bluuebunny/crossref_metadata_2025 huggingface dataset:
High-freqeuncy pitch perception
Authors: Burns Edward M. , Feth Lawrence L.
Published: October 2005

The alleged absence of musical pitch information for pure-tone frequencies above about 5 kHz is often cited as constituting evidence for temporal coding of pure-tone pitch below 5 kHz. However, the experimental data on which this allegation is based are both meager and equivocal. In this study, pure-tone pitch perception for frequencies above 10 kHz was investigated using three paradigms: open set melody recognition by naive observers, and melodic dictation and musical interval adjustment by musicians. The results indicate that significant musical-pitch information is available for frequencies above 10 kHz, although it is severely degraded relative to information at low frequencies. We have also investigated another pitch phenomenon which is assumed to be temporally based, the pitch differences associated with amplitude differences between the two components of unresolved two-component complex tones. This pitch information is also significant, but degraded, for frequencies above 10 kHz. The degree of degradation of both types of pitch information at high frequencies is consistent with the relative increase in pure-tone frequency DL's at high frequencies and is thus not incompatible with certain temporal models. [Supported by NINCDS.]
Speech discrimination ability and its relation to psychophysical tuning curves (PTCs)
Authors: Stelmachowicz P. G. , Gorga M. P. , Jesteadt W. , Mort J.
Published: October 2005

PTCs were obtained from normal and hearing-impaired listeners using a simultaneous masking paradigm. All impaired listeners presented fiat moderate sensorineural hearing losses with excellent (90%) speech discrimination ability (NU-6) in quiet. Probe stimuli were presented at 10 dB SL for impaired listeners and were varied from 10–60 dB SL for normal subjects. PTCs were quantified in terms of Q10 and tip-to-tail intersect. Performance-intensity functions for monosyllabic words were obtained as a function of signal-to-noise ratio for broadband and low-pass (500 Hz) filtered noise. Normal listeners demonstrated a decrease in tip-to-tail intersect and an increase in Q10 with increasing probe level. Regardless of whether comparisons were made across constant SL or constant SPL, impaired listeners exhibited smaller tip-to-tail differences than normal subjects with less obvious changes noted in Q10. While speech discrimination performances in broadband noise was similar for both groups, performance in low-pass noise was poorer for the impaired group. The significance of changes in PTC shape in the presence of hearing loss will be discussed.
Stability of earplug attenuation over time
Authors: Humes Larry E. , Ahlstrom Jayne B.
Published: October 2005

The real-ear-attenuation-at-threshold (REAT) procedure was used to measure the stability of the sound attenuation provided by earplugs over time. Two earplugs were evaluated (E-A-R plug and V-51 plug) on six-eight normal-hearing young adults. Attenuation was measured at 500, 1000, 4000, and 8000 Hz following “use” of the protector for 1, 7, 14, 28, 56, and 84 days. “Use” of the protector consisted of insertion of the earplug for a period of 5 min, repeated five times daily. The E-A-R plug demonstrated greater attenuation on the initial day of testing, prior to repeated use. The E-A-R protector, however, exhibited a 10–15 dB decrease in attenuation at all frequencies at the end of the test period while the V-51 plug demonstrated stable attenuation throughout the 84-day period. Consequently, the E-A-R plug provided greater attenuation than the V-51 plug following short periods of use (⩽28 days) while the V-51 plug offered more attenuation following longer periods of use (56 and 84 days). [Work supported by Grant 5-R01-OH00895 from NIOSH.]
Real-ear calibration of circumaural earphones
Authors: Bilger R. C. , Matthies M. L. , Yuchtman M. , Molloy J.
Published: October 2005

The absence of an acceptable calibration technique for circumaural earphones has prevented their use in audiometry in spite of their many advantages (e.g., less stringent control of background noise). A calibration of circumaural earphones (Pedersen) based on direct reading of sound pressure level under the phones is reported here. Measures were obtained for the occluded ear, a mid-concha placement, and a mid-canal placement of an electret microphone (Knowles, 1759a) in response to third-octave bands of noise (100–10 000 Hz). Occluded-ear measures were obtained systematically for six adult, male and six adult, female subjects. In addition, thresholds for the 21 bands of noise were obtained for the Pedersens and for TDH-39 receivers mounted in MX 41/AR cushions. Analysis of acoustic data show an interactive effect of sex and frequency that is smaller than 3 dB and is presumably related to pinna size. The correlations between acoustic and behavioral data were acceptably high. Interpreted in terms of Shaw's extensive measures of sound pressure in an around the human ear, these data provide an acceptable acoustic calibration of these earphones. [Work supported by NINCDS.]
Stereo-laryngoscopy
Authors: Kakita Y. , Hirano M. , Kawasaki H. , Matsuo K.
Published: October 2005

