bluuebunny/crossref_metadata_2025 huggingface dataset:
Spectrum transformation properties of the clarinet and saxophone
Authors: Keefe D. H. , Benade A. H.
Published: October 2005

The spectrum transformation function T mr(f) transforms the mouthpiece pressure spectrum into the room-averaged spectrum. One part of T mr is T1(f), proportional to the pressure ratio p1/p0 between the first open tone hole and the mouthpiece. T1(f) is measurable in a lab setup or in a playing experiment. We measured T1(f) at harmonics of the playing frequency f0, with T1(f0) normalized to unity. Odd harmonics of the clarinet's written B♭3, T1(f) = (f/f0)1.25 for 1 ⩽ f/f0 ⩽ 7 and T1(f) = 10 for 7 < f/f0 < 10. By chance, even harmonics show similar behavior in the lower range. They are very irregular above. For the alto sax written G4, T1(f) = [1 + 0.4(f/f0 − 1)2.8] for 1 ⩽ f/f0 ⩽ 4. There is an unexplained discrepancy between lab and played T1(f) for, 4 < f/f0 ⩽ 10. In the lab T1(f) = 10, played T1(f) = 0.024(f/f0)2.4. Throughout all experiments the break in functional form is associated with the tone hole cutoff frequency fc. For the clarinet note fc ≃ 7f0, for the sax note fc ≃ 3.7f0. [Work supported by NSF.]
A test of precategorical acoustic storage for vowels
Authors: Winitz Harris , Rupf John A.
Published: October 2005

A common element among several proposed speech perception models is that an acoustic representation of the speech signal persists briefly in memory prior to phonetic processing. It is also commonly hypothesized that with the passage of time the auditory signals are degraded. These hypotheses were tested in a perceptual experiment in which subjects were presented short segments of vowels for identification. A single glottal pulse from each of the vowels /ɑ, i, I, æ, ε,/ was excised from natural speech and repeated to produce three types of stimuli: two contiguous glottal pulses, four contiguous glottal pulses, and two contiguous glottal pulses followed by two more contiguous glottal pulses with interspersed silences of 10, 20, 40, 60, 80, 120, 240, 480, 1000, and 2000 ms. The stimuli for all vowels and types were randomized and presented to subjects. As expected, the four contiguous glottal pulse stimuli gave better recognition scores than the two contiguous glottal pulse stimuli. It was predicted that as the interstimulus silent intervals increased for the third class of stimuli that the recognition scores would approach those of the two glottal pulse stimuli. In general, this hypothesis was confirmed except for an apparent perceptual recovery interval between 60 and 240 ms.
The degradation of cork-rubber decouplers
Authors: Thompson Corley M. , Horsley L. E.
Published: October 2005

Composites of cork and rubber—commonly called Corprene—have been widely used as decouplers in underwater transducers because of their low sound speeds and densities as well as low cost. However, the acoustic properties of Corprene degrade upon exposure to acoustic decoupling fluids. Gravimetric analyses were carried out to determine the rate of fluid absorption. Resulting losses of the decoupling properties with fluid absorption were also examined. The investigation was performed on five different grades of cork-rubber composite material encompassing neoprene, nitrile, and silicone rubbers. Samples were exposed to various decoupling fluids including castor oil, polyalkylene glycol, and a dimethyl silicone. In order to minimize the degradation of decoupler performance, acceptable fluid types for each composite will be discussed in this paper based on compatibility consideration.
Nearfield and farfield of pulsed acoustic radiators
Authors: Tjøtta J. Naze , Tjøtta S.
Published: October 2005

The acoustic field of a baffled piston source of any shape and with nonuniform velocity distribution is considered, utilizing a generalized version of the impulse response approach developed by Stepanishen [J. Acoust. Soc. Am. 49, 1629–1638 (1971)], and with emphasis on obtaining analytical results and general properties. Effects of amplitude shading and rigidity at the edge are investigated for planar sources. Asymptotic farfield expressions are obtained, which show how the shape of the pulse as well as that of the source do influence the farfield directivity. Nonplanar sources are also considered briefly. [Work supported by the Norwegian Research Council for Science and Humanities and the Office of Naval Research.]
Midfrequency dysfunction in noise-exposed persons
Authors: Humes Larry E.
Published: October 2005

Several recent reports have observed that persons having high-frequency sensorineural hearing loss attributed to noise exposure exhibit abnormal performance on various suprathreshold psychoacoustic tasks for stimuli confined to the midfrequency region where hearing thresholds are normal. It has also been suggested that this apparent midfrequency dysfunction may be partially responsible for the extreme difficulties in understanding speech experienced by some of these individuals. To explore this issue further, the present study measured psychophysical tuning curves, masking patterns and unmasking at 500 and 1000 Hz in 15 listeners having high-frequency noise-induced hearing loss. In addition, measures of speech understanding were obtained in quiet and in noise. Results are compared to similar data obtained from normal hearers. [Research supported by grant 1-R01-OH00972 from the National Institute of Occupational Safety and Health.]
Changes of tinnitus localization following exposure to a pure tone
Authors: Young I. M. , Lowry L. D.
Published: October 2005

