bluuebunny/crossref_metadata_2025 huggingface dataset:
Oral and Direct Subglottal Pressure in Korean Stops
Authors: Lee Charles Y. , Smith Timothy S.
Published: October 2005

Recordings of the intraoral air pressure (Po) and direct subglottal air pressure (Ps) were made of one native speaker pronouncing the Korean aspirated, tense, and lax stops. Labial, dental, and velar stops in the frame /i-kəs-i + Vl+i-ta/ (“This is _______.”), where the vowel (V) was /i, a, u/, were investigated. It was found that systematic differences in the Po were not reflected in the Ps. There was little difference between the tense and aspirated stops in terms of Po; however, the aspirated stops showed considerably higher Ps. In addition, it was found that there was no systematic difference in Ps between the tense and lax stops. These findings support the claim of Chomsky and Halle [The Sound Pattern of English (Harper and Row, New York, 1968), p. 326] that the Korean aspirated stops have heightened subglottal pressure; however they do not support their claim that the tense stops also have heightened subglottal pressure. The relation of these data to Chin-W. Kim's [“A theory of aspiration,” Phonetica 21, 107 (1970)] findings concerning glottal constriction in Korean stops will also be discussed.
Identification of Stops and Vowels for the Burst Portion of /p t k/
Authors: Scheir M. E. , Winitz H.
Published: October 2005

The stops /p t k/ were tested for identification in the context of /i a u/. The burst portions of initial and final stops were isolated from conversational speech through electronic gating circuitry. Subjects were instructed to identify either stop or stop and vowel (given binary or trixsty choices). The /t/ stop was most often accurately identified. Coarticulation effects were apparent in that a high proportion of the vowels could be identified on the burst portion.
Variations in Cochlear Microphonics (CM) and Eighth Nerve Action Potentials (AP) with Age in Rats
Authors: Crowley D. E. , Schramm V. L. , Swain R. E.
Published: October 2005

CM responses to pure tones, and whole-nerve AP responses to clicks with and without noise masking, were measured acutely from the round window in groups of male Sprague-Dawley rats at ages of 1, 6, 12, 18, and 24 months. According to average CM measures made from 0.5 to 100.0 kHz, the greatest sensitivity occurred in the 1-yr group, with the younger and older animals (up to 18 months) showing comparative losses up to 18 dB at 1.0 kHz. Comparative losses for the 2-yr group ranged from 26 dB at 1.0 kHz to 36 dB at 30.0 kHz. Mean N1 peak amplitudes varied similarly with the highest values found in the 1-yr group and the lowest values in the 1-month rats. When compared with the 1-yr rats, older and younger rats showed a reduction in the slope of the N1 amplitude versus click level curve. Mean N1 latency did not vary with age. Derived masked AP amplitudes [Teas, Eldredge, and Davis, J. Acoust. Soc. Amer. 34, 1438–1459 (1962)] varied with derived masking band center frequency (c.f.) in an age dependent manner. Below c.f. = 24.0 kHz the descending rank order of amplitudes was 12, 6, 18, 24, and 1 month. As with the unmasked AP, the derived masked AP latency did not vary with age. According to these measures, peripheral auditory maturity is not reached in the rat until 1 yr, and significant auditory deterioration does not occur until 2 yr of age or approximately 23 of the rat's lifespan.
Role of Inner and Outer Hair Cells in the Hearing Process
Authors: Ward W. D. , Duvall A. J.
Published: October 2005

Exposure of monaural chinchillas to 700–2800-Hz noise at 123 dB SPL produced an initial TTS of about 85 dB, which decreased to zero after 2 weeks of recovery. However, at sacrifice (30 days postexposure) it was found that all the outer hair cells were missing in the first 1.5 turns of the cochlea. In this case, then, the normal behavioral threshold must have been mediated by the normal-appearing inner hair cells. Thus the notion that inner and outer hair cells have different thresholds is probably incorrect, as is the assumption that the absolute threshold is a valid indicator of the state of the cochlea. Exposure to this same noise at 114 dB SPL for 20 min produced an initial TTS of 65 dB that recovered in 4 days; in this case no damage to inner or outer hair cells could be detected 30 days postexposure. The dividing line between a safe and a hazardous TTS—for the chinchilla, at least—therefore lies somewhere between 65 and 85 dB. [Research supported by the National Institute of Neurological Diseases and Stroke.]
Acoustic Radiation from Two Spheroids
Authors: Van Buren A. L. , King B. J.
Published: October 2005

The acoustic radiation from two spheroids whose surface normal velocity distributions are specified is calculated using a Green's function approach. The necessary Green's function is expanded in spheroidal wavefunctions about both spheroids. The unknown expansion coefficients are determined from the boundary condition that the normal derivative vanishes over both surfaces. An addition theorem which expresses spheroidal wavefunctions in one system in terms of spheroidal wavefunctions in another system was developed to facilitate application of this boundary condition. The resulting Green's function is then used to obtain the acoustic fields for any desired velocity distribution over the two spheroidal surfaces. Numerical results for several different two-spheroid configurations and velocity distributions are presented and discussed. Extension to more than two spheroids is also discussed.
Use of Small Arrays for the Study of Acoustic Waves Associated with Atmospheric Disturbances
Authors: McDonald John A. , Herrin Eugene
Published: October 2005

