bluuebunny/crossref_metadata_2025 huggingface dataset:
Masking Level Differences for Signals Yielding Place and Periodicity Pitch
Authors: Small Arnold M.
Published: October 2005

A question raised repeatedly is whether periodicity pitch is coded through the action of that part of the auditory system ordinarily responsive to low-frequency sinusoids. Release from masking as a function of interaural phase, the masking level difference (MLD), is known to be essentially a low-frequency phenomenon and thus might be used to assess whether high- or low-frequency portions of the auditory system were active in signal processing. This study examines MLDs as a function of noise-masker spectrum level for wide-band noise, 2250 and 225 Hz, and for the 2250-Hz and noise signals pulsed 225 times/sec. The last two conditions are perceived with a pitch approximating that of a sinusoid of 225 Hz. MLDs for signal in phase vs out of phase (N0S0-N0Sπ) were estimated using a two-alternative temporal forced-choice procedure for five subjects in the first and 24 subjects in the second experiment. The results indicate that the pulsed-tone condition, while possessing a low pitch, yields an MLD that approximates that of the high-frequency sinusoid (less than 2.5 dB). The MLDs for the two noise conditions are large (as great as 14 dB) but do not differ.
Detection, Lateralization, and the Phase Angle, α
Authors: Jeffress Lloyd A. , McFadden Dennis
Published: October 2005

The reference output of a phase shifter receiving a 50-Hz band of noise centered at 500 Hz went to two mixers, left and right, and became a continuous, in-phase masker. The variable output went to the mixers via an electronic switch and a step attenuator, and it was used as the signal. A switch allowed selection of an in-phase (N0-S0) or an out-of-phase (N0-Sπ) signal, and the shifter controlled the phase angle α between the masker and the signal. For N0-S0, the signal is an increment when α is zero and a decrement when it is 180° For N0-Sπ, when α is zero, the signal is an interaural difference in level, and when α is 90° the signal generates an interaural phase (time) difference but no interaural difference in level. Psychometric functions were obtained for various values of α using a single-interval, forced-choice procedure—“YES-NO” for detection and “LEFT-RIGHT” for lateralization. Detection for all N0-Sπ conditions was substantially better than for the corresponding N0-S0 conditions. Lateralization shows much shallower functions than detection. At α greater than 90°, the value depending upon the observer, lateralization becomes virtually impossible, while detection remains good.
Lateralizability and Detectability of a Low-Frequency Tone as a Function of the Level of a Tonal Masker
Authors: Yost William A. , Robinson Donald E.
Published: October 2005

Using a two-alternative temporal, forced-choice technique (2ATFC), the detectability of a 250 Hz, 128 msec tonal signal masked by a gated tone of the same frequency and duration was measured as a function of the signal level, the phase angle at which the signal was added to the masker, the masker level, and the interaural phase difference of the signal. When the interaural phase difference is 0° and the signal-masker phase is varied, the detectability of the signal is found to follow a simple energy- or power-detection model. At the two signal-masker phase angles used (0° and 90°), the signal-to-masker level required for 75% correct detection in the 2ATFC task remained constant when the masker level was varied. When the interaural phase difference is 180°, the lateralizability of the signal is found to be independent of the signal-masker phase angle. At signal-masker phase angles of 0° and 90°, the signal-to-masker level required for 75% correct detection in the 2ATFC task remained constant when the masker level was varied. The results are discussed in terms of the Durlach equalization-cancellation model of binaural masking-level differences and in terms of the Weber fraction. [The research was supported in part by a PHS Pre-Doctoral Fellowship.]
Single-Concept Films for Classroom Demonstration
Authors: Beck M. , Strong W.
Published: October 2005

An effort to develop classroom demonstrations for a course, “Descriptive Acoustics of Music and Speech,” is currently in progress. This paper discusses and presents three film sequences that were produced as part of that effort. Computer animation was used to show an ideal string with rigid supports for various initial conditions, different amounts of damping, and so forth. The sequence, “The Superposition Principle as Illustrated by the Ideal String,” shows the decomposition of a complex wave into its simple components and the synthesis of a complex wave from simple components. “The Attenuation of the Vibrations of a Plucked String” illustrates the decay of a complex waveform first with frequency-independent damping and then frequency-dependent damping. “The Resonance of a Driven String” depicts a lightly damped string driven at several different frequencies. The action in the film is slowed by a factor of 800:1 for easy observation of the string motion. A 10-min film will be presented. [Work supported in part by an HEA V-C Fellowship.]
On the Modes and Mechanisms of Perception of Speech Sounds
Authors: Fujisaki Hiroya , Kawashima Takako
Published: October 2005