A subject with unilateral high pitched tinnitus was exposed to a steady tone, 2000 Hz, at the intensity of 107 dB re 0.0002 μbar for a period of 10 min. Right after cessation of an exposure, tinnitus has completely disappeared from the stimulated ear with ringing and was localized only in the opposite nonstimulated ear. Three days after stimulation, tinnitus returned to the stimulated ear. Since then, localization of tinnitus has been fluctuating, i.e., either limited in one ear only or heard in both ears simultaneously. Frequency of tinnitus in both ears was a pure-tone type of approximately 10 000 Hz and the intensity was 50–55 dB SPL. Whenever tinnitus was localized only to the nonstimulated ear, there was overlapping of interrupted and continuous tone Bekesy tracings at a frequency of 6000 Hz in both ears. When tinnitus was heard either in the stimulated ear only or in both ears, there was separation between two tracings. Factual Bekesy findings were discussed and tried to relate with high-level neuronal mechanism. Fluctuating localization of tinnitus seems to be originating from higher level of central auditory pathways and it is presumed that the slow vertex responses may be used to demonstrate tinnitus objectively.
Cardiovascular effects of occupational and environmental noise: Status report
Authors: DeJoy David M.
Published: October 2005

There has been considerable recent interest in determining whether a relationship exists between exposure to noise and adverse cardiovascular system responses and health outcomes. This paper presents a report of the status of research in this area with respect to assessing the extent and magnitude of the relationship and to supporting the development of protective health effects criteria. Using a comprehensive research plan developed to provide an operational framework for research in this area, recent and ongoing studies are placed along a continuum of required research. An attempt is made to assess the status of animal and human experimental, and epidemiologic research on the cardiovascular effects of noise, to identify major research needs in this area, and to discuss some of the major stumbling blocks hindering progress in understanding this category of noise effects. Consideration is also given to issues related to the validity and generalizability of human and animal data, their complementary nature, and to assessing the evidence for the existence of a causal relationship between noise exposure and cardiovascular system effects.
The development of performance-based auditory aviation classification standards in the United States Navy
Authors: Thomas G. B. , Williams C. E.
Published: October 2005

In an effort to replace chronological age as the determinant of pilot fitness, the Acoustical Sciences Division of the Naval Aerospace Medical Research Laboratory has been evaluating a number of auditory measures for possible inclusion in a performance-based pilot assessment battery. Preliminary data are reported for age-dependent performances on such measures as pure-tone thresholds, clear and degraded speech perception, auditory short-term memory, response time to auditory signals, and auditory selective attention. The relevance of these laboratory-gathered values to operational environment requirements is discussed. [Work sponsored by the Naval Medical Research and Development Command.]
A cost effective extended range audiometer
Authors: Thigpen F. Bruce , Chari Nallan C. A.
Published: October 2005

Current interest in the clinical evaluation of hearing at high frequencies (greater than 8000 Hz), and scarcity of low cost commercial units prompted us to investigate the feasibility of assembling an extended range audiometer. A search of the electronic components and earphones available in the market indicated that an extended range audiometer could be assembled with considerable savings on the cost. The audiometer under consideration is designed for measuring pure tone thresholds at standard 13 octave intervals in the frequency range of 100 through 25 000 Hz. Preliminary results on the audiometer performance will be presented.
Speech discrimination assessment with three closed-response word tests
Authors: Williams Thomas F. , Kernaghan Jean F. , Chari Nallan C. A. , Price Lloyd L.
Published: October 2005

Word discrimination scores for 24 sensori-neural heating-impaired subjects obtained on the Modified Rhyme Test (MRT), Oklahoma University Closed-Response Test (OUCRT), and the California Consonant Test (CCT) were compared. The three word lists were recorded by a male speaker with general American English dialect. The lists were presented monaurally at most comfortable loudness level under four listening conditions: lowband (100–1500 Hz), highband (2000–6000 Hz), speech-in-noise (S/N ratio = 0 dB), and wideband (100–6000 Hz). The subjects scored highest (81.8%-54.6%) on the MRT and lowest (71.9%-40.4%) on the CCT across the conditions employed. Implication of these results in the assessment of speech discrimination skills of the hearing impaired will be presented.
Failure of formant bandwidth narrowing to improve speech reception in sensorineural impairment
Authors: Summerfield Quentin , Tyler Richard , Foster John , Wood Elizabeth , Bailey Peter J.
Published: October 2005