In the spring of the year north Texas is the scene of many natural violent atmospheric disturbances, fronts, tornados, and, in particular, thunderstorms. Five-kilometer triangular microbarograph arrays have been operated in this area for 2 yr and data have been routinely collected in a digital form which is immediately compatible with a computer. This has enabled beam forming techniques to be applied to the data so that as infrasonic signals are associated with a particular event their spectral content and coherency can be examined. Periods of storm-associated signals have been found to be within the Brunt-Vaisala range. On-line beam forming techniques are now being applied to the data and criteria are being established to determine the duration and azimuthal stability that defines such signals. The weather patterns producing the signals are also being examined by routine daily examination of satellite weather photographs.
Transient Analysis of Transducer Arrays Using Z-Transform Techniques
Authors: Stepanishen P. R.
Published: October 2005

An approach is developed to compute the set of time-dependent piston velocities resulting from a prescribed set of electrical inputs which are applied to transducers in a planar array. The approach is based on the application of an approximate Z-transform method to solve the set of equations relating the velocities to the electrical inputs. It is noted the effects of the dynamic response of the transducers and the time-dependent mutual interaction effects among the transducers are included in the approach to compute the velocities. Unlike the timedomain approach which was presented earlier to evaluate the solution [P. R. Stepanishen, J. Acoust. Soc. Amer. 50, 964–974 (1971)], the response of a single transducer can be obtained using the Z-transform approach without solving for the entire set of time-dependent velocities. Numerical results for small multielement arrays are then presented. The results show the transient response time to reach a steady-state condition for a transducer in an array can be either increased or decreased relative to the response time of the uncoupled transducer. Finally, numerical results for infinite sized arrays are presented and discussed.
Environmental-Acoustics Model for Sound Propagation in a Geostrophic Flow
Authors: Franchi E. R. , Jacobson M. J.
Published: October 2005

In most previous studies in underwater acoustics, the effects of ocean hydrodynamics on sound propagation have been ignored. Some recent studies admit current effects but assume them to be independent of the sound-speed structure. In actuality, current and sound speed are related through hydrodynamic and thermodynamic considerations. A shallow-water model of an ocean region, experiencing geostrophic motion, is assumed. Under suitable simplifying assumptions, equations of motion are solved for the pressure and density distributions. Then, assuming constant salinity, the temperature distribution is obtained by means of a state equation. With temperature, pressure, and salinity known, the sound-speed distribution associated with the geostrophic current is computed. Having determined current and sound speed, ray geometry, spreading loss, travel time, and boundary loss and phase shift can be found and are investigated for all rays between a specified CW sound source and receiving point in the current direction. Then, the amplitude and phase of the total acoustic field are considered, and the relation of acoustic variations to environmental variations is studied. [This work was supported in part by Acoustics Programs, Office of Naval Research.]
Fast Field Program (FFP) and Attenuation Loss in Hudson Bay
Authors: DiNapoli F. R. , Powers M. R.
Published: October 2005

In August of 1970, shallow-water attenuation measurements were conducted in Hudson Bay [“Sound Propagation and Reverberation Measurements in Hudson Bay,” paper N13 presented at 81st meeting of the Acoustical Society of America, J. Acoust. Sac. Amer. 50, 103 (A) (1971)]. The experimental results indicated that below 1 kHz, values for the attenuation coeffecient were considerably higher than those which would be predicted from Thorp's formula. One possible explanation for this anomaly was that the experimental values were affected by a significant bottom loss in addition to the attenuation loss in the water column. In order to obtain a better understanding of the governing factors, theoretical propagation loss predictions were made with the FFP. The model incorporates both a layered bottom and a complex wavenumber and the influence of each can be determined parametrically. It was found that at the frequencies of interest the bottom was not a significant factor and that the values of the theoretical attenuation coefficient, needed to match the FFP and experimental propagation-loss results, were in good agreement with the values determined experimentally.
Project Bold Venture: Lake Tanganyika Sound Attenuation Experiment
Authors: Browning D. G. , Gorman J. M. , Jones E. N. , Mellen R. H. , Thorp W. H. , Weigle F. G.
Published: October 2005

In order to determine the cause of the low-frequency sound attenuation anomaly in seawater, comparative measurements are being made in freshwater. The Lake Tanganyika experiment (23°C) was conducted to determine the temperature dependency of the anomaly and to overcome the depth limitations of the previous Lake Superior experiment. The sound propagation run was conducted along a 130-mile track with an average water depth of 2000 ft. Attenuation coefficients were obtained for center frequencies of 315–10000 Hz in 13-oct bands. The results show an anomalous behavior, but the relaxation frequency is poorly defined. Comparison is made with the absorption predicted from chemical analysis of the water.
Model for Signal Fluctuations Induced by Internal Waves
Authors: Stanford G. E.
Published: October 2005