Previous studies indicated the existence of categorical and continuous modes in the perception of speech sounds. It has been shown, however, that these two modes are not dichotomous, but the degree of categorical response depends largely on various factors such as acoustical properties of stimuli, context, and test procedures [H. Fujisaki and T. Kawashima, Int. Congr. Acoust., 6th, Tokyo (1968), paper B-3-6]. A method is described for the precise measurement and quantitative expression of a subject's ability of discriminating between two speech sounds. The perception of various categories of speech sounds including vowels, semivowels, liquids, voiced stops, and fricatives is studied experimentally and analyzed using stimuli compiled by a digital computer. A mathematical model of the process of speech perception is then presented that relates discrimination to identification, taking into account the reliability of the short-time memories involved in the task of discrimination. The latter is shown to be affected by various stimulus attributes and test conditions, which explains the variability in the degree of categorical response to various speech sounds.
Silent Interval Duration as a Perceptual Cue of Speech Pauses
Authors: Ruder Kenneth F. , Jensen Paul J.
Published: October 2005

The effect of silent-interval duration on the perception of pauses in speech was investigated. The stimuli consisted of five recorded sentences within which the words “lost” and “contact” were manipulated so that their syntactic relation to one another varied in complexity from sentence to sentence. Twelve subjects, working individually, mechanically adjusted the silent-interval duration between the words “lost” and “contact” within each sentence in order to locate (1) the pause detection threshold, (2) optimal fluent pause duration, and (3) minimal hesitation pause duration. Across the five sentences, the mean durations were 6, 186, and 505 msec, for the three pause types, respectively. Statistical analysis, however, showed that silent interval durations for these three pause types differed significantly only when occurring between phrases. This was not true of within-phrase pauses. Thus, contrary to usual assumptions in the literature, these results suggest that for within-phrase pauses, at least, silent-interval duration is an insufficient perceptual cue for differentiation of fluent and hesitation pauses. [Supported by NSF and NIH grants.]
Effect of Vowel Duration upon the Perception of the Voicing Characteristic of Word-Final Consonants
Authors: Raphael Lawrence J.
Published: October 2005

Previous studies by Denes, Sholes, and others have suggested that the duration of a vowel is a significant cue to the voicing characteristic of the consonant that follows it. The present study investigated the effect of preceding vowel duration upon the perception of word-final stops, fricatives, and clusters in synthetic speech. A variety of minimal CVC(C) pairs was synthesized and the vowel of each was varied over a range of values derived from those found in real-speech samples. It was found that, regardless of the cues for voicing or voicelessness used in the synthesis of the final consonant or cluster, listeners perceived the final segments as voiceless when they were preceded by vowels of short duration and as voiced when they were preceded by vowels of long duration. Discrimination tests revealed no peaks at the perceptual phoneme boundaries, indicating that when the voicing characteristic is cued by vowel duration, perception is continuous rather than categorical.
Soluble Problems in Scattering from Rough Surfaces
Authors: DeSanto John A.
Published: October 2005

The problem of scattering a scalar plane wave from some periodic, rectangular, one-dimensionally corrugated surfaces is solved exactly by using residue calculus techniques from complex function theory. There are no parameter restrictions in the treatment. Solutions of the Helmholtz equation are assumed to be a superposition of upgoing plane waves above the corrugation wells and standing waves within the wells. Both have unknown amplitude coefficients. The solutions and their normal derivatives are matched across the common boundary yielding two infinite sets of linear equations for the amplitudes. Meromorphic functions are constructed whose integrals in the complex plane yield residue series, which match the sets of linear equations. The amplitude coefficients are given by either specific values or residues of the known functions. The velocity potential is shown to satisfy the usual edge conditions. The chief advantage of the method is that matrix inversion is unnecessary. One example is done in detail, that of infinitesimally thin, semiinfinite depth corrugations, the Carlson-Heins problem. Other soluble examples and previous work are discussed. Since the residue calculus technique and the Wiener-Hopf method are related, an extension of the class of soluble Wiener-Hopf type problems is presented.
Direct Observation of a Phonon Bottleneck in Nickel-Doped Magnesium Oxide Using Brillouin Light Scattering
Authors: Brya W. J. , Geschwind S. , Devlin G. E.
Published: October 2005

The technique of Brillouin scattering has been used to observe directly a microwave phonon bottleneck produced by relaxing Ni2+ ions in MgO at 2°K. Phonons at 25 kMHz on the lowest acoustic branch traveling close to a [110] direction were monitored with Brillouin scattering. When the Ni2+ spin system, in an external field of approximately 8000 G, is saturated with resonant microwaves at 25 kMHz, these phonons are heated to several hundred degrees while the lattice temperature remains at 2°K. This phonon heating is confined to a frequency bandwidth of less than 200 MHz, while the inhomogeneous spin-resonance linewidth is approximately 500 MHz. In addition to the 25-kMHz phonons generated by the Δm = −1 transitions, 50-kMHz phonons corresponding to the Δm = 2 transitions were also observed. The actual heating is found to be spatially nonuniform throughout the crystal and in addition under certain conditions the Stokes line is found to be orders of magnitude greater than the anti-Stokes.
Approaches to the Characterization of Talker Differences by Statistical Operations on Speech Spectra
Authors: Li K.-P. , Hughes G. W. , House A. S.
Published: October 2005