A new method for obtaining the vertical movements of a selected point on the vocal fold during vibration in living human subjects is reported. In a regular ultra-high-speed cinematography of the vocal fold vibration, the vertical movements cannot be measured. To obtain the stereoscopic image, a specially designed mirror (stereo-laryngo mirror) is used. This mirror consists of two identical small mirrors (1 cm×2 cm apiece) combined to keep a specific angle π − 2θ. By using this, two slightly different images of the vocal fold can be photographed in one picture. Provided that θ<1, h≈(d/4θ sin φ), where h is the vertical distance between the mirror and a particular point on the vocal fold; d, the distance between the two images of that point in the picture; and φ, the tilting angle of the mirror from the vertical line. From preliminary data photographed at 2600 frames/s, a resolution of Δh≈0.25 mm was obtained for the vertical movement of hmax − hmin≈0.6 mm. (To reduce the error, measurement of d was repeated four times for an identical frame.) An eight-shaped movement in the frontal plane [first reported by T. Baer, Ph.D. thesis, MIT, Cambridge, MA (1976), for the canine larynx] was also observed.
Pressure-flow relationships in a laryngeal airway model having a diverging glottal duct
Authors: Scherer Ronald C.
Published: October 2005

A laryngeal airway model was constructed using polyester resin in order to empirically test current pressure-flow equations of the larynx. The model was five times larger than life size and had 13 pressure holes to allow measurement of wall pressure profiles. The glottis had a diffuser shape with a 40° divergence angle. This glottal shape is prominent during glottal closure within a vibratory cycle. Three glottal diameters were used: 0.0054, 0.0200, and 0.0918 cm (prototype values). There were essentially two regions of laryngeal pressure for steady flow conditions. The subglottal pressure acted on the laryngeal surfaces upstream of the entrance to the glottis, and the supraglottal pressure acted on the laryngeal surfaces within the diverging region of the glottis. Flow separation near the glottal entrance is suspected. Large negative pressures were not found in the glottis, and the two-duct, glottal pressure-flow equations of Ishizaka and Matsudaira were not well supported. The translaryngeal pressure drop coefficient essentially was dependent only on Reynolds number. [Work supported by NIH and The Voice Foundation.]
Viscoelasticity of vocalis muscle
Authors: Alipour Fariborz , Titze Ingo R.
Published: October 2005

Mechanical analysis of vocal fold vibration requires information on the viscoelastic properties of vocalis muscle. Force response of a dog's vocalis muscle was measured by one-dimensional stepwise elongation of the tissue with a computer-controlled ergometer in an isometric mode. A quasilinear viscoelastic model as proposed by Fung (Biomechanics: Its Foundations and Objectives, 1972) was used to evaluate two time constants and viscoelastic parameters of the vocalis muscle. [Work supported by NINCDS Grant No. NS 16320-03.]
Physiological effects on stop consonant voicing
Authors: Keating Patricia A.
Published: October 2005

Voiced and voiceless stops in initial position are known to differ in Voice Onset Time; they may also differ in closure duration and the duration of closure voicing. Besides the voicing distinction, such factors as the position of the stop in a word or utterance, the stop's place of articulation, and the stress of adjacent syllables will also affect these measures. Data from the literature have been extended with new acoustic measurements for English, Swedish, and other languages. While there is some variation across languages, general patterns, which might be attributable to vocal tract physiology, can be identified. In an attempt to account for these patterns, an electrical analog of vocal tract aerodynamics can be used to study the probable effects of place of articulation and syllable stress on closure voicing offset. The model allows variation of subglottal pressure, glottal area, supralaryngeal cavity volume, wall mechanics, and oral constriction geometry. The extent to which these variables determine universal, presumably inherent, patterns of variation will be considered.
An ultrasound demonstration of tongue anatomy and speech activity
Authors: Shawker Thomas H. , Stone Maureen
Published: October 2005

Soft tissue detail, motion display, and subject safety are features of ultrasound imaging that make it ideal for viewing the tongue and floor of the mouth during speech. In addition to the tongue surface, much of the intrinsic anatomy can be identified including the genioglossus, geniohyoid, mylohyoid, and digastric muscles; fascial boundaries such as the median fibrous septum of the tongue, the floor of the mouth intermuscular septum, and paramedian septums; and the hyoid bone. This study correlates the clinical ultrasound image with the anatomy demonstrated by tongue dissection. In addition, tongue movement during production of the speech sounds /ɑ/, /i/, and /k/ was examined in ten normal speakers and three patients with dysarthria. Tongue movement was found to be consistent for the normal speakers of /i/ and /k/. The three patients with neurological disease showed varying but significant differences in articulation compared to normals including prolonged movement, decreased intrinsic tongue activity, and a tendency to move the tongue as a unit with increased utilization of the floor of the mouth muscles. It appears that real-time ultrasound imaging of the oral cavity is a potentially valuable technique for the investigation of normal and abnormal speech.
Aerodynamic characteristics of the early stages of speech adaptation to a dental appliance
Authors: Hamlet Sandra L.
Published: October 2005