A mathematical model has been derived which describes fluctuations of a CW signal in terms of internal waves. This model demonstrates how oscillations at the Brunt-Vaisala frequency phase modulate a CW signal, producing signal fluctuations. The signal fluctuations are represented by multiple products of exponential functions, each of which is a generating function for Bessel functions. The arguments for these Bessel functions are not time dependent and it becomes relatively easy to obtain a Fourier transform which yields the frequency spectrum for the signal fluctuations. This model is compared with signal fluctuations measured during some Eleuthera to Bermuda propagation experiments conducted in March and July 1968. This comparison is made between the model power spectrum for amplitude fluctuations and the power spectra density functions for the measured amplitude fluctuations. In the region between 1 and 20 cycles/h, where internal waves exert a dominant influence on signal fluctuations, the fit between model and measured data is very good.
Extensions of the Circle Diagram
Authors: Hudimac A. A.
Published: October 2005

The representation of the normalized point impedance of a uniform transmission line by means of a bipolar coordinate system is extended to propagation in nonuniform transmission lines and stratified media (even with a turning point, provided certain asymptotic approximations are also made). The normalized impedance is the solution of a generalized Riccati equation. It is solved approximately by applying the circle diagram to a set of homogeneous lines such that the stepwise varying wavenumber approximates the continuously varying one. The technique is applied to obtain the reflection from and transmission through layers, the eigenvalues and eigenfunctions, and Green's functions. Some further applications will be discussed. [This research is supported by the ONR Acoustics Program.]
Ray-Tracing Evaluation of the Field at and near Caustics
Authors: Spofford C. W.
Published: October 2005

A fundamental limitation of geometrical acoustics is its inability to describe the field near caustics. evaluation of the wave solution near the caustic due to Brekhovskikh [Waves in Layered Media, Academic Press, New York, 1960), pp. 484–485]. For most applications this approach is too restrictive in that it applies only to layered media and only within a thin boundary layer about the caustic. Uniform asymptoti c expansions due to Ludwig [Comm. Pure Appl. Math. 19, 215–250 (1966)] and Kravtsov [Radiofiziko 7, 664–673 (1964)] are valid for arbitrary refracting media and provide the connection between the field at the caustic and coherent ray theory which is valid far from the caustic. Implementation of these results in standard ray-tracing programs is readily achieved since only the travel times and geometrical amplitudes of ordinary ray theory are required. An example is presented to illustrate both the utility of the Ludwlg-Kravtsov theory and its advantage over the older method. [Work supported by Office of Naval Research.]
Analysis Method for Performing a Wavefunction Representation of Speech
Authors: Carey B. J.
Published: October 2005

A high-speed method has been developed for accurately representing the time-domain behavior of human speech. Using, as a starting point, previous work performed at the University of California at Santa Barbara, by Culler and then Markel, a general computer algorithm is defined for representing any sample of human speech, either voiced or unvoiced, by sets of Hanning Cosine Modulated (HCM) wavefunctions. A criterion is presented for defining contiguous wideband filters of fixed characteristics which prefilter the acoustic waveform into a form amenable to wavefunction analysis. A noniterative solution is then discussed for calculating the sets of [A,S,C,φ,F] parameters which define the wavefunctions which represent the time behavior of the speech utterance. These parameter sets are a complete data base which provide a closed form solution to the acoustic waveform from which speech features of interest may be easily extracted. The timevarying information is directly described and a Fourier transform to the frequency domain is available using the HCM model.
Scheme for Automatic Formant Analysis of Speech
Authors: Strong William J. , Purves R. Byron
Published: October 2005

The concept of a formant as representing a peak in the pressure spectrum is assumed to be applicable to both voiced and unvoiced speech. A scheme has been developed that estimates five formant frequencies and amplitudes. The cepstrum technique is used, along with intensity, zero-crossing, and slope-change information, to make voiced-unvoiced decisions and to estimate the fundamental voicing frequency. An attempt is made to detect bursts of energy (due primarily to stop consonants) in the time waveform and to analyze them with sufficient time resolution so that the burst characteristic is preserved. The formant estimating procedure is based on assumed formant exclusive domains in frequency space. Several smoothing procedures are used to remove discontinuities from the formant and fundamental frequency data, and the smoothed values are used to control a five-pole parallel synthesizer. The synthesizer is excited with a pulse train, noise, or a mixture of the two. Examples of the natural and synthetic speech are presented.
On Evaluating Formant Synthesizer Control Signals
Authors: Wathen-Dunn W. , Michaels S. B. , Kriger L. V. , Soron H. I.
Published: October 2005

The problem of evaluating the inherent capability of a given synthesizer to produce good speech is inseparable from the problem of evaluating the signals that control it. Those values of control signals that give the best speech output must be accepted as optimum, and the resulting performance of the synthesizer must be accepted as maximum. Optimizing the control signals must be done, however, not in some average fashion, but with reference to the specific phonemic content of the speech. The objectives of psychoacoustic test procedures for this purpose are thus somewhat different from those used in preference testing of speech communication systems, and so are the means. Forced-choice paired comparisons have been used to evaluate different control signals for one synthesizer producing specific phonemes in specific environments. The results are used as a guide for choosing the best means for deriving the signals from real speech. Evaluations of the control signals for stop consonants from a Glace-Holmes Synthesizer will be presented.
Speech Frequency and Duration Measures as a Function of Chronologic Age
Authors: Shipp Thomas , Hollien Harry
Published: October 2005