As part of a general study of the statistical characteristics of speech spectra, a number of characteristics relevant to talker identification are being investigated. The use of some statistical measures for this purpose has been described, in which mean values of filter outputs [P. B. Denes and M. V. Mathews, J. Acoust. Soc. Amer. 32, 4150 (1960)] and statistical manipulations of spectral features [S. Pruzansky, M. V. Mathews, and P. B. Britner, J. Acoust. Soc. Amer. 35, 1877 (1963)] were applied to word and talker recognition. Extensions of these methods are possible, such as evaluation of the temporal stability of the correlation matrix; characterization of patterns in the correlation matrix, particularly in the form of isocorrelation contours [K-P. Li, G. W. Hughes, and A. S. House, J. Acoust. Soc. Amer. (to be published)]; and extraction of eigenvector correlations and patterns. Results of some preliminary experiments utilizing these techniques are described and potential applications to speech-analyzing and talker-identifying schemes are discussed. [This work was supported in part by the Air Force Cambridge Research Laboratories, Office of Aerospace Research.]
A Modern Approach to Sound Spectrograph Instrumentation
Authors: Presti A. J.
Published: October 2005

A new sound spectrograph has been developed that features direct analysis from a prerecorded tape plus several new features that facilitate ease of operation and provide a higher degree of reliability and consistency. The design uses a tape scanner that repeats any 2.5-sec segment of tape for spectrographic analysis, and includes an automatic tape-advance mode for producing contiguous spectrograms. The basic model features only a single type of display—namely, the wide-band “bar” display for applications in areas where visual pattern matching and speech therapy are desirable. The typical wide-band analyzing-filter bandwidth of 300 Hz has been found to be ideally suited where visual comparisons of speech sounds are desired. This display is also useful for pattern studies of heart sounds, mechanical and industrial sounds, underwater sounds, etc. One of the several new design improvements to be described is the ability to mark single-line frequency calibration marks across the entire spectrogram. All of the features presently included in the standard spectrograph, namely the choice of filter bandwidth, frequency shaping, contour display, amplitude cross sectioner, and frequency scale expansion, can be added depending on the requirements of the user.
Data Processing for Analyzing Intonational Signals of American English
Authors: Takefuta Y. , Morehead D. M.
Published: October 2005

Eleven different methods for processing the measures of F0 on the melody curves of American English were evaluated with regard to their relative efficiencies in detecting intonational signals. The processing methods to be evaluated were chosen on the basis of their ability to analyze the measures of F0 and reduce them to a single digit to be related with any one of such psychological sensations as pitch pattern, pitch level, rate of pitch change, range of pitch change, and duration of pitch. A method of combining a linear regression analysis with an appropriate segmentation was introduced in this study to analyze a complex melody pattern into a set of simple linear patterns. Relative efficiencies of all the processing methods were evaluated by computing the classical indexes of signal detectability. The method introduced to quantize the melody patterns of utterances was the most efficient of all the processing methods tested for detecting the intonational signals. [This research was supported by a contract between The Ohio State University Research Foundation and the Office of Naval Research.]
Real-Time Random-Signal Processing
Authors: Keller A. C.
Published: October 2005

Signal-processing techniques and a commercially available system are described for making automatic measurements of spectral density or normalized spectrum level of stochastic signals. The system described employs a hybrid analog-and-digital technique to present spectrum displays on line in optimally minimum time. Digital time-compression techniques are used in conjunction with a swept heterodyne spectrum analyzer. A 1500-work data sample varying in length from 25 msec to 50 sec is processed continuously. Up to 4096 such samples may be combined for purposes of examining long-term average properties. Averaging times selectable from 25 msec to 200 000 sec are furnished with the analysis system. A fixed-integration-time technique is used employing nine-bit digital samples that are averaged and stored in 16-bit registers. A typical dynamic range of analysis without switching of 54 dB is furnished, with a frequency range of operation from 0.03 Hz to at least 1 MHz. The parameters available from the analysis system include analog voltages proportional to frequency and Fourier content in both linear and logarithmic form. The class of signals best handled by the analysis equipment includes periodic, complex periodic, transient, and stationary random waveforms. The analysis speed of the system lends itself to the investigation of the stationarity of random signals. Concepts such as automatic data normalization, signal analysis of machinery, and digital computer interface are also presented.
Stored-Energy Retrieval in Ultrasonic Image Converter
Authors: Render James M.
Published: October 2005

The electron beam employed to scan the conversion plate of an ultrasonic image converter serves as a switching device, which momentarily connects individual elements of the plate to the termination. If sufficient time has lapsed since their previous interrogation, these elements possess a considerable amount of stored energy due to resonant vibration. With continuous-wave illumination of uniform intensity, and with constant sweep velocity, the energy flowing to the termination consists of a constant-amplitude component of the same frequency as the driving source, and transient components due to switching from one element to the next. The constant-amplitude component may be associated with the impinging ultrasonic signal and the others associated with release of energy stored in the plate. Calculations show that the signal due to release of stored energy increases with beam sweep velocity, and may exceed the amplitude of the steady-state signal obtained by holding the beam in a stationary position.
Upper and Lower Bounds on the Nth Eigenvalue of Vibrating Anisotropic Solids
Authors: Hertelendy Paul
Published: October 2005