Sibilants, especially /s/, present the greatest difficulty in speech adaptation to a dental appliance. The flow profile for /s/ in an intervocalic context is double humped, reflecting the dynamic interplay between vocal fold opening and tongue gesture. Air-flow profiles for /s/ during the first few minutes of adaptation show a marked change (either much higher or much lower) compared to a compensated condition two weeks later. These flow profiles were also more variable than in the compensated condition—the greatest contribution stemming: from the very first reading of the speech sample. Successive readings showed continued, but more gradual, adjustments. Consideration is given to alterations in timing of articulatory gestures as the mechanism producing these early adaptive changes. [Work supported by NIH.]
Propagation of noise over short distances above asphalt—Refractive and impedance effects
Authors: Daigle G. A. , Piercy J. E. , Embleton T. F. W.
Published: October 2005

On a typical summer day the average temperatures and wind velocity varies with height above the ground. Measurements indicate that the meteorological gradients are very steep near the ground and relatively much smaller above 1 m. The effects of these gradients have usually been ignored in measurements at short distances for source and/or receiver close to the ground—e.g., in the testing outdoors of noise sources such as motor vehicles. We have generated frequencies between 500 Hz and 16 kHz from a point source on a flat asphalt surface. The signals were measured at horizontal distances between 3.5 and 21 m for receiver heights varying between 0.05 and 2 m. The refraction measured under these conditions is found to be strongly frequency dependent because the major part of the gradient occurs in a height which is comparable to the wavelength. In the absence of refraction we have also found by measurements that the acoustic impedance of the asphalt surfaces can be characterized by an effective flow resistivity of about 3×104 cgs units. This is smaller than the equivalent characterization of the heat conduction and viscous boundary layer (105–106 cgs units).
Model experiments of propagation above mixed-impedance ground and of barrier insertion loss
Authors: Nicolas J. , Daigle G. A. , Berry J.-L.
Published: October 2005

The insertion loss of a barrier is the difference in sound level with and without the barrier but with the ground present in both cases. A source on hard ground (asphalt) and a receiver on soft ground (grass) is a common practical situation; however, propagation above mixed soft-hard ground without the barrier is not well understood. We have therefore made measurements indoors above a surface consisting of two known impedances. Frequencies between 100 Hz and 10 kHz were generated by a point source and measurements were made for various configurations of source and receiver, and for varying percentages of soft-hard ground. The measured sound levels are compared with some recent theories of propagation above mixed-impedance ground as well as predictions for the all-hard and all-soft case. For the case of a barrier on a mixed impedance surface the insertion loss can now be calculated more precisely and some results will be discussed.
Routing in the near-terminal area to achieve reductions in airport community noise impact
Authors: DeLoach Richard
Published: October 2005

The introduction of advanced airport terminal-area guidance technology, such as the Microwave Landing System (MLS), will permit aircraft to fly ground tracks through airport communities with significantly greater precision than is possible with current avionics and flight control procedures. It is expected that this improved guidance capability will permit ground tracks to be safely flown which are designed to avoid major population centers in an airport community. This paper will present a general method for defining ground tracks in an airport community which minimize noise impact without violating operational constraints dictated by such factors as air traffic control, aircraft performance, pilot workload, passenger acceptance, etc. A model of airport community noise impact, recently developed at NASA Langley Research Center, has been used to quantitatively assess the reduction in noise impact associated with such flight procedures for a number of airport operating scenarios and airport community population distributions. Results presented in this paper indicate the degree of noise impact reduction which it is practical to expect from noise-minimal aircraft routing in the near terminal area.
Noise control case study: Clark University cogeneration plant
Authors: Foulkes Tim , Copley Lawrence
Published: October 2005

Noise emissions were a major design consideration for an 1800-kW diesel generating plant to be located in the middle of a quiet college campus. The significant noise control features will be discussed along with measurements to verify the field performance of the special doors, windows, and silencers used to control the diesel noise. This case study demonstrates that with adequate noise and vibration control provisions specified in the design stage, a power plant can be compatible with a college campus environment.
Training away the phonemic restoration illusion
Authors: Ressler William H. , Samuel Arthur G.
Published: October 2005

Phonemic restoration is an auditory illusion which arises when a phoneme is removed from a word and replaced with noise. Listeners hear the word as intact: they perceptual]y restore the missing phoneneme. Recently it has been shown that the magnitude of the illusion diminishes with practice [H. C. Nusbaum, A. C. Walley, T. D. Carrell, and W. H. Resslar, Res. on Speech and Hearing Prog. Rep. No. 8, Indiana University (1983)]. One explanation for this result is that subjects learn to ignore their expectation of hearing an intact word by directing attention to the speech waveform itself, specifically attending to the location of the target phoneme. To test this explanation, two groups of subjects received training in the illusion. Group 1 (Control) saw each stimulus word printed on a CRT screen prior to hearing that word. Group 2 (Directed Attention) also saw each word on the screen, but with the target phoneme underlined in each. The performance of the second group relative to the first has implications both for an explanation of the illusion as a misdirector of attention, as well as for theories of the role of attention and the degree to which it may be controlled in the perception of fluent speech.
The representation of speech signals in a model of the peripheral auditory system
Authors: Goldhor Richard
Published: October 2005