Fundamental frequency and duration measures were obtained on 175 normal male subjects whose ages were evenly distributed throughout the range 20–89 yr. It was found that the fundamental frequency contours by age decade show a steady but gradual lowering from the 20-yr-olds at 120 Hz to the lowest mean frequency of 107 Hz in the 40–49-yr-old group. Average frequency then raised steadily throughout the 50-, 60-, and 70-yr decades to reach a high in the 80-yr-old group of 146 Hz. Mean oral reading rate showed a steady increased duration with increased chronologic age, with the 80-yr-old group having a mean duration 50% greater than the mean for the 20-yr-olds. The increase in duration was principally a function of the increase in silent periods during the utterance or in intraphrasal pauses. For both frequency and duration measures the greatest variability was found in the 80-yr-olds with some of these older subjects having frequency and duration values similar to the young adult subjects.
Objective Measures of Peak Clipping and Threshold Crossings in Continuous Speech
Authors: Brady Paul T.
Published: October 2005

This study reports data on the statistics of instantaneous speech levels in continuous speech samples, with special emphasis on threshold crossings and other quantities related to peak clipping. Peak clipping is also treated as speech-correlated noise by assuming that it is caused not by a voltage limiting process, as actually occurs, but by an additive “phantom noise signal” that will cause the original signal to appear to be clipped. Empirical measures are obtained for the percent time a clipping level is exceeded and for the relation between phantom signal (noise) power and clipping level. For each speech sample, clipping levels and power measurements are specified with respect to the equivalent peak level (epl), a recently developed speech level measure [J. Acoust. Soc. Amer. 44, 695–699, (1968)]
Comparison of Two Methods for the Subjective Evaluation of Speech
Authors: Sessler G. M. , West J. E.
Published: October 2005

Speech samples were generated by bandpass filtering and reverberating a sentence recorded with a broad-band system in an anechoic chamber. The speech samples were rated by a large number of subjects utilizing two testing methods: (1) The standard forced-choice paired comparison test where all possible combinations of the samples were presented to the subjects. In this test each subject had to select from each pair the sample he found preferable. This test, although lengthy, gives very reliable results since each sample is compared with all other samples. (2) A rating method where the samples were presented to the subjects just once in random order. In this much shorter test, each subject has to assign a number expressing his preference to each sample. In both cases, the subjective responses were evaluated with a multidimensional scaling method on a digital computer. The results show that the two testing methods yield related representations of the samples. Thus, the second method, which requires considerably less testing, is often adequate for the subjective evaluation of speech samples.
Acoustic Invariance for Stop Consonants
Authors: Cole Ronald A. , Scott Brian L.
Published: October 2005

Previous research using synthetic speech has revealed that rapid frequency changes in the speech wave (transitions) are important cues for the perception of stop consonants. It has also been shown that these transitions are different for a given consonant in different vowel environments. Much emphasis has been placed on this finding, since the demonstration of acoustic invariance suggests that phonemes must be “abstracted” from the speech wave. The present research demonstrates that, for real speech, only a small initial segment of the transition is needed in order to correctly perceive all stop consonants. In fact, a small initial segment of each stop consonant contains all of the necessary acoustic cues needed to hear that consonant. For each stop consonant, we have been able to isolate a specific segment of sound, with most of the transition removed, which is correctly perceived in any vowel environment. For example, it is possible to record /di/ and /du/ on magnetic tape, remove the first 30 msec of each /d/, discard the second 30 msec, and splice the initial segment of either /d/ to either vowel. When this is done, subjects always report hearing a clear /di/ and /du/ These data show that there is an invariant set of acoustic cues for stop consonants.
Auditory Illusions as Caused by Embedded Sounds
Authors: Scott Brian L. , Cole Ronald A.
Published: October 2005

An auditory illusion can be induced by listening to a speech sound presented repeatedly on a loop tape for a period of time. The present experiment shows that auditory illusions reported for CV syllables are caused by consonant sounds “embedded” in the syllable. Embedded sounds were identified for English consonants by removing successive segments from the onset of the sound. For instance, if the initial high-frequency segment present in the sound /sa/ is removed, the remaining sound is a perfectly clear /da/. In fact, subjects report hearing an illusory /da/ while listening to the sound/sa/repeated on a loop tape. Results of the experiment show that illusions reported for each consonant phoneme are perfectly predicted by identifying each of the embedded sounds which exist within them. To further test the hypothesis that embedded sounds cause the illusions, the embedded sounds were removed from the original stimulus and presented to the subjects in the same fashion. This technique does not alter the original syllable, but does eliminate the illusions, showing that the illusions are indeed determined by the embedded sounds.
On the Type of Nonlinearity Generating the Cubic Difference Tone
Authors: Smoorenburg Guido F.
Published: October 2005