Within the context of linear theory of elasticity, it is shown that for any arbitrary anisotropic material, having as many as 21 independent elastic constants, one can compute lower-bound and upper-bound isotropic moduli that consistently yield lower and higher eigenvalues, respectively, than those of the anisotropic material. Computations are carried out for a variety of common cubic, hexagonal, tetragonal, trigonal, and orthorhombic crystalline materials.
Studies of Resolution in a Bragg Imaging System
Authors: Wade Glen , Smith Roy A. , Landry C. John , Powers John P.
Published: October 2005

This paper presents the results of experiments that were performed to investigate the resolution capabilities of a Bragg imaging system using a cylindrically convergent light beam. The usual images formed with such a system show up as silhouetts of the objects being imaged. These silhouettes are superimposed on a light background, which constitutes the visual representation of the incident sound field. The Rayleigh criterion governs the resolution for thin cylindrical objects oriented parallel to the direction of the convergence line of the light beam. For orientation at right angles to this direction, the resolution is of quite a different character. The experimental resolution capabilities for both directions are presented in this paper and are compared with theoretical predictions. Resolution data is also presented for the case of dark-field imaging, i.e., where the objects appear as bright images on a dark background. In addition, experiments were performed to determine the loss of resolution resulting from the introduction of optically opaque plates between the scattering object and the light-sound interaction region.
Performance-Intensity Functions for Continuous Discourse
Authors: Speaks Charles , Parker Barbara
Published: October 2005

Previous experiments on the intelligibility of continuous discourse employed a speech-Békésy procedure to define subjective thresholds of intelligibility (TI) and detectability (TD). Such procedures produced data grossly representative of two points on an underlying performance-intensity (P-I) function. No validating criterion existed, however, to permit specification of a percent-intelligibility score for either TI or TD. Two experiments were conducted. In the first, listeners tracked to TI, TD, and to three constant-percentage criteria of 75%, 50%, and 25% for noise-masked continuous discourse. The data yielded a P-I function rising at a rate of 12%/dB. Standard deviations across 10 replications ranged from 0.8 to 1.3 dB. In the second experiment, the listeners judged the intelligibility of 15-sec samples of continuous discourse. This procedure produced a comparably steep P-I function of 11%/dB. TI, the “lowest level at which the meaning of almost every sentence and phrase could be understood” corresponded to an estimated intelligibility of approximately 80%. Stability of these measures across noise levels ranging from 50 to 80 dB SPL are described.
Voiceless-Versus-Voiced Consonant-Vowel Perception in Dichotic and Monotic Listening
Authors: Berlin C. I. , Willet M. E. , Thompson C. L. , Cullen J. K. , Lowe S. S.
Published: October 2005

We had previously reported [J. Acoust. Soc. Amer. 45, 299 (A) (1969)] that in dichotic listening to consonant-vowel (CV) utterances in natural speech, more voiceless consonants were correctly perceived than voiced consonants. In that experiment, the voiceless CVs had a slightly higher fundamental frequency than the voiced; therefore, synthetic CVs with uniform fundamental frequency and duration were used in the present experiment. Twenty normal right-handed females listened to simultaneous (±212 msec) (stop+/a/) synthetic nonsense syllables both monotically and dichotically. In addition to the expected right-ear laterality effect in dichotic listening, we confirmed our previous finding: dichotically, voiceless consonants predominated (73% vs 48%). Monotically, voiced consonants were most often heard correctly (60% vs 47%). An explanation related to onset of change from aperiodic to periodic portions of voiceless vs voiced utterances is presented. [We are grateful to Arthur Abramson and Lee Lisker for the use of theiry synthetic speech material. Supported in part by NINDS.]
Response of One Type of Neuron in the Inferior Colliculus of the Kangaroo Rat (Dipodomys spectabilis) to Low-Frequency Binaural Tones
Authors: Stillman Robert D.
Published: October 2005

About 75% of the low-frequency binaural neurons encountered exhibit PST histogram distributions to binaural tonal bleeps, which characteristically show an early response followed by a quiescent period and then later firing. All components of the PST response occur approximately within the temporal confines of the stimulus. Such neurons cannot be adequately described solely in terms of response rate. An analysis of the PST histograms often reveals the presence of complex facilitatory and inhibitory interactions. These interactions are usually acutely sensitive to interaural time and intensive differences and to diotic changes of over-all level and frequency. All three components of the PST response vary simultaneously with some stimulus configurations, while other stimulus changes may selectively alter one or two of the components. This raises the possibility that such neurons convey information concerning several aspects of the acoustic stimulus. [Research supported by NINDS Postdocotral Fellowship and NIH Grant to George Moushegian.]
Sensitivity of Single Neurons in the Dorsal Nucleus of the Lateral Lemniscus of the Cat to Binaural Tonal Stimulation
Authors: Brugge J. F. , Anderson D. J. , Aitkin L. M.
Published: October 2005

