics of the evoked otoacoustic emission were explored. Suggestions for analyzing, smoothing, and quantifying the microstructure of the threshold are made and methods for relating evoked otoacoustic emissions to the auditory threshold are discussed.
Localization ability in noise
Authors: Smith Kim , Wright Diana , Blood Ingrid
Published: October 2005

This investigation examined the accuracy of directional judgment in various signal-to-noise (S/N) ratio conditions. Sixteen subjects were required to localize a tape-recorded stimulus phrase, “where am I,” randomly directed six times to each of seven loudspeakers mounted in a 360-deg array around each listener. The stimulus phrase was presented in quiet and with white noise in 11 S/N ratios ranging from + 12 to − 18. Thus data were collected on a total of 504 presentations per test session. Variability of azimuth and S/N error were computed. Mean localization accuracy decreased and range scores became wider as the conditions became more adverse. These data are discussed with reference to error patterns at each position of the source. Localization repeatability was measured during a second session. No significant differences were found between sessions for any of the S/N conditions.
Hearing protector attenuation performance under different simulated work and fitting conditions
Authors: Park M.-Y. , Casali J. G.
Published: October 2005

Four hearing protectors—a foam-cushion earmuff, foam and triple-flanged polymer earplugs, and an earmuff over foam earplug combination—were evaluated to determine the effects of wearing time, movement activity, and fitting procedure on spectral attenuation. Using a psychophysical real-ear-attenuation-at-threshold measurement procedure (as per ANSI S12.6-1984), attenuation data were collected from 40 naive subjects before, during, and after the activity movement tasks, which induced highly kinematic subject movements. Loss in attenuation over the 2-h wearing period was up to 6 dB for all devices except the foam earplug, and all protectors except the earmuff significantly improved protection (from 4 to 14 dB) at 1000 Hz and below due to trained fitting. The foam plug benefited more than other devices from the training for proper fitting but was resilient to subject movement. [Work supported by NIOSH-NIH.]