The cubic difference tone (CDT), the combination tone corresponding to 2f1−f2 which is often produced by a two-frequency stimulus f1,f2 (f1<f2), was studied with different techniques. Measurements with the cancellation technique gave results in agreement with other studies. The other techniques used were measurements of the effect of the CDT in forward masking, loudness matching with a simple tone of frequency 2f1−f2, and a type of loudness matching with the matching tone and the stimulus f1,f2 alternating at 4 Hz. In the last procedure the level of the test tone had to be adjusted such that a continuous CDT was perceived. These three methods gave about equal estimates of the CDT level. The estimates were significantly lower than the cancellation level. This difference is explained by the fact that in the last three experiments the probe (matching) tone is not presented simultaneously with the stimulus f1,f2 whereas the cancellation tone is. A separate experiment shows that the stimulus components f1,f2 suppress the cancellation tone such that its effective level equals the CDT level revealed with the other three methods. Both CDT behavior and the suppression may be explained by an (idealized) nonlinear transfer function f(x)=xp, 0<p<1.
Pitch Perceptibility for Multicomponent Signals
Authors: Bilsen Frans A.
Published: October 2005

In order to get insight into some aspects of periodicity pitch of complex tones, we determined the masked threshold for perception of periodicity pitch. This threshold was defined as the relative power level of white noise needed to mask a 6%-pitch change in a two-interval forced-choice test; it was assumed, on the base of earlier experiments [F. A. Bilsen and R. J. Ritsma, J. Acoust. Soc. Amer. 47, 469–475 (1970)], to be an adequate quantity indicating pitch perceptibility. The following conclusions could be drawn: (1) For multicomponent signals, pitch perceptibility is independent of phase. The signals investigated were a periodic pulse and periodic noise, both having the same power spectrum, but different phase spectrum. (2) For 3-component signals, pitch perceptibility depends on phase, if n>∼8. (3) For 2- or 3-component signals. pitch perceptibility is optimal and equal to pitch perceptibility for multicomponent signals, if n=∼4. This result underlines and, to a certain degree, explains the existence of the so-called dominant region. (4) The results (2) and (3) together are able to account for the result (1).
Discriminability of Pure-Tone Sequences
Authors: Watson C. S. , Wroton H. W.
Published: October 2005

The detectability of small changes in the frequency of one of the components of a ten-tone sequence was investigated with a same-different psychophysical method. The tones were 40 msec in duration, were randomly selected from the frequency range 250–750 Hz, and were presented at 75 dB SPL. The detectability of a change in the frequency of one of the tonal components is greatest when that component occurs toward the end of the sequence, and when its frequency is toward the outside of the over-all (long-term) power spectrum of the sequence. Under the best conditions Δf is about as detectable in the context of the pattern as when the same 40-msec tone is presented in isolation. In a separate “saturation” procedure one sequence was repeated at 2.5-sec intervals for up to 20 min to determine whether repeated stimulation with an auditory pattern might lead to perceptual distortions. In these experiments it did not.
Some Experiments with Periodically Phase-Reversed Sinusoids
Authors: Corliss Edith L. R. , Burnett Edwin D.
Published: October 2005

When periodically phase-reversed sinusoids are presented to a subject at levels successively raised above auditory threshold, the first perception is of a faint, pure tone. At higher levels a “roughness” is sensed, followed at still higher levels by detectable sidebands. Subjects can match successively the pitches of components of the sidebands; usually, however, just either the sum or difference components are noticed. When reversals occur within just a few periods, the sidebands begin to dominate, but the sensation of the central tone returns if white noise is applied at the contralateral ear. At still shorter intervals between reversals, the central tone can be sensed only at high test levels.
Detection and Discrimination of Interrupted Noise Signals: Short-Term Autocorrelations?
Authors: Pastore R. E.
Published: October 2005

The detectability of an interrupted, band-limited (150–2400 Hz) noise signal was studied as a function of whether the interruption process was periodic or random. A standard monaural 2-1FC detection procedure with a low-level contralateral cue (MDCC) was employed. The cue, an independent noise interrupted synchronously with the signal, could provide the system with frequency, time, and phase information about the interruption process. After considerable practice, performance under the MDCC condition was no better than under the simple monaural detection condition described at the last meeting. The two types of interruptions were discriminable at levels 10–12 dB above the levels required for 75% detectability but not below. The data indicate a fundamental difference between the ability to detect a noise signal and the ability to discriminate the temporal pattern of the signal. These results seem to indicate that the auditory system either does not perform the predicted short-term autocorrelation analysis, or caunot use the results of such an analysis to augment the detection processes. [Supported by PHS grants from NINDS.]
Auditory Induction: Illusory Continuity of the Fainter of Two Alternating Sounds
Authors: Warren Richard M. , Obusek Charles J.
Published: October 2005

If a louder sound (α) and a fainter sound (β) are presented alternatively (α,β,α,β,α⋯), β will seem to be on continuously (i.e., β will not only be heard alone, but will seem to be present along with α) if the intensity difference between the two sounds exceeds a critical value (varying from less than 3 dB to more than 50 dB depending upon the intensity of α and the relative spectral characteristics of the sounds). This illusory carry-over or “auditory induction” (AI) can persist for durations of α lasting several seconds. AI of the fainter sound is a general phenomenon: α and β may be different noise bands, tones of different frequencies, or one a noise and the other a tone. As a rough approximation for some stimulus pairs, if α is capable of masking β with simultaneous presentation, then AI will usually occur when they alternate. Silent gaps of 50 msec between α and β prevent AI. “Phonemic restorations” [R. M. Warren and C. J. Obusek, “Speech Perception and Phonemic Restorations,” Perception Psychophys. 9, 358–362 (1971)] appear to be a special type of AI with illusory phonemes determined by semantic and grammatical rules. Another form of AI having unique rules is found when α and β are identical except for intensity.
Summary of the Surface Transportation Noise Session, Purdue Noise Control Conference
Authors: Close William H.
Published: October 2005