An ipsilateral stimulus, when paired with an effective contralateral stimulus of the same frequency, can affect the discharges of cells in the dorsal nucleus; this response depends upon interaural delay time and intensity. For some neurons excited by tones below about 2500 Hz, the spike count is a periodic function of interaural delay, the period being equal to the period of the tones. Below 1000 Hz, and at interaural delays favorable for activating the neuron, spikes may occur at intervals corresponding to integral multiples of the period of the tones. Units with higher best frequencies are usually excited by contralateral ear stimulation, and the resultant discharge may be suppressed by ipsilateral ear stimulation. Maximum suppression is usually achieved when the interaural difference in sound-pressure level is about 20–30 dB. When a contralateral best-frequency tone is paired with ipsilateral tones at various frequencies and intensities the resultant response area overlaps, but need not be coextensive with, the contralateral response area.
Binaural Interaction in the Dorsal Cochlear Nucleus
Authors: Mast Truman E.
Published: October 2005

Anesthetized chinchillas were used to study the effects of contralateral tonal stimulation on spontaneous and evoked activity of single neurons of the dorsal cochlear nucleus (DCN). Extracellular recording was done by use of steel microelectrodes. Most of the neurons studied had best frequencies above 3000 Hz. Our results follow: (1) contralateral tones inhibited ipsilaterally evoked activity and also spontaneous activity (if present) in roughly 20% of neurons of the DCN; (2) the best frequency for contralateral inhibition was usually similar to the best frequency for ipsilateral excitation; (3) the latencies for contralateral inhibition were similar to or only a few milliseconds greater than the latencies for ipsilateral excitation; (4) neurons showing contralateral inhibition were localized to the middle layers of the DCN; (5) neurons showing contralateral inhibition were sensitive to small interaural differences in stimulus intensity. These results indicate that the DCN may play a significant rôle in localization of sound. The anatomical pathways mediating the contralateral inhibition are not known but probably are associated with the efferent system. [Supported by NINDB, National Institutes of Health.]
The Open-Plan Office, Concepts and Attributes
Authors: Propst R. L.
Published: October 2005

In the last ten years, new information from the behavioral sciences, coupled to pressures caused by rapid change and growth, have been utilized to conceptualize office facilities with essentially open interiors. This thinking attempts to reconcile a wide variety of environmental and organizational objectives and has demonstrated, in a variety of installations, a major shift in viewpoints of the purpose and function of offices. Communication proficiency is now assessed as a prime goal. Research and analysis are showing that the actual communication structure of an organization can be determined and that it can be improved and upgraded in performance by using open and arrangeable facilities designed to support a communications strategy. It is also noted that the open environments tend to cause a significant behavioral shift, moving toward more moderate and socially responsive behavior. The cultural climate is observed to vary considerably from organization to organization and country to country. The high cost of modifying fixed wall enclosures to meet rapid growth and change requirements has been documented, showing very significant advantages to adjustable open plan space.
Designs for Open-Plan and Conventional Offices
Authors: Rodgers Sydney G.
Published: October 2005

Interior design views concerning open-plan and conventional offices will be presented. Planning concepts, function, appearance, materials, and costs will be discussed and contrasted for various office types. Experiences with clients' reactions to various office environments will be reviewed.
Acoustical Experiences to Date with Open-Plan Schools
Authors: King Jonathan
Published: October 2005

A summary of experiences of Educational Facilities Laboratories with open-plan schools will be presented. In general, fewer acoustical problems have been encountered in practice than were anticipated at the outset of widespread construction of this type. However, an acceptable acoustic environment in an open-plan school is related to certain fundamental planning principles, which will be discussed. In addition to physical factors, there are psychological factors that affect acceptability of the acoustic environment, and these must not be overlooked.
High-Pressure Properties of Acoustic Decoupling Materials
Authors: Higgs R. W. , Eriksson L. J.
Published: October 2005

The sound velocity and attenuation of Corprene and onionskin paper were measured as a function of pressure, temperature, and frequency using an impedance tube technique. In these samples, the properties change with the penetration of water; therefore, they were formed as composites with silicone rubber. Knowing the sound velocity in the rubber and using modeling techniques for composite materials, the sound velocity for the sample material can be determined from the sound velocity of the composite. The data and modeling techniques are described. The sound velocity is shown to increase nonlinearly from about 500–1400 m sec−1 for Corprene and from about 500–900 m sec−1 for onionskin paper as the pressure is increased from about 0–2000 psi. Both materials are shown to have absorption coefficients that decrease nonlinearly from approximately 4 dB/cm at 0 psi to less than 1 dB/cm at 2000 psi.
High-Pressure Degradation Of Performance by Transducers Using Onionskin Paper for Acoustic Decoupling
Authors: Haan D. E. , Higgs R. W. , Eriksson L. J.
Published: October 2005