The effect of processing speech signals with a simple model of the peripheral auditory system will be described. Compared with a standard speech spectrogram or filter bank analysis, this model transforms the signals in each of the frequency, amplitude, and temporal dimensions. In particular, the temporal transformation models the adaptation of auditory neurons to sustained energy in a particular frequency hand, and results in distinctive responses to segments of speech exhibiting rapid changes in frequency or amplitude (consonants). These transformations result in new relationships between the phonetic identity of the speech and the observable characteristics of the transformed signal. Examples of the response of this model to speech signals will be presented, and the response properties that correspond to the phonetic identity of the signals will be discussed. [Work supported by NSF.]
A pitch-synchronous, variable frame-length, variable bit-rate automatic LPC code generation system
Authors: Vemula N. Rao , Lee Eugene H. , McLaughlin Philip T.
Published: October 2005

There are a number of LPC-based speech synthesis integrated circuits available on the market today. For practical applications of these chips, it is advantageous to utilize an automatic code generation system to help us generate speech code fast. We have developed a pitch-synchronous, variable frame-length and variable bit-rate automatic code generation software system that runs on the HP-1000 minicomputer. The system performs pitch detection, voice/unvoice/silence classification and then marks analysis frames using perceptual distance measures on speech spectra, and also pitch and energy parameters. Using the distance measures, four to five synthetic sounds at different bit rates are generated for each original sound. The system is organized such a way that occasional errors made by it can be corrected manually in a convenient way. A demo tape of sounds synthesized by the General Instrument SP 0256 integrated circuit using the code developed by the automatic code generation system will be played.
Molecular energy transfer measurements with a pulsed spectrophone
Authors: Bass Henry E. , Yan Hai-Xing
Published: October 2005

Taking CO2 and CO2/N2 mixtures as examples, the pulsed spectrophone has been used to investigate vibrational relaxation processes in gases. The electromagnetic excitation of the CO2(001) mode gives rapid cooling of the gas measured as a decrease in pressure followed by heating as the energy supplied by the laser pulse is transferred to translation. The gas then gradually returns to equilibrium due to spontaneous radiation and thermal conduction. By comparing measured pressure waveforms with theoretical simulation, information can be obtained about the energy transfer processes taking place. The vibrational relaxation times compare favorably to those measured using laser induced fluorescence and ultrasonic absorption techniques. The primary advantage of the pulsed spectrophone is its ability to measure multiple relaxation times and determine relaxation mechanisms. [Work partially supported by the Office of Naval Research.]
Sound propagation studies in dilute model dispersion in the low-frequency range
Authors: Gültepe M. A. Barrett , Gültepe M. E. , Yeager E. B.
Published: October 2005

Acoustic absorption measurements using a spherical resonator in the range. 10 to 300 kHz were made in 0.126% by wt polystyrene latex dispersions in an aqueous medium. The particles were synethesized and carefully cleaned and purified after preparation in order to obtain well characterized surfaces. After cleaning and purification, the particle surfaces exhibit mainly sulfonate end groups with hydrogen counterions. The excess absorption measured over and above that attributable to viscous drag and heat conduction losses considerably exceeds that from an aqueous solution of styrene sulfonic acid of comparable molarity, measured with the same technique, and further, differs in frequency dependence. [Work supported by ONR.]
Pressure dependence of sound absorption in an aqueous solution of CaSO4
Authors: Hsu C. C. , Fisher F. H.
Published: October 2005

Sound absorption has been measured in a 0.011M aqueous solution of CaSO4 at 25 °C. The values of maximum absorption per wavelength, (αλ)max × 106, are 10.3 ± 0.1 at 1 atm and 7.7 ± 0.1 at 307 atm. The relaxation frequency is 177 ± 9 kHz at 1 atm and, within experimental error, is independent of pressure. This corresponds to an inner coordination sphere substitution rate of about 106, two orders below that of 2 × 108 quoted by Eigen and Tamm. At 1 atm, the CaSO4 absorption is about 30% that of the same concentration of MgSO4.
Contribution of release burst cues to the perception of stops following [s]
Authors: Repp Bruno H.
Published: October 2005

The contribution of release burst cues to the perception of stop manner and place of articulation was investigated in a fixed [s-læt] context. Truncated natural [p], [t], and [k] bursts of variable intensity were preceded by variable silent intervals. The truncated bursts, though spectrally distinct, conveyed little specific place information but contributed significantly to the perception of stop manner by reducing the amount of silence needed (relative to a burstless stimulus). Burst amplitude was a cue for both manner and place; higher amplitudes led to more “t,” lower amplitudes to more “p” responses. The silent closure interval, a major manner cue, emerged also as the primary place cue in this situation: short intervals led to “t,” long ones to “p” responses. [Research supported by NICHD.]
Using the duplex phenomenon to distinguish speech and nonspeech perception of second formant transitions
Authors: Bentin Shlomo , Mann Virginia
Published: October 2005