Poor auditory frequency resolution resulting in a “blurred” representation of the short-term speech spectrum may underly the particular difficulty experienced by sensorineurally impaired listeners in distinguishing consonants produced at different places of articulation. To determine whether compensation might be achieved by enhancing the distinctiveness of formant peaks, whispered exemplars of “bet,” “debt,” “get,” “bib,” “bid,” and “big” were synthesized with formant bandwidths varying between unnatural extremes. Psychoacoustical measures of frequency resolution in a combined group of 12 normal and 12 sensorineurally impaired listeners correlated significantly with accuracy of identifying these words although correlations within each group were not systematic. Independently, the psychoacoustical and speech identification measures each suggested that frequency resolution in the impaired group was about four times poorer than normal. The normals' speech-identification accuracy was near 90% at narrow and normal bandwidths, and deteriorated as bandwidth increased. Impaired listeners scored near 60% at normal bandwidths, and also performed worse as bandwidth increased, but contrary to expectations did not improve as bandwidth was narrowed. Possibly, unnaturalness outweighed any advantages of improved formant resolution. Alternatively, little may be gained by making formant bandwidths narrower than the widths of listeners' auditory filters. [Work supported by MRC and by SRC U.K.]
Effects of hyperbaric HeO2 breathing mixtures on divers' speech
Authors: Hollien H. , Hicks J. W. , Søvik S. , Gelfand R. , Lambertsen C.
Published: October 2005

It has been shown that heliox breathing mixtures, plus high ambient pressures, have a deleterious effect upon speech communication. This report attempts to provide more comprehensive information relative to the effects of the HeO2/P environment on speech—especially speech intelligibility, speaking fundamental frequency, and vowel formant shifts; data also are reported on the performance of a speech processor (i.e., the Marconi “unscrambler”). The data base was developed from the recorded speech of divers who read word lists during the decompression phase of a 1600-ft dive; the speech at the surface and at 560, 1000, 1400, and 1600 fsw was analyzed. Speech intelligibility was found to be severely degraded as a function of depth and helium concentration—but not as seriously as had been reported earlier. These effects probably were due to the experience of the divers and improved equipment. As expected, speaking fundamental frequency rose as a function of increasing depth; the shifts were consistent with those previously reported. The vowel formant data provide, for the first time, a comprehensive evaluation of the HeO2 effects on these speech features. As expected, the data revealed an upward shift in the formants as a function of increasing depth but the patterns were not as systematic as expected. The data were utilized to compute “K factors” for the formant shifts [G. Fant and J. Lindquist, STL-QPR 1, 7–17 (1968)]. Finally, the effectiveness of the Marconi system, and its implications, will be discussed. [Work supported by the Norwegian Underwater Institute, Bergen.]
The acoustics of whistling
Authors: Shadle Christine H.
Published: October 2005

Although the acoustics of tones produced by flow through orifices with and without resonators has been studied extensively [Chanaud and Powell, J. Acoust. Soc. Am. 37, 902–911 (1965), G. Succi, physics Ph.D. thesis, M.I.T., June 1977], the extension of such results to an analysis of typical pursed-lips whistling has been largely neglected. In the light of current knowledge about the acoustic properties of both hole tones and the vocal tract, it is theorized that the typical narrow-band peak produced by vortex shedding at the lips should occur at a frequency in the vicinity of the second formant, and should therefore lock in to the second formant and to some extent be driven by it. It should also be possible to observe frequency jumps of a whistle for a given configuration as different flow speeds allow the source frequency to excite neighboring formants. Both of these expectations were borne out in analysis of various types of whistling and phonation of one subject. Implications for extension of these results to unstable flow in other parts of the vocal tract during speech will be discussed.
Position constraint on segment exhange errors in production and memory
Authors: Shattuck-Hufnagel Stefanie
Published: October 2005

Constraints on spontaneous speech error patterns often suggest hypotheses about the mechanism of normal speech production planning which can be tested experimentally. For example, in spontaneous error corpora, phonemic segment exchange errors (like “reef boasts” for “beef roasts”) are found most often in word-initial position: 80% of the consonantal exchanges in the 1981 count of the MIT corpus are word initial, and 84% of the vowel exchanges involve the first vowel of a word. This observation suggests that the initial segments of a lexical entry are differentially susceptible to serial order errors in production. In contrast, a tongue-twister experiment using nonsense syllables like /pof bus bif pas/ elicited more segment exchange errors in final than in initial position. In a pilot experiment using monosyllabic words instead of nonsense syllables, the prevalence of final-position exchanges was somewhat reduced. This positional difference in errors between words and nonwords, if reliable, suggests that segmental order errors can arise at two separate loci in production processing; the possible relation between these two mechanisms and short term memory will be discussed.
Noise Criteria in condominiums and acoustic comfort
Authors: Aziz Javaid
Published: October 2005

The paper deals with noise criteria in condominium situations. Since total inaudibility of intrusive sounds is impossible to achieve with standard construction systems for condominiums, the presence of background noise is considered an important item in the design equation. An attempt is made to define acoustic comfort and related background noise spectrum and level is discussed. A scheme to determine comfort level for occupants in condominiums is presented.
Passive ranging with focusing beamformers and shaded arrays
Authors: Campos Alfredo J. , Futterman John A. H. , Garcia Alex N. , Lockhart Clayton M.
Published: October 2005