Acoustic decoupling behavior of onionskin paper is examined with emphasis on sonar-transducer applications. Effects of changes in the acoustic impedance of a decoupling support structure are discussed in detail. Nonlinear changes in sound velocity with increased uniaxial pressure are shown to degrade both frequency and directional response of the transducer. Predicted characteristics obtained by computer analysis of a longitudinal-vibrator equivalent circuit are compared to measured characteristics of a specific transducer configuration.
Large-Amplitude Pulse Propagation: A Transient Effect
Authors: Moffett M. B. , Westervelt P. J. , Beyer R. T.
Published: October 2005

An experimental demonstration has been made of transient-signal generation during the propagation of a large amplitude rf pulse in a liquid. An electrical pulse with a carrier frequency of 10 MHz is fed to a quartz transducer. The rf acoustic pulse thereby generated (pressure amplitude about 1.9 atm) was allowed to propagate through a tank containing CCl4 to a wide-band acoustic receiver. As the path length in the liquid was increased, transients appear at the front and rear of the pulse, in accord with the theoretical treatment of Berktay [J. Sound Vib. 2, 435–461 (1963)]. [Supported in part by Office of Naval Research.]
Calculation of Classification Functions Using a Large Number of Spoken Word Samples
Authors: Chiba Seibi , Sakoe Hiroaki
Published: October 2005

A two-phase procedure for calculation of multiple linear classification functions has been developed. Phase I approximates the distribution of samples in the measurement space by a series of linear functions and preselects only those samples lying in the critical region. Phase II utilizes the reduced number of samples for calculation by a linear programming algorithm, the multiple linear classification functions that can separate given samples strictly. The procedure was applied to the problem of spoken-word recognition. About 3300 samples of Japanese spoken digits, uttered by 55 male speakers and represented as 56-dimensional measurement vectors by a vocoder-type frequency analyzer are collected. During Phase I, the number was reduced to 16 linear functions on the average. Precise classification functions were then calculated for each digit in Phase II. Most of the digits were separated by a single linear function while others were separated by two functions. An error rate of less than 0.2% was achieved for 550 samples newly uttered by the same 55 speakers.
Control Rule of the Tongue Movement for Dynamic Analog Speech Synthesis
Authors: Hiki Shizuo
Published: October 2005

As part of a study of the controls rules of speech synthesis with the physiological parameters, the control rule of the tongue movement is discussed. Effect of the movement of each of the several muscles involved in the tongue on the change in the three-dimensional shape of the tongue is analyzed based on the anatomical data, and the neuromuscular commands of these muscles are estimated utilizing the EMG data. The result is combined with the control rules of the jaw and lip movements to calculate area function of the vocal tract with a digital computer and to drive a dynamic analog vocal tract circuit. [This research is partially supported by the National Institutes of Health, U. S. Department of Health, Education, and Welfare, and the Office of Naval Research.]
Combining the Information from Two Temporally Separated Observations in a Two-Alternative Forced-Choice Signal-Detection Task
Authors: Zagorski Michael
Published: October 2005

The signal (a shift in frequency, intensity, or both, of a sinusoidal pedestal masked by wide-band noise) was made unavailable in the first or the second observation interval to determine the relative weight given to each by the subject in arriving at an over-all decision as to which trial type was presented. The data seem to be consistent with a process in which the subject decides whether or not to listen to the second interval before reaching the final decision on a given trial.
Receiver Operating Characteristics for Detection by Systems with Decision-Making Relay Stations
Authors: Egan James P.
Published: October 2005

Assume that the performance of an observer is constrained by receiver operating characteristic (ROC) with (d′)1. The observer responds with YES or NO according to some criterion. Also assume that a signal is sent to a second observer when the first observer responds with YES. If NO, the signal is not sent in that observation interval. The ROC for the second observer is specified by (d′)2. Thus, a signal is relayed to a second observer by way of an intermediary who thereby introduces a threshold. An ROC for the over-all system shows the relation between the probability that the second observer responds YES, given that a signal was sent to the first observer, and the probability that the second observer responds YES, given that the signal was not sent to the first observer. The ROC for the over-all system may be generated by independently or jointly varying the criteria of the observers. ROCs are a function of (1) values of (d′)i; (2) criterion variations by the observers; (3) number of observers. [Supported by Air Force Office of Scientific Research.]
Calculation of the Perceived Level in PLdB
Authors: Stevens S. S.
Published: October 2005

Modest improvements now appear possible in the procedure used for the calculation of loudness or noisiness. Young and Peterson [J. Acoust. Soc. Amer. 45, 834–838 (1969] have reexamined the precision of the two procedures, Mark VI loudness level and perceived noise level (PNL), which are both based on the same loudness summation rule [S. S. Stevens, J. Acoust. Soc. Amer. 28, 807–832 (1956)]. Mark VI predicted judgments of noisiness with slightly greater precision than PNL. That finding has led to a review of procedures for noise evaluation. Experiments with bands of noise over the past 25 years have provided some 24 contours of equal loudness, equal annoyance, equal acceptability, etc., from which a composite frequency-weighting function has been derived. This new weighting contour may replace the somewhat oversimplified loudness-index contour of Mark VI and the somewhat exaggerated contour used to calculate PNL in PNdB. The use of a common weighting contour for both loudness and noisiness suggests the possibility of a revised procedure (Mark VII) with which to calculate what may be called Perceived Level in PLdB.
Subharmonics of the Bubble Equation Addenda
Authors: Solomon L. P. , Evans L. L.
Published: October 2005