A duplex percept results when the second formant transition and the remaining “base” portion of a two formant syllable are presented dichotically in timed synchrony. Subjects integrate information in the transition and base to arrive at a unitary phonetic percept, while they simultaneously perceive the transition as a nonspeech “chirp.” The present study demonstrates that the two components of the duplex percept can be distinguished by their sensitivity to three stimulus properties: the synchrony of base and transition, the relative amplitude of the transition, and the presence of a white noise mask immediately following the transition. This was accomplished with stimuli which consisted of a constant base syllable presented to the left ear and either a [ba] (rising) or a [ga] (falling) second formant transition to the right ear. In one phase of the study, subjects labeled their speech percepts of [ba] or [ga], and in another phase they labeled their nonspeech percepts of rising or falling chirps. Relative to the labeling of nonspeech percepts, the labeling of speech percepts was more susceptible to manipulation of the synchrony of base and transition, but less disrupted by decreases in transition amplitude and by the backward masking of transitions. These findings lend further support to a modular view of phonetic perception. [Work supported by NICHD.]
Asymmetries in the influence of vocalic context on fricative perception
Authors: Mann Virginia A. , Soli Sigfrid D.
Published: October 2005

The perception of a synthetic fricative noise as /∫/ or /s/ is influenced by an adjacent vocalic segment. The influence is greater when the synthetic noise replaces the natural friction in FV syllables than in VF syllables [S. D. Soli and V. A. Mann, J. Acoust. Soc. Am. Suppl. 1 71, S75 (1982)]. These differences could be due to differing articulatory interactions in the production of FV and VF syllables, or, alternatively, to a strategy favoring later-occurring acoustic cues. These hypotheses were tested in a perceptual study using vocalic segments from natural FV and VF syllables, where F = /∫/ and /s/, and V = /ɑ/ and /u/. The vocalic segments were combined with synthetic friction noises to yield VFV stimuli, allowing a direct assessment of the relative strength of vocalic cues on either side of the friction noise. The results showed that the influence of fricative place of articulation cues was restricted to following vocalic segments. However, lip-rounding cues in both the preceding and following segments influenced fricative perception. These findings are consistent with the view that the observed perceptual asymmetries are based on coarticulatory asymmetries. [Supported by NICHD.]
Acoustic and articulatory asymmetries in fricative-vowel and vowel-fricative productions
Authors: Soli Sigfrid D. , Mann Virginia A.
Published: October 2005

When synthetic noise replaces the natural friction in FV syllables and VF syllables, the influence of the vocalic segment on the perception of the noise as /s/ or /∫/ is greater for the FV syllables [S. D. Soli and V. A. Mann, J. Acoust. Soc. Am. Suppl. 1 71, S75 (1982)]. The acoustic and articulatory bases of this perceptual asymmetry were examined. Mean formant tracks displaying confidence intervals and amplitudes were computed for natural FV and VF syllables, where F = /∫/ and /s/, and V = /ɑ/ and /u/. FV and VF transitions were equally stable, although FV transitions exhibited larger coarticulatory differences due to fricative context. FV transitions also occurred during the highest amplitude portion of the vocalic segment, while VF transitions occurred as voicing amplitude was decreasing rapidly before the upcoming voiceless fricative. These acoustic differences would lead to larger vowel context effects for FV syllables. Thus the perceptual asymmetries appear to be a consequence of differences in the organization and timing of the supralaryngeal and laryngeal events which produce initial and final voiceless fricatives. [Supported by NICHD.]
Intelligibility of consonants in VCV fragments excised from connected speech
Authors: Schmidt-Nielsen A.
Published: October 2005

Confusions between pairs of intervocalic consonants excised from connected speech were investigated under several conditions of speech degradation: digital voice processing, noise, bandpass limiting. The stimuli were created by digitally extracting only the VCV portion from read sentences such as the following: “…wants to study teals and their breeding patterns.” /“…wants to study seals and their breathing patterns.” When these were presented to listeners, the distribution of the types of errors that were made, e.g., voicing, nasality, place of articulation, differed from those made on citation form syllable-initial consonants. Intervocalic consonants that were taken from word initial position scored higher initially than those taken from word medial or word final position, but the medial and final segments dropped less under mild degradations than did the initial ones. [Work supported by ONR.]
Perceptual distances between vowels located in a limited phonetic space
Authors: Kewley-Port Diane , Atal Bishnu S.
Published: October 2005

Knowledge of objective measures of perceptual differences between vowel spectra is of crucial importance in the design of speech coders. Toward this goal we have developed procedures for obtaining a multidimensional representation of the perceptual space over a limited vowel region. First, natural sounding vowel stimuli were synthesized using an lpc formant synthesizer. Starting with formants measured from a male talker's vowels, stimuli were generated by varying F1 and F2 in small increments. Listeners judged the dissimilarities between all pairs of vowels several times. These perceptual differences were mapped into distances between the vowels in an n-dimensional metric space using multidimensional scaling. In the first study, 11 steady-state stimuli were synthesized in the region of /i/ and /I/. Although the physical stimulus space had only two dimensions (F1 × F2), the resulting perceptual space had at least three dimensions. In this paper, perceptual spaces for stimuli located in other vowel regions, as well as for vowels embedded in a consonantal context, will be compared. Finally, we will discuss a measure of the difference between these perceptual spaces and their physical stimulus spaces calculated from nonlinear regression techniques.
Analysis and measurement of double frequency spectra of ship noise
Authors: Wu Guoging
Published: October 2005