We consider focused beamforming using passive arrays as a means of achieving rapid range solutions and demonstrate that a new class of geometry-dependent amplitude shading distributions can improve range resolution. The distribution is determined by treating the array as if it were a set of samples distributed in a hypothetical curved wave front originating from the array focus and by projecting the hydrophone coordinates onto the spherical surface along radii emanating from the focus. We define a continuous shading distribution on this surface to assign weights to each hydrophone input and divide the shading distribution by the wave front sample density. This technique is called “wave front sample density compensated (WSDC) shading.” Simulated performance is presented for a large surveillance towed array with a target whose signature contains a characteristic tonal and is located broadside to the array. A comparison among unit shading, a simple element placement scheme, and the WSDC method shows that, although there is some loss in gain, the focused beamforming method with the WSDC shading provides greater resolution and computational speed and lower side-lobe levels than the other methods. [Work supported by ONR.]
Time delay estimates in passive sonar target localization
Authors: Quazi Azizul H.
Published: October 2005

In passive sonar systems, a target is localized using time delays obtained from signals received at spatially separated points. The variance of the time delay estimates about the true time delay at low signal-to-noise ratio (SNR) depends on SNR, bandwidth, center frequency, and spectral slopes of signal and noise. The accuracy of time delay estimation can be enhanced using a pre-shaping filter. We shall show in this paper that if signal and noise spectral fall off rates are close to each other (within 3 dB/oct), then the preshaping filter does not appreciably enhance the accuracy of time delay estimation. However, if the signal and noise spectral slopes are significantly different (about 6 dB/oct), then the preshaping filter increases the accuracy.
O-ring loss data for square law detectors followed by an O-ring device and an accumulator
Authors: Struzinski William A.
Published: October 2005

A mathematical model is developed to compute the O-ring loss for square law detectors followed by an O-ring device and an accumulator. The O-ring loss is computed using Gaussian and chi-square statistics for a detection probability of 0.5 and false alarm probabilities of 10−3, 10−3, 10−4, and 10−6. The O-ring loss is computed as a function of the number of channels O red and the number of samples in the accumulator. It is concluded that there will be a large signal-to-noise ratio loss if integration is performed following O ring.
The Helmholtz resonator for use as a low-frequency underwater acoustic source
Authors: Young A. M.
Published: October 2005

The principle of the Helmholtz resonator as applied to low-frequency underwater acoustic sources is discussed. The advantages, disadvantages, and limitations of this type of transducer are addressed in general and then applied to a design example. The example will be followed from the equivalent circuit analysis and fabrication of a working scale model through to the experimental evaluation of the full-scale transducer.
An implicit finite-difference PE computer model
Authors: Lee Ding , Botseas George
Published: October 2005

A computer model for predicting propagation loss based on an implicit finite-difference solution of the parabolic wave equation is introduced. Prominent features of this model are: (1) Generality: Capable of solving a wide range of range-dependent and range-independent problems. Automatic handling of the horizontal interface and arbitrary irregular bottom boundary conditions. Capable of handling multiple layers and an artificial bottom. (2) Easy to use: User's efforts are minimized. (3) Portability: Programs are written in FORTRAN and can be transferred easily from one computer to another. (4) Flexibility: A number of options are included. (5) Efficiency: A unique solution is produced with accuracy. Solutions of problems under different environments will be discussed.
Comparison of physical acoustics and geometrical theory of diffraction predictions for backscattering by impedance-covered wedgelike scatterers
Authors: Sachs D. A.
Published: October 2005

Radar scattering studies have shown that scattering predictions based on physical acoustics are generally invalid away from specularly reflecting directions. In this paper, comparisons are made between the predictions of physical acoustics and the geometrical theory of diffraction for acoustic backscattering by impedance-covered wedgelike scatterers. Substantial differences are seen in the results at angles of incidence well off the secularly backscattering direction where diffraction by the wedge edge dominates the return. [Work sponsored by Naval Underwater Systems Center, New London, CT.]
Generation and propagation of the coherently scattered boundary wave over a ridge
Authors: Hollis Stephen J. , Medwin Herman
Published: October 2005

Recent theoretical studies by I. Tolstoy and experiments by H. Medwin and co-workers have shown that a low-frequency sound at near grazing incidence to a surface of rigid hemispherical bosses generates a coherently scattered boundary wave which can be much larger than the spherically divergent direct radiation wave. In the present experiment the axial grazing diffraction of the boundary wave over a slightly rough wedge is measured. Two components of the boundary wave are identified in the shadow region of the wedge: one (RS) develops along the rough upslope then diffracts over the crest in the same manner as the volume wave; the ratio of this boundary wave amplitude to the volume wave amplitude (BWA/VWA) grows as range r01/2 upslope and is constant with increasing range r beyond the crest. The other component (SR) is generated by the diffracted volume wave along the rough downslope and BWA/VWA grows initially as range r0.6 from the crest. For both components the amplitude ratio has a frequency dependence ∫2. Low wave-number grazing propagation over a rough wedge thereby produces a boundary wave in the shadow region whose amplitude can be several times greater than that of a diffracted volume wave over a smooth wedge. [Work supported by Office of Naval Research.]
A study of the pitches produced by violinists while playing a short piece of music
Authors: Mathews M. V. , Miller J. E.
Published: October 2005