A previous paper by the authors [J. Acoust. Soc. Amer. 44, 1121 (A) (1968)] presented evidence obtained with the hybrid analog computer that tended to confirm the earlier approximate analytical results obtained by Eller and Flynn on bubble subharmonics of order one-half. This paper presents investigations into the general possibility of occurrence of subharmonics of the form (m/n) ωr, where m and n are integers, m<n, and ωr is the linear resonance frequency of the bubble. The presentation includes discussion of the numerical and analog techniques, which should be generally applicable to the investigation of the subharmonic content of any nonlinear differential equation.
Wavenumber-Frequency Spectra of Turbulent Boundary Layer Pressure Measured by Microphone Arrays
Authors: Blake William K. , Chase David M.
Published: October 2005

Measurements of frequency spectra of smooth-wall pressures have been made by single microphones and by a longitudinal array of four flush 0.8-in. circular microphones connected with alternating and with common phase. The microphones were selected to suppress high wavenumbers. The measured levels set upper limits on low-wavenumber boundary-layer pressure. The array was designed to suppress, by its wavenumber filtering, the background acoustic duct noise at frequencies near 3 kHz. Analysis indicates that a very high phase-speed low wavenumber pressure field dominates near this frequency. At low frequencies, single-microphone and array spectra are dominated by the convective wavenumber component of boundary-layer pressure and satisfactory agreement is found with theoretical predictions. [This work was supported by the Acoustics Branch of the Office of Naval Research.]
Measurement of the Convection Velocity in Spectral Space
Authors: Brackenridge J. B. , Geib F. E. , Maidanik G. , Remmers G.
Published: October 2005

A convecting pressure field on a boundary is modeled analytically and the convection velocity is defined. The convection velocity of the pressure field on a boundary induced by a turbulent boundary layer is then related a priori to the convection velocity of the model pressure field. In this manner, a convection velocity is defined in real space—{x,t} space, and in spectral space—{k,ω} space. The spatial vector variable x is the difference between two position vectors on the boundary and the temporal variable t is the difference between two instants of time. The wave-vector variable k is the Fourier conjugate of x and the frequency variable ω is the Fourier conjugate of t. Techniques to measure the convection velocity in each space are discussed and compared. Experimental data of the convection velocity of the pressure field on a boundary induced by a turbulent boundary layer and measured with spectral filters are reported. The measurements were carried out in a 12-in. water tunnel.
Synthesis of Stop Consonants in Initial Position
Authors: Klatt D. H.
Published: October 2005

A complete set of rules is described for the production of English stop consonants in initial position with a terminal analog speech synthesizer. The five-pole series-connected synthesizer is simulated on a medium-sized general purpose computer, and has several novel features. Frication spectra are generated by controlling a five-pole fricative tract so that frication poles are continuous with the poles of the segment that follows, but several of these poles may be effectively canceled by zeros depending on the place of articulation. An aspiration noise source having a flat spectrum is used to produce a good approximation to natural aspiration when the noise is inserted in place of the voicing source. Rules to be described govern the characteristics of (1) a short duration of frication at the release of a stop; (2) aspiration in a voiceless stop; and (3) formant transitions into the following segment. These rules have been used to generate a randomized list of 90 CV nonsense syllables consisting of the stops /p,b,t,d,k,g/ followed by 15 English vowels. A panel of naïve listeners can identify more than 95% of these consonants. [This work was supported in part by the National Institutes of Health and in part by the U. S. Air Force Cambridge Research Laboratories, Office of Aerospace Research.]
Absorption of Sound in Air
Authors: Piercy J. E.
Published: October 2005

Recent analyses of both laboratory and field measurements of the absorption of sound in moist air have indicated an absorption at low frequencies considerably in excess of that predicted for the vibrational relaxation in oxygen and classical processes. The contribution from the vibrational relaxation in nitrogen, tradionally neglected, has now been calculated by use of the known physics of vibrational relaxation in mixtures of gases and molecular constants. It is found to have the right magnitude, dependence on frequency, and dependence on humidity to eliminate effectively the discrepancy with laboratory measurements, but to indicate for lower frequencies (<100 Hz) an absorption much lower than field measurements and recent empirical formulas of Harris and Sutherland. It appears, however, that for relative humidities >25% and temperatures >10°C, in which most people live, the dominant mechanism of sound absorption in air over a major portion of the audible frequency range is the vibrational relaxation in nitrogen, not oxygen as supposed hitherto.
Reflective Surfaces for Reinforcing Unamplified Speech in a Lecture Room and a Thrust-Stage Theater
Authors: Knudsen V. O. , Walker B. E. , Delsasso L. P.
Published: October 2005