A practical measurement procedure of periodically locally stationary random process, which is a specific class of locally stationary process, and a method of local stationariness test, are proposed. These approaches are applied to measurement of radiated noise of a ship. It reveals that radiated noise of a ship consists of three parts: stationary noise, periodical signal, and periodically locally stationary random process. From their double frequency spectra, which are obtained using the procedure, we see that the three parts are separated clearly, and some physical characteristics of ships, e.g., rotative velocity of the propeller, number of blades, etc., are exactly measured. So the approaches supply a useful tool for recognition of ship targets. Finally some measurement results of radiated noise of merchant ships and fishery boats are given.
Sound propagation from a coastal source to a deep receiver
Authors: Carey William M.
Published: October 2005

An experiment has been performed to investigate the coupling of surface ship noise to the deep ocean sound channel. These calibrated measurements of sound propagation from an 18-m, 135-Hz source were obtained at a deep ocean receiver from a source tow proceeding from deep water up the Sable Island Bank at ranges between 730 to 910 km. The sound propagation path was from the edge of the Sargasso Sea, through the Gulf stream, and into the cold slope waters over the bank. The mean value of the transmission loss was 110 dB with a slope enhancement estimated to be 4 dB resulting from the combined effects of trapping in the strong shallow sound channel and reflections from the slope. Comparisons with PE calculated results were good and indicate a strong coupling to the deep ocean sound channel. The acoustic field, sampled by 16 transverse hydrophones over six consecutive (12.5-s, 0.08-Hz) samples, yielded a standard deviation of 1.7 dB (6⩽S/N⩽20 dB), consistent with theoretical expectations of a well behaved multipath field. Coherent summation of slope reflected and deep refracted hydrophone signals yielded estimated mean values of the spatial coherence of 0.89 and a spatial coherence length of 460 m, when multipath effects were not dominant; however, these estimates were found to range as low as 0.63 dB and 150 m. These results will facilitate interpretation of the spatial coherence of the ship-induced, “slope-enhanced” contribution to deep ocean noise.
Flexural mechanism of noise generation in the marginal ice zone (MIZ)
Authors: Ngoc Tran D. K.
Published: October 2005

Strong interaction between ice floes and ocean surface waves is normally observed in the MIZ (except in fetch-limited waters). As a result, one of the potential mechanisms of generating ambient noise in the MIX is the flexural vibration of ice floes in response to incoming surface waves. This study is an initial effort to formulate, develop a method of solution, and obtain some theoretical predictions on the problem of noise generation by wave-induced flexural vibration of ice floes. The problem was formulated and solved in two steps: investigation of the flexural response of ice floes to incident ocean waves and evaluation of the sound field radiated from the vibrating floes. The resultant source spectrum calculation shows a potentially significant contribution of 100–150 dB in the 0.1-to 5.0-Hz range. Details of the source spectrum are controlled by floe size and thickness, surface wave spectrum, and ice elastic properties. In general, the source spectrum is extended to higher frequencies and its overall level is raised significantly, when the floe thickness is reduced and/or the floe size is increased. A similar variation of the source spectrum is also predicted for surface wave spectrum associated with stronger wind, but to a much less extent. Predicted levels at frequencies above 5 Hz are insignificant in comparison with measurements. The present study can be readily extended to determine the radiation pattern of this noise generation mechanism.
Linear minimum variance filter for passive localization of moving source
Authors: Wu Guoqing
Published: October 2005

A new method for the optimization estimate of the time varying state of a linear dynamic system is proposed and applied to passive localization. It is a parallel way to the Kalman Filter which sequentially estimates every state. The new method which is called the LMV filter takes the average of K-number observations with different weight coefficients. So in fact the LMV filter is a statistical average method. Compared with the Extended Kalman Filter (EKF) [Hassab et al., J. Acoust. Soc. Am. 70, 1054–1061 (1981)], the new method adopts two more effective measures to improve performance: (1) EKF approximately takes the first term in a Taylor series and ignores other high terms. In the new method a nonlinear estimator is obtained in consideration of high terms. (2) In order to limit extra big noise points which occasionally appear, a threshold is used. Finally, some computer simulation results are given and show that the range estimate of the new method is much better than the EKF in a big noise case and the the LMV filter is an effective approach.
An acoustic seam tracking system for welding and sealing operations
Authors: Prinz F. B.
Published: October 2005

Automatic seam tracking and seam characterization is vital for a successful implementation of arc welding robots in industry. Thus far several contacting and noncontacting seam tracking sensors have been developed. None of the presently available seam tracking systems, however, can be shown to meet all requirements of the arc welding users. The present work discusses the merits and potential of a sonar system for seam characterization in robotic arc welding. A piezoelectric transducer emitting a focused sound beam (frequency, 1 MHz) is mounted on a two-degree of freedom manipulator ahead of the welding torch. The manipulator scans the surface by moving the transducer in a periodic fashion. By measuring the distance to the surface with the pulse method, the surface geometry and the width of the welding gap can be determined.
A simple ultrasonic ranging system
Authors: Maslin Gerald D.
Published: October 2005