Computer recordings were made of the pitches produced by four violinists playing a section of a Mozart (G Major) violin-piano sonata and a G Major scale. Twelve repetitions of the scale and sonata were recorded. The sonata was played with a recorded piano accompaniment. Selected notes in the recorded material were analyzed to study the variation in pitch from repetition-to-repetition of the same note, variation in pitch in different parts of the note, and to observe the pitch variations produced by vibrato. The pitch transitions between successive notes were also observed.
Potential for reducing fan noise by housing design
Authors: Swim William B.
Published: October 2005

Noise is caused by unsteady flow on or near solid surfaces. Reducing the noise generated by fans requires smooth steady internal flow, particularly across the blades and near the cutoff. The housing appears to be the critical design element as it forces a pressure field on the rotor. Data show strong tangential pressure gradients do exist in fan housings. These undesirable gradients produce a continually changing load or pressure rise on the blades. Near the cutoff a load, much above the design point value, is imposed on the blades. As a blade moves from the cutoff to the housing discharge the lead decreases continuously to below the design point value. Each blade, experiencing a cyclic lead, likely goes from pressure surface stall at the cutoff to suction surface stall at the discharge opening. Design point flow occurs only for a short interval between these extremes. A redesigned housing using a radial diffuser upstream of the cutoff and suitably sized collector will reduce the tangential pressure gradient. Measurements at TTU have shown noise reduction of 8 to 12 dB in experimental fans which eliminate housing pressure gradients.
Hammermill blower muffler
Authors: Kessler Frederick M.
Published: October 2005

A unique reactive muffler was designed for a hammermill blower. The hammermill grinds magnesium into a fine powder which is transported to a collector using air flow provided by the blower. The magnesium powder-air mixture is potentially explosive; thus, the required muffler could not contain absorptive material, bends, or chambers which might trap the magnesium powder. An absorptive type of muffler, even one with a Mylar liner, or a multichambered reactive muffler (i.e., expansion chamber, pi, etc.) could not be used. A reactive muffler with an inner passage equal to the exhaust pipe diameter (8 in.) and with an outer chamber properly sized [D. D. Davis, Jr. et al., NACA Tech. Note 1192 (1954)] was designed, built, and tested. Sound level tests have been conducted using only the hammermill blower and in situ; both without magnesium powder. The design technique, resulting muffler, and test results are discussed. Sound level tests using an operating unit grinding magnesium are awaiting the owner's permission. The results of these tests will he reported, if available.
Loudness recruitment in the dihydrostreptomycin-treated patas monkey
Authors: Feitosa Angela G. , Moody David B. , Stebbins William C.
Published: October 2005

Previous research in our laboratory has revealed the extreme sensitivity of the patas monkey (Erythrocebus patas) to the ototoxic effects of dihydrostreptomycin (DHSM) [J. E. Hawkins, Jr. et el., Acta Otolaryngol. 83, 123–129 (1977)]. In this report we present evidence suggesting that this animal can be used for the study of the changes in loudness perception that accompany sensorineural hearing loss. Three patas monkeys were trained to report the presence of pure tones monaurally presented through earphones. Reaction time-intensity functions were obtained at selected stimulus frequencies by the constant stimulus method. After training, DHSM (20 mg/kg/day i.m.) was administered until a change of 40 dB in absolute threshold, independently determined by the staircase method, was observed at either 16 or 32 kHz. The following changes which continued long after drug treatment had ceased were observed: (1) An initial increase in reaction time to tones near threshold, followed in time by an increase at progressively higher SPLs, the magnitude of the change being an inverse function of SPL. (2) A related increase in reaction time to progressively lower frequencies. These results are interpreted as indicating progressive loudness recruitment in a uniquely sensitive animal model. [Supported by Research Grant NS 15108 and Program Project Grant 05785 from NINCDS.]
Pitch salience of various complex sounds
Authors: Houtsma Adrianus J. M.
Published: October 2005

Pitch properties have been reported in the literature for a variety of different sounds. The relative salience of several of these pitch effects was examined through an open-set melody dictation test using several musically trained subjects. The task was to identify on each trial all the notes of a four-note melody, randomly selected from an eight-note diatonic major scale (do-do), by playing them back on a keyboard. The sounds included pure tones, three-tone complexes of random simultaneous and of random time-sequential harmonics, echoed pulses, sinusoidally modulated noise, and 400-Hz wide bandpass noise. Pitch salience was measured by the correlation between the (quantified) melody and response patterns. For pure tones, a criterion of better than 90% correlation determined the fitness of a subjects for this experiment. Measured correlation values ranged from near 100% for pure tones or complexes of simultaneous harmonics to about 50% for complexes of time-sequential harmonics, with intermediate values for the other sounds. Pros and cons of the dictation method are discussed. [Work supported by NIH, Grant NS 11680.]
High-frequency deep-water normal mode calculations
Authors: Ferla M. C. , Jensen F. B. , Kuperman W. A.
Published: October 2005