Speech-articulation tests in a lecture room, designed to provide optimal reflections of sound for the lecturer, reveal that the articulation index is practically independent of the direction the speaker is facing. Even for a listener with normal hearing in a rear-side seat, the number of typical test words heard correctly is greater than 90% for speaker orientations of 0°, 90°, and 180°. The average percentages for five listeners, two of whom (V.O.K. and L.P.D.} had typical high-frequency hearing losses, were 90%, 87%, and 92%, respectively, for the speaker orientations of 0°, 90°, and 180°. The acoustical features of the 108-seat lecture room is described. A tape recording demonstrates how typical listeners hear a lecturer who first faces the listeners and then turns away 90° and then 180°. Tests and speech recordings in a thrust-stage theater at Washburn University, designed to optimize the benefit of reflective surfaces, which is described, show that a listener in the least-favorable seat hears a speaker on stage almost equally well irrespective of the direction the speaker is facing. Tape recordings and sound level records compare these directional effects on the hearing of speech in the theater with those obtained in the anechoic room.
Experimental Analysis of the Frequency Response of Air-Filled Targets in Water
Authors: Shirley D. J.
Published: October 2005

Steady-state frequency response functions for thin-wall air-filled targets over the nondimensional frequency (ka) range, 30 ≦ ka ≦ 62, are shown. Both spherical and cylindrical targets were used. Similarities between the frequency response functions for the two target forms are related to similarities in their acoustic geometry. Both curves compare favorably with a theoretical curve for the sphere computed by Hickling. Evidence is presented that the frequency response results from interference between the specular return and a secondary return produced by circumferential waves propagating in the metal shells of the targets. Two velocity modes of these waves are observed. [This work was supported by the Office of Naval Research.]
Experimental Tests of an Underwater Solid Acoustic Lens
Authors: Katzenmeyer W. E.
Published: October 2005

Experimental results are given for tests of a 15-in. diam plexiglas acoustic lens. Using effectively a point source at a distance of 8 ft from the lens, the focused sound field behind the lens was scanned with a detector at various frequencies (from 250 KHz to 1 MHz) with the source both on-axis and off-axis. The lens performance is shown to be diffraction limited up to a frequency of about 500 kHz corresponding to a beamwidth (at −3dB) of 12°. The results agree with predictions from a computer lens design program. Calculations are made showing the efficiency of the lens as a function of frequency. By using vertical pipes as a target, the best focal surface and modulation transfer functions are determined.
Effects of Frequency, Intensity, and Vowel-Type Variations on Maximum Phonation Time
Authors: Lass Norman J. , Michel John F.
Published: October 2005

Forty adult subjects, 20 males and 20 females, performed 14 maximum phonation time tasks: sustained phonations of /a/ using soft, moderate, and loud intensity levels at the 25th, 50th, and 75th percentile points in the subject's pitch range; sustained phonations of /i/, /u/, and /æ/ using a moderate intensity level at the 50th percentile point in the subject's pitch range; and two uncontrolled phonations. It was found that (1) for low-frequency phonations, subjects showed a progressive increase in phonation time with an increase in intensity level; (2) for high-frequency phonations, subjects exhibited a progressive decrease in phonation time with an increase in intensity level; (3) for moderate-frequency phonations, the male group showed a progressive increase in phonation time with an increase in intensity level, while the female group exhibited a greater phonation time for the soft than for the loud intensity level conditions; and (4) subjects exhibited the longest phonation time for /i/ and the shortest time for /æ/. These results indicate that frequency, intensity, and vowel-type variations should be taken into consideration in all tasks involving maximum duration of phonation. [Research supported by the National Institute of Neurological Diseases and Blindness.]
Effects of Echo-Free Transmission Delay on Speech Patterns in Two-Way Telephone Conversations
Authors: Brady Paul T.
Published: October 2005

Satellite communication systems typically introduce round-trip transmission delays of at least 600 msec in telephone calls. Previous studies [e.g., E. T. Klemmer, Bell System Tech. J. 46, 1141–1147 (1967)] have sought subjective opinions of the acceptability of circuits containing long delays and no echo. In the present study, 16 pairs of male subjects conversed over an echo-free telephone circuit on zero delay and 600-msec round-trip delay, and 16 additional pairs conversed on 0- and 1200-msec delay. Each condition lasted 10 min. Subjects were selected to have minimal experience on delay circuits and were successfully led to believe they were conversing at all times over a standard circuit. This experiment examined differences in conversational behavior on the standard and delay circuits in the following aspects: (1) durations of “events” in ON-OFF patterns (talkspurts, mutual silences, etc.), (2) changes of pattern-generating model parameters [P. T. Brady, Bell System Tech. J. 48. 2445–2472 (1969)], (3) speaking levels, (4) instances of “confusion” observed by experimenter, and (5) reaction of subjects. Some model parameters change with delay, causing predictable changes in durations of certain events (double talks and mutual silences increase). Confusions are more numerous on delay. Speaking levels and subjects' opinions are unaffected by delay in their specialized task.