Eleven papers were presented in the Surface Transportation Noise Session pertaining to community annoyance; component source contributions to total vehicle noise; and present techniques for reducing the levels of dominant vehicle noise sources. Recent social survey data were presented which indicated community annoyance with surface transportation noise to be comparable to annoyance with aircraft noise. The ranking of vehicles in descending order of community annoyance was disclosed to be: buses, motorcycles, trucks, automobiles, and traffic in general. Representative sound levels and relative importance of fans, engine, exhaust, and tires were shown to vary from vehicle to vehicle and with speed for a given type. Present design practices and theoretical approaches for automotive exhaust muffler optimization were summarized, as were factors influencing tire noise. An example of application of noise control techniques to a snowmobile was presented. An evaluation of rapid transit system wayside noise was discussed, including projections of expected noise of future high-speed rail systems. Finally, a summary of a scale-model highway noise propagation study was given.
Vibration Control and Biodynamics
Authors: Crocker Malcolm J.
Published: October 2005

There were two significant themes in this session. The first theme concerned vibration control and noise reduction. The first paper (Ruzicka) reviewed fundamental vibration control concepts for reducing undesirable mechanical vibration and noise. Other papers discussed: sound-structural interaction and the use of statistical energy analysis in predicting response and noise radiation (Lyon), vibration control and noise reduction by means of damping materials (Jones), and reduction of machine noise by vibration isolation (Crocker and Hamilton). The second theme concerned biodynamics. One paper discussed active vibration isolation for seated personnel and its particular use to isolate pilots from low frequency vibration in aircraft (Calcaterra). Two other papers (Reynolds and Soedel) and (Abrams and Suggs) discussed hand impedance measurements on human subjects. The motivation behind this work was the growing concern in recent years with the effects of vibrating hand-held tools on the human operators. One remaining paper (Collins, Keller, and Kosmakos) discussed the physiological effects of mechanical vibration, the experimental studies being conducted on male rats.
Review of Aircraft Noise, Rotor Noise, Jet Noise, and Sonic Boom
Authors: Eldred Kenneth McK. , Hubbard Harvey H.
Published: October 2005

This paper discusses the papers presented in the two sessions on aircraft noise at Purdue. The first session on Aircraft and Rotor Noise included invited papers on airport noise (MCK. ELDRED) and rotor noise (REETHOF) and several contributed papers on the prediction of rotor noise both from propellors and from compressors. Two papers also discussed the problem of propagation and absorption of compressor noise in ducts. The second session on Jet Noise and Sonic Boom included an invited paper on response to jet noise (Clarkson) and several contributed papers on supersonic jet noise and on rocket noise. Three papers also discussed the prediction of sonic boom overpressures.
Long-Range CW Sound Propagation in the Shallow Bering Sea
Authors: Mackenzie Kenneth V.
Published: October 2005

Analyses have been completed for measurements to 5, 10, 20, 40, and 80 kyd for 15-min signals of 350, 700, and 1300 Hz transmitted from an anchored source ship to a vertical string of 14 hydrophones spaced 10 ft apart. The receiving ship was anchored 600 yd from the hydrophones. Propagation was studied over flat, 25-fm selected paths where thickness of the substrata had previously been determined by an acoustic survey. Many bathythermograms were obtained by both the source and receiving ships. Every resulting sound speed profile was different—varying continually with time at fixed ranges and along the track. Earlier procedures retained sufficient scatter to obscure any dependence on receiver depth [K. V. Mackenzie, “Long-Range Sound Transmission in the Shallow Bering and Chukchi Seas,” unpublished USNEL Rep. 640 (Nov. 1955)]. Present methods are similar to analysis for four hydrophones [J. Acoust. Soc. Amer. 34, 67 (1962)]. Although closer hydrophone spacing and higher precision demonstrated depth dependence, sound pressure levels computed from average p2 for all hydrophones agree with random phase ray theory [J. Acoust. Sac. Amer. 33, 1505 (1961)]. Not only were practical descriptions of mean propagation acquired, but also information on fluctuation and frequency spreading
Two-Timing in Underwater Acoustics
Authors: Solomon L. P. , Comstock Craig
Published: October 2005

Correlation studies of high-frequency signals over long periods of time must take into account the slowly varying velocity structure of the medium. Experimental studies have pointed out that amplitude and phase of a CW signal are modulated by the fluctuation frequency of the channel stratification. A theory based upon the Method of Multiple Scales as expounded by Van Dyke is applied to the wave equation. The technique appears to be successful, but complicated. An extension of Lighthill's rule, which was originally developed for the Method of Strained Coordinates, is presented.
Origin of the Sound Speed Gradient
Authors: Stickler D. C.
Published: October 2005