There are two basic problems associated with performing high-frequency deep-water normal mode calculations. First, in deep water the number of modes is approximately of the order of the frequency, and lengthy calculations must therefore be performed. The second problem is that numerical instabilities occur because of the evanescent nature of the lower modes which are “trapped” in the deep sound channel. Because of these problems, normal mode calculations have typically been done only for frequencies less than a few hundred hertz in deep water. A method has been developed to perform such calculations in the multi-kilohertz region in deep water of the order of 5 km. The results from this technique have been favorably compared with infinite frequency solutions. The wave theory calculations presented, therefore, provide a set of test cases for possible future numerically more efficient approximations.
Virtual mode model of the propagation of underwater explosive sound in a surface duct
Authors: Venne H. Joseph , Holmes Eleonor S.
Published: October 2005

A computer program for computing the received time signature of explosive sound propagated in a surface duet is described. A synthesized pressure-time waveform for the explosion is combined with Fourier transform techniques to calculate the source spectrum, which then is converted with the complex frequency response of the surface duct, as predicted by virtual mode theory [F. M. Labianca, J. Acoust. Soc. Am. 53, 1137–1147 (1973)] to obtain the spectrum of the received signal. The key here is the ability to rapidly evaluate the propagation loss at several hundred frequencies. The received shot time signature is found by an inverse FFT of this spectrum. The model is particularly useful for exploring the dispersion of the propagating modes and the interference effects of the source bubble pulses. A mode-by-mode decomposition of the received signal is also readily obtained. Results are presented for various shot conditions as well as for source signals other than those arising from shots.
Detection of frequency and rate modulation by the chinchilla
Authors: Long Glenis R. , Clark William W. , Miller James D.
Published: October 2005

Thresholds for detection of frequency modulation were obtained from four chinchillas at seven frequencies between 320 Hz and 12.75 kHz and at three sensation levels (20, 40, 60 dB SL). These were compared with human thresholds at 40 dB SL obtained under the same conditions. Our data confirm Nelson and Kiester's [J. Acoust. Soc. Am. 64, 114–125 (1975)] report that the chinchilla has much poorer frequency discrimination than man. In an attempt to separately assess the chinchilla's capacity for temporal coding of frequency, thresholds for detection of a change in the modulation rate of amplitude modulated noise were obtained for modulation frequencies between 10 and 320 Hz. The chinchillas needed a larger rate change for detection than most humans but the relative difference was less than that found for frequency modulation detection. [Work supported by NINCDS Grant NS 03856.]
The role of coarticulation in Spanish fricatives identification
Authors: de Manrique A. B. , Massone M. I.
Published: October 2005

In a previous work [A.M. B. de Manrique and M. I. Massone, J. Acoust. Soc. Am. 69, 145 (1981)] a vowel effect on the phonetic identification of Spanish fricatives /f, s, ∫/ was observed. It was assumed that the effect might be due to the absence of transitions or to the absence of forward coarticulation influences in the friction portions of the syllables used as stimuli. The present experiment was undertaken to evaluate the latter hypothesis. The stimuli were formed by gating out the fricative portions of the natural syllables /fi, fa, fu, si, sa, su, ∫i, ∫a, ∫u/ and combining these portions in CV syllables with the steady-state natural vowels /i, a, u/ by tape-splicing technique. Six listeners were asked to identify the fricative. Results show that the vowel effect completely disappears. This suggests the perceptual relevance of the spectral variations in the fricative portion, due to vowel context, for the identification of the fricative.
The role of acoustic properties in the recognition of nasal and liquid Spanish consonants
Authors: Massone María I. , Gurlekian Jorge A.
Published: October 2005

The present study tested, in experiment I, the relevance of nasal and liquid portions [m, n, r, I, rr] gated out from CV and CVC syllables with vowels [i, e, a, o, u]. The consonantal portions were gated out of the syllables using a general purpose computer program—AUDITS—in a PDP9 computer at the Research Laboratory of Electronics of the Massachusetts Institute of Technology. The split segments were recorded in separate tapes and presented for identification under two conditions: free and forced-choice. Results showed that: (a) the condition employed influenced listeners responses; (b) [r] is the sound less identified, the transition being an important cue for its recognition; (c) the nasal murmur provides cues for manner identification; (d) the [l] portion indicates the presence of a cue place; and (e) [rr] is better identified in initial position. In experiment II, nasal sounds were gated out from Spanish words using the same experimental procedure as in experiment I. The word is correctly identified when the same sound sequence does not conform another Spanish word.
Range effect for /r-w/ coninue of adult and child talkers
Authors: Sharf Donald J. , Benson Peter J.
Published: October 2005