There are many instances in robot design where an inexpensive compact ranging system can be of great benefit. The system developed by a well-known camera company to automatically focus cameras is ideally suited to such applications. This system is composed of an electrostatic transducer and a single electronic module containing the drive, receiving, and processing circuitry. It can be used effectively to measure distance or detect the presence of objects within a range of 0.9–35 ft, operating at four discrete frequencies between 50 and 60 kHz. The system exhibits an acceptance angle of approximately 20°. To determine the distance to an object, the time interval between the transmit signal and the received echo must be externally determined. With an appropriate clock, a resolution of 0.25 in. can be obtained. Correction for changes in air temperature must be made to attain good resolution and accuracy throughout a wide temperature range. The system can therefore be used to map a robot's environment, or simply provide an effective means of collision avoidance.
How do we measure hearing by bone conduction?
Authors: Corliss Edith L. R. , Khanna Shyam M.
Published: October 2005

Measurements of the threshold of hearing by bone conduction are made clinically by the use of a driver calibrated upon an artificial mastoid. The driver is calibrated for its voltage response. Normalization data have been obtained by a number of investigators with groups of young adults in terms of force, displacement, and acceleration. This paper discusses the relationship between these norms and the stimulus presented at the otic capsule. The paper discusses the physical factors leading to variability: the head, the driver, and the mechanical coupling between them. Given the problems observed, the force stimulus eliciting threshold sensation is probably the most reliable quantity to measure, and the simplest to calibrate.
Changing pitch and duration in LPC synthesized speech using multipulse excitation
Authors: Caspers Barbara E. , Atal Bishnu S.
Published: October 2005

Pitch and duration changes are fairly simple to realize in pitch-excited LPC vocoders. Recent work [Atal and Remde, Proc. ICASSP 82, pp. 614–617) has shown that use of multi-pulse excitation provides a significant improvement in the quality of synthetic speech. However, procedures for changing pitch with multi-pulse excitation are not known. We discuss methods for introducing pitch and duration changes in speech synthesized with multi-pulse excitation. Pitch is changed by modifying the length of individual pitch periods. Two methods of adjusting the period length were investigated: one by linear scaling of the time axis of the multi-pulse excitation, the other, by adding or deleting zeros in the excitation. The second method produced very little distortion in the synthetic speech, except in those cases where there was a significant interaction between the first formant and the pitch frequency. The scaling of the time axis produced significantly more distortion. Duration changes are accomplished by adding or removing pitch periods. The LPC area parameters and the amplitude of the major excitmion pulse are interpolated as necessary in creating additional pitch periods.
SRS prosody rules for Japanese
Authors: Beckman Mary E. , Hertz Susan R. , Fujitours Osamu
Published: October 2005

SRS, Hertz's language-independent synthesis rule development system, has been used recently to write a set of phoneme-based rules for Japanese. This paper will describe the Japanese prosody rules, which operate on phonological phrases of various sizes to generate natural-sounding prosodic patterns. The phrases are demarcated in the romanized input string by an easy-to-learn annotation scheme. The annotations are translated into boundary symbols and associated boundary features. These features, together with accent marks, are used to generate a segment-by-segment specification by prosody features, such as [low] (low-pitched) and [ace] (accented). Synthesizer parameter rules then refer either to the boundaries' features or to the intervening segments' features to produce various aspects of an utterances' prosodic pattern, such as phrase-final lengthening or the post-accentual pitch fall. Thus, although SRS in its present implementation allows only one level of explicit user-defined synthesis unit, it is relatively easy to set up implicit units of the larger sizes needed to write straightforward prosody rules.
Correlates of acoustic reflex latency in normal-hearing adults of two age groups
Authors: Hannley M. , Ray G.
Published: October 2005

Parametric studies of the acoustic stapedial reflex may be made by measuring its threshold, amplitude, and time course, including latency. Previous reports have established that aging affects the acoustic reflex in at least two ways: by raising its threshold to noise and tone activators; and by reducing the amplitude. In this study, we compared acoustic reflex latency in females of two age groups, 20–40 years and 60–80 years, when the activator tone was presented at equal sound pressure level (SPL), at equal sensation level (SL), and when reflex amplitudes were matched. The results indicated that acoustic reflex latency can be predicted more reliably on the basis of amplitude than on absolute or relative levels of the activator signal. Implications of these findings to the use of latency measures as indicants of central auditory dysfunction are discussed.
Tobramycin-induced hearing loss in a patient with end-stage renal disease
Authors: Johnson David W. , Matzke Gary , Anderson David , Halstenson Charles , Abraham Paul A. , Keane William F.
Published: October 2005

A 60-year-old female undergoing chronic hemodialysis for end-stage renal disease was evaluated audiologically for the frequencies 0.25, 0.5, 1, 2, 3, 4, 6, 8, 9, 10, 11, 12, 13, 14, 15, 16, 17, 18, 19, and 20 kHz over a 7-week period during and after a 4-week course of tobramycin used for control of diverticulitis. Blood levels of the medication were monitored serially and pre- and post-hemodialysis. BAR audiometrics were administered periodically over the 7-week period. Despite conservative medical management where blood levels of medication were allowed to peak at 7 mcg/ml concentration and trough at 2 mcg/ml, the patient demonstrated progressive hearing loss which commenced in the extended high tones and progressively affected lower tone acuity. By the completion of the 7-week period, 6 and 8 kHz were just beginning to show decrease in hearing function while lower primary speech frequencies remained unaffected. BAR data demonstrated no overt changes in wave latencies. Other BAR findings and blood studies were explored with respect to hearing loss findings.
A general theory for regenerative stability
Authors: Thompson R. A.
Published: October 2005