An approach more fundamental than that usually taken in underwater acoustics is needed to explain one of the more striking features of the deep ocean, the constant sound speed gradient. This gradient is induced by the earth's gravitational field. The differential equation satisfied by the acoustic pressure is not the usual scalar Helmholtz wave equation. It is shown, however, that the propagation speed is given by the usual formula and, furthermore, the high-frequency behavior is the same as that predicted by the usual scalar Helmholtz equation with variable sound speed. Calculation of the sound speed gradient from the equation of state yields a gradient of 0.019 which compares favorably with the observed value of 0.018. [This work was supported by Contract N00039-71-C-0307].
Seismic Holography
Authors: Fitzpatrick Gerald , Mueller Rolf K. , Steinberg Ron F. , Gupta Ray R.
Published: October 2005

Long-wavelength acoustic holography is a relatively new approach to seismic exploration. A brief history of the field will be given and some new experiments performed by the authors will be described. One of the experiments was conducted in the Gulf of Mexico over a known salt dome. The acoustic source in this case was a hydraulic system emitting single frequency bursts. The other was a land experiment mapping a 100-ft-diam spherical fracture zone at a depth of about 80 ft. A wide-band explosive source was used and the holographlc information obtained by suitable filtering. Preliminary experimental results for both experiments will be given and discussed. A discussion of the new processing methods for image enhancement will conclude the paper.
Speech Intelligibility during Various Work Loads at Normal and Increased Ambient Pressures
Authors: Murry Thomas , Nelson Edward , Swenson Edward W.
Published: October 2005

As diver training and operations expand, it becomes increasingly important to determine the effects of undersea activity on basic functional demands such as communication. The purpose of this study was to assess the effects of work loads at increased ambient pressures on speech intelligibility, Six subjects were exposed to depths equivalent to 0, 50, 100, and 150 ft of seawater. At each depth, the subjects recorded one 25-word PB list at each of four work loads; 0, 50, 100, and 150 W. Subjects were exercised on an ergometer and peddling was maintained at the desired work load by monitoring a preset metronome. Heart rate and respiratory rate were monitored continuously using EKG electrodes and a nasal thermister. The speech signals were recorded with a microphone attached to a headband thus maintaining mouth to microphone distance constant. The recordings were played to naive listening panels to obtain speech intelligibility scores. The results indicate that at rest, speech intelligibility decreases by approximately 20% from surface to 150 ft. With the addition of a work load, there is an additional drop in intelligibility; however, intelligibility does not continue to decrease as work load increases.
Interaural Time Delay of Speech
Authors: Gerber Sanford E.
Published: October 2005

Speech and noise were recorded on tape together at a vowel-to-noise ratio of 0 dB. Time delays were introduced under computer control such that the tape was copied onto two channels with one lagging the other by the specified amount. Five delays were employed: 0, 25, 50, 75, and 100 msec. Six listeners heard 500 CNC words at each value of delay where the words were grouped into lists of 100 and then the lists were randomized in order of presentation with respect to delay. A dip in the intelligibility function was noted at delays of 50 and 100 msec. Discussion of this finding with respect to other findings at 50 msec and with respect to findings dealing with coherent versus incoherent noise follows presentation of these results.
Speech Quality Judgments Using Speech-Correlated and Envelope-Correlated Noise
Authors: McDermott Barbara J. , Bricker Peter D.
Published: October 2005

Several authors and standards committees have proposed that a noise whose amplitude is proportional to that of the speech signal be used as a standard for speech quality judgments. More recently, Horii, House, and Hughes [J. Acoust. Soc. Amer. 49, 1849–1856 (1971)] have used a noise whose amplitude is modulated by the speech envelope as a masker in intelligibility tests. The present experiments employed approximations to both of these types of multiplicative noise along with additive (constant amplitude) noise and other degradations; subjects made evaluative or similarity judgments of the degraded speech signals. Multidimensional analysis was applied to the data in an attempt to determine how the multiplicative noises related, perceptually, to added noise and bandwidth reduction, which had been shown [McDermott, J. Acoust. Soc. Amer. 45, 774–781 (1969)] to be widely separated in perceptual space. Although the three types of noise were perceptually distinct, there was considerable agreement among subjects as to their effects on speech quality.
Acoustic Parameters of Personality
Authors: Brown B. L. , Strong W. J. , Rencher A. C.
Published: October 2005

Twenty-four adult males of diverse socioeconomic background were recorded as they recited a standard passage. They also recited the same passage while pretending to be excited and pretending to be depressed. The normal passages were analyzed and then synthesized by means of a terminal analog synthesizer, and then again synthesized in the following altered forms: (1) raised or lowered pitch, (2) increased and decreased variance of intonation, and (3) faster or slower rate of speaking. Judges listened to normal, excited, depressed, and the various machine-manipulated forms of all voices and made personality judgments on semantic differential adjectives. In every case slowing the voice mechanically made the person sound less competent and speeding it made him sound less benevolent. Mechanically increased intonation makes a speaker sound more competent. Profound changes in the ratings assigned a voice were also induced by the “excited” and “depressed” instructions. The effects of such conscious changes by the speaker were not predictable, whereas those for mechanically produced changes were. Implications of the findings for objective acoustic personality analysis and psychiatric diagnosis are being explored.
Speaker and Listener Pitch Judgments of Human-Produced Vowels
Authors: Wood C. A. , Michel J. F.
Published: October 2005