The purpose of this study was to determine the relative vulnerability of /r-w/ continua to the effect of range of stimuli on category boundaries. Two nine-step /r-w/ continua varying in F2 onset were synthesized in the frame syllable /Cej/, one for an adult talker and one for a child talker. Eight college students were paid to listen to the test stimuli during four listening sessions and to respond by pressing response buttons marked “R” or “W.” Stimuli from the full-range continua were presented at the first and fourth sessions and stimuli from the truncated /r/ and truncated /w/ continua were presented at the other sessions. Truncated tests were achieved by omitting the two endpoint stimuli. Analyses of variance indicated the following: (1) Category boundaries shifted away from the truncated ends of the continua; (2) Boundary shifts were greater for the child continua than for the adult continua; and (3) There was no difference between pre- and post-experimental boundaries for the full range continua. These results agree with those obtained for continua varying in voice onset time [Brady and Darwin, J. Acoust. Soc. Am. 63, 1556–1558 (1978)] and formant frequency [Rosen, J. Phonetics 7, 393–402 (1979)]. [Work supported by National Institutes of Health Biomedical Support Grant to the Vice President for Research of The University of Michigan.]
Estimation of current profiles by inversion of tomographic data
Authors: Eisler Thomas F. , New Ronald , Calderone Denise
Published: October 2005

Pulse traveltime data contains information in combined and integrated form on both sound velocity and currents. Inversion methods can be generalized so as to yield estimates simultaneously of sound velocity and current vector components as functions of position. We are concerned with establishing bounds on the estimates derived from inversion procedures. One such procedure is described. In this approach, the estimate of one of the unknown functions may be interpreted as a weighted local average of the actual function with contamination by the other unknown functions. Statistical error may also be included in the analysis. The quality of the inversion is determined by the degree to which contamination is minimized; this is a function of the geometry of the tomographic configuration. Examples are presented for a horizontal ocean slice. The inversion may be performed with or without reciprocal transmissions (interchange of sources and receivers); however, their use dramatically improves isolation of sound velocity from the current vector.
Differences in the F patterns of speech: Tone language versus stress language
Authors: Eady Stephen J.
Published: October 2005

A comparison was made between the fundamental frequency (F0) patterns of continuous speech in Mandarin Chinese and American English. Seven adult male native speakers of each language were asked to read an unemotional narrative text written in their language. The recorded speech signal was analyzed using an F0 extraction program that produced F0 values at 10-msec intervals. The analysis showed the F0 patterns of Chinese to have a greater amount of dynamic movement than those of English. The speech of the Mandarin subjects displayed a greater average rate of F0 change than that of the American subjects. The Chinese speech was also characterized by more F0 fluctuations (peaks and valleys) as a function of time and as a function of the number of syllables. The results are consistent with the notion that the F0 patterns of Mandarin Chinese (a tone language) are determined mainly by the patterns of American English (a stress language) are determined mainly by the placement of primary stress on only a few of the lexical items in a sentence. [Work supported by the Social Sciences and Humanities Research Council of Canada and by BRS Grant RR-05596 to Haskins Laboratories.]
Resolvent formulation of the Green's function for a wedge-shaped ocean
Authors: Kamel A. , Felsen L. B.
Published: October 2005

By the method of resolvents or characteristic Green's functions [L. B. Felsen and N. Marcuvitz, Radiation and Scattering of Waves (Prentice-Hall, Englewood Cliffs, N J, 1973), Chap. 3], one may construct general contour integral representations of Green's functions for separable range-independent and range-dependent ocean profiles. From these general integral representations, one may derive a variety of alternative sound field representations, including those involving normal (discrete and continuous) modes, leaky modes, conventional and generalized rays, hybrid ray-mode combinations, etc. The method cannot be applied rigorously to nonseparable problems such as a wedge-shaped ocean with penetrable bottom. However, for small bottom slopes, we have managed to construct an approximate characteristic Green's function integral based on the ray-cycle invariant and on the requirement that the resulting normal mode representation arising from residues at the resonance poles yields the properly normalized and symmetrized trapped adiabatic modes downslope from their cutoff points. For upslope propagation, as a mode approaches cutoff, its resonance pole approaches a branch point and therefore couples strongly to its radiation field. In this transition region, the Green's function may be reduced asymptotically to a canonical integral which is discussed in detail. Beyond the transition region, the previously trapped mode pole becomes a leaky mode pole. The resulting formulation is compared with recent results obtained by other techniques (A.D. Pierce, J. Acoust. Soc. Am., to be published; J. M. Arnold and L. B. Felsen, this meeting). It is also shown how ray, ray-mode, and other relevant field representations can be obtained from the original integral. [Work supported by ONR Ocean Acoustics Branch.]
Post-vocalic and syllabic /r/ and /l/ in English
Authors: Stemberger Joseph P.
Published: October 2005

The psychological status of postvocalic and syllabic /r/ and /l/ is unclear. Are the postvocalic /r/ and /l/ in bar /bar/ and ball /bal/ part of a complex vowel, like the glide in buy /bay/, or part of the syllable coda, like the stop in bought /bat/? Are syllabic /r/ and /l/ vowels or consonants? A corpus of 6200+ speech errors was examined. Vowel + glide acts as a unit in errors, while vowel + consonant rarely does so. Vowel + /r/ and vowel + /l/ were intermediate, often acting as units, often not. Unlike consonants, /r/ and /l/ are a part of the syllable nucleus, but, unlike glides, are not a part of the vowel. Syllabic /r/ and /l/ show the error patterns of both vowels and consonants. They are syllabic consonants rather than vowels, but show vowel error patterns by virtue of their syllabicness and sharing many features with vowels.
Quality limitations in residual-excited LP speech coding
Authors: Kubina James G. , O'Shaughnessy Douglas
Published: October 2005

In this paper, we report on a study of two factors which determine the RELP speech coder overall transmission rate and output quality, namely, the transmitted residual baseband bandwidth and the baseband encoding technique. The first subjective test revealed a linear relationship between uncoded baseband bandwidth and coder quality referenced to the log PCM standard. The optimum baseband encoding method out of a subset (PCM, APCM, ADM, PPADPCM, ATC) was determined by both an objective (SEGSNR) measure and a subjective (preference) measure derived from the results of a second subjective test. The ADPCM based RELP coder maximized both measures and was used thereafter in a final subjective test which sought to establish a relationship between transmission rate and quality. Test results showed a quasi-linear dependance of quality and transmission rate and also showed that toll quality (7-bit log PCM) cannot be presently achieved with RELP coding.
Acoustic basis for universal constraints on phoneme combinations
Authors: Kawasaki Haruko , Ohala John J.
Published: October 2005

Phonological studies reveal many cross-language similarities in permissible/impermissible phoneme sequences. We attempt to account for such universally attested phonotactic constraints by reference to two aspects of acoustic properties of these sound sequences. First, the magnitude of acoustic modulation in these sound sequences should be directly proportional to their perceptual saliency, which in turn would affect their viability in languages. Second, the similarity between two or more sound sequences should determine the likelihood of confusion and their susceptibility to merger. The magnitude of, and distance between formant trajectories were computed for various sequences of stops, liquids, glides, and vowels. The results correctly predicted the infrequent occurrence of [bw-], [-wu], [-yi] as opposed to frequently observed [-wa], [-ya], [-yu]. Such sequences as [gw-] and [dw-] are predicted to be confusable with [b-], and [by-] is likewise confusable with [d-]. The results also indicated that the combination of a liquid and a back vowel should be rare.
Frequency of occurrence of word sequences
Authors: Umeda N. , Kahn D.
Published: October 2005

A problem encountered in speech synthesis using concatenation methods is the unnaturalness and lowered intelligibility which arise from reducing the duration of stored units (diphones, demisyllables, words, etc.). in word-concatenation systems, this problem is most severe in the case of a sequence of function words, where an extreme amount of contraction is necessary in order to approach natural-speech prosody. We suggest this problem can be circumvented by storing not only isolated words but also certain sequences of words. To this end, we studied the frequency of occurrence of two- and three-word sequences in English, based on the million-word corpus of Kucera & Francis. We find that many sequences of extremely common words occur more frequently than all but the most frequent single words. For example, of the is one of the ten most common “words” of English; 17 two-word sequences have corpus frequencies greater than 1000, making them more common than much, well, should, how, etc. We suggest that the results of our study allow one to properly select a “word”-concatenation vocabulary. Our word-sequence frequency tables should be useful to psychologists and workers in speech recognition as well.
Software synthesis with an array processor
Authors: Levas Anthony , Mattingly Ignatius G.
Published: October 2005

The SYNTH software serial synthesizer [Mattingly, Pollock, Levas, Scully and Levitt, J. Acoust. Soc. Am. Suppl. 1 69, S83 (1981)] has been implemented using a Floating Point Systems AP 120 Array Processor interfaced with a PDP-11/45. A batch of filter coefficients and excitation information is computed by the host from the input parameter values for an utterance and passed to the array processor, which then computes for each sample the glottal excitation and the output of filters representing the vowel and consonant branches of the synthesizer. Coefficients are stored in “table memory” and delay values in one of the two “data pads” of the array processor. Only one instruction apiece. in which a multiplication, an addition (or subtraction), a “main data memory” transfer, and a pointer incrementation are initiated simultaneously, is required for the numerator and denominator loops of the filter. The use of the array processor eliminates any appreciable delay in the execution of synthesis, which has been the one disadvantage of a software synthesizer as compared with a hardware synthesizer. [Work supported by NSF Grant PFR 8006144.]
A new compression technique of speech waveform
Authors: Morito M. , Hosoda K. , Yamada K.
Published: October 2005

A new compression technique of speech waveform has been developed to synthesize natural speech at lower bit rates by a simple method. The basic process of this technique was constructed with waveform symmetrization, ADPCM coding and decoding, segment repetition, and amplitude interpolation. With this technique, it is possible to compress speech information at bit rates 2.8 to 6.5 kbite/s. In the analysis, 256 points of digital speech data were converted to a symmetric waveform having the same amplitude characteristics. The segment waveform (one pitch period length if voiced, 32 sampling periods length if unvoiced) was determined by the converted symmetric waveform and was coded using ADPCM. Further, in order to attain highly information compression, we defined the characteristic value giving periodicity of speech waveforms. In synthesis, one segment of the speech waveform was reproduced using ADPCM data. The same process was repeated several times according to the characteristic value determined in the analysis. This repetition is very efficient for the compression technique of speech waveform. Amplitude interpolation of segments