The governing equations for a simple, singly regenerative system are solved in closed form. A metal cutting system is chosen as the model for the theory. The exponential growth index α is expressed in terms of various machine and operating parameters. When α > 0, chatter grows exponentially with time. When α < 0, the system is stable. Plots of α versus speed indicate that at higher spindle speeds αmax is greater, but there is a greater range of stable speeds between unstable zones. Plots of α versus chatter frequency, cutting stability index, lobe precession, and a newly defined variable (the ratio of apparent to actual cutting compliance) indicate that the lobe precession effect is the best sensor for a stability detector. In the special case, α = 0 the general theory reduces to the stability boundary theory developed previously.
Multichannel data acquisition and signal analysis techniques for shock, vibration, and noise measurement
Authors: Deel James C.
Published: October 2005

Advances in computerized digital instrumentation have given the test engineer extremely flexible and powerful dynamic measurement capabilities. However, there are still some inherent tradeoffs and limitations which must be taken into account when selecting digital test equipment or when designing a test using any given equipment. This paper discusses these issues with particular reference to multichannel frequency domain measurements and it describes the design philosophy and architecture of a commercially available multichannel analyzer. Specific applications to signature analysis, modal analysis, acoustic intensity, and rotating machinery monitoring are described.
Patterns for consonant transitions—An iceberg model
Authors: Fujimura O.
Published: October 2005

Using pellet movement data, obtained by the x-ray microbeam system at the University of Tokyo [S. Kiritani, K. Itoh, and O. Fujimura, J. Acoust. Soc. Am. 57, 1516–1520 (1975)], articulatory movement patterns for the tongue and the lower lip have been analyzed for a variety of material, including difference in emphasis placement and different phrasing patterns in sentences. An automatic pattern matching program has compared pairs of sentences to assess relative invariance of iceberg patterns [O. Fujimura, Phonetica 38, 66–83 (1981)]. Preliminary results indicate that emphasis conditions the movement distance strongly, but its effect on the consistency of the crucial parts of the transitions is not apparent.
On the nature of the listener's adjustment for speaking rate during phonetic perception
Authors: Miller Joanne L. , Aibel Iona L. , Green Kerry
Published: October 2005

It is now well established that speaking rate exerts a systematic influence on a variety of temporal properties that convey segmental information, and that the listener appropriately adjusts for speaking rate when using these properties to make phonetic decisions. The concern of the present investigation is the nature of the rate information to which the listener adjusts, whether physical rate or perceived rate. Our experiments focused on the distinction between stop consonant and semivowel in the syllables /ba/ and /wa/. We have reported previously that when listeners use transition duration to identify syllables as /ba/ or /wa/, they do so in relation to the overall duration of the syllables: as the syllables become longer and are perceived as having been produced at a slower rate of speech, a longer transition is required to convert /ba/ to /wa/. In the experiments we now report, we investigated whether the adjustment for rate is based on the actual physical rate of the syllables (i.e., syllable duration) or on their perceived rate. We used a contrast procedure to alter the perceived rate of the syllables while keeping physical rate constant, and then assessed the effect of this manipulation on the identification of the syllables as /ba/ or /wa/. We found that changes in perceived rate induced by the contrast procedure had no reliable effect on syllable identification. However, in a control study, changes in physical rate that produced equivalent changes in perceived rate did reliably alter identification of the syllables. Phonetic identification is thus influenced by the physical rate of the speech, not its perceived rate.
Underwater sound pulse generated by a highly attenuated laser beam moving at supersonic speed over the surface
Authors: Pierce Allan D.
Published: October 2005

A steady laser beam with Gaussian intensity distribution and power P is moving at Mach number M = V/c along the x axis on a water surface (z representing depth). The inhomogeneous wave equation with a source term corresponding to laser beam heating is solved for circumstances when the laser beam penetration depth (reciprocal of absorption coefficient μ) is much less than the beam radius a. The analysis predicts that at r>a [where r = (y2 + z2)1/2] the predominant arrival is a pressure pulse propagating normal to the Mach cone which has a peak amplitude proportional to 1/r1/2, to P/[a5/2μ], to V/M2 − 1)1/4, and which varies with direction about the x axis as cos θ/(cos2θ + M2 sin2 θ)5/4, where cos θ = z/r. All such farfield waveforms are similar with a characteristic plus-minus-plus shape described by a function of a single variable (expressible in terms of parabolic cylinder functions); the characteristic time duration is of the order of a (cos2 θ + M2 sin2 θ)1/2/V. Nonlinear effects become important when the Mach number approaches unity. [Work supported by ONR].