When speakers attempt to match the pitch of their phonations across different vowels to a constant pure tone, the high vowels have a consistent 2- to 3-Hz higher fundamental frequency (F0) than low vowels. The current study examined whether listeners would demonstrate this same phenomenon if matching a pure tone to the pitch of the human vowel phonations. Samples of the sustained vowels /i, a, æ, u/ recorded by 10 speakers under two conditions, were presented to ten listeners, none of whom served as a speaker. The two conditions were (1) spontaneous “natural” production and (2) matching the F0 of a reference pure tone using a visual frequency aid. The listeners were asked to match the pitch of a pure-tone oscillator to the F0 of the samples. The listeners' average frequency difference between pitch judgments and sample F0 was always less than 0.5 Hz. The listeners' pitch judgments were more directly related to the samples' F0 than were the pitch judgments of the speakers. Implications for speaker self-pitch perception being a different process than listener pitch perception will be discussed.
Five English Vowels Under Eleven Bandpass Filtering Conditions as a Test of “Optimal Octaves” of Perception
Authors: McKenney Elizabeth E. , Asp Carl W.
Published: October 2005

Two adult speakers, one male and one female, spoke the vowels /u,o,a,e,i/. These vowels were bandpass filtered through 11 overlapping octave bands ranging from 100–200 to 3200–6400 Hz, and re-recorded in a randomized order. The stimuli were presented under earphones at 70 dB SPL. Fifteen normal-hearing listeners judged each experimental condition by making a forced choice among one of five vowels, /u,o,a,e,i/. For one analysis, these data were converted to 1–5 values; significant differences were found for filter conditions, vowels, and speakers. In another analysis, the “optimal octave” was determined by the octave band where most listeners perceived each vowel [Guberina (1963)]. In general, the “optimal octave” was low-frequency for /u/, mid-frequency /a/, and high-frequency for /i/.
Threshold Shifts at the Edge of Noise Bands
Authors: Lovell John D. , Carterette Edward C. , Barnebey A.
Published: October 2005

Masked thresholds were taken at frequencies between 700 and 1300 Hz. The maskers were octave-width, bandpass, pink noises at 50-dB sensation level, with component frequencies between 500 and 1000 Hz and between 1000 and 2000 Hz. Masing was found to be 6 to 10 dB less at the edge of the noise bands than within the bands. The critical-band theory prediction is 3 dB less of masking. Predictions from an additive, linear model or a multiplicative, nonlinear model of auditory neural inhibitory interaction lead to different expected amounts of masking at the band edge, even though the sensation pattern in either case can be such as to produce auditory Mach bands. In particular, an increase in masking is predicted near an edge by the linear model whereas a decrease in masking is predicted by the nonlinear model. [Work supported by the U. S. Public Health Service under Grant MH-7809.]
Temporary Threshold Shift Following Exposure to Intermittent Noise with Short-Burst Duration
Authors: Ahaus W. H. , Ward W. D.
Published: October 2005

TTS2 (the TTS 2 min following cessation of exposure) was measured for 11 listeners following a 15-min exposure to both continuous and intermittent, band-limited (1000–2000 Hz) white noise at 114 dB SPL. Twenty-one combinations of on-fraction and burst duration were used. In addition to continuous stimulation, the study employed on-fractions of 0.125, 0.25, 0.50, and 0.75 for each of five burst durations: 12.5, 50, 200, 800, and 3750 msec. For all four noise on-fractions, TTS2 (averaged over 2, 3, and 4 kHz) was significantly less for the 800 and 3750-msec burst durations than for the three shorter burst times. Burst lengths of 12.5, 50, and 200 msec produced equal TTS for on-fractions of 0.25, 0.5, and 0.75, while for the 0.125 on-fraction, a burst duration of 50 msec produced more TTS than did either 12.5 or 200 msec. [Research supported by the Deafness Research Foundation.]
Transient Masking with Sinusoids
Authors: Leshowitz Barry , Cudahy Edward
Published: October 2005

The detection of a 5-msec tonal signal presented at various delays after the onset of a 500-mesc tonal masker was investigated in several transient-masking experiments. How the amount of masking changes as the onset of the signal is delayed relative to the onset of the longer-duration masker was measured for several signal and masker frequencies. In agreement with previous investigations of transient masking there was a large decrease in masked threshold with an increase in signal delay. For example, when the signal was a brief 600-Hz sinusoid masked by a longer duration 800-Hz sinusoid the decrease in masked threshold was almost 30 30 dB. The striking aspect of the above data was the appearance of a large “transient overshoot” (approximately 20 dB) at masker termination. The magnitude of the threshold elevation at masker cessation was found to be dependent on signal frequency, diminishing as signal frequency was increased. These data are interpreted in terms of a stationary, time-invariant energy-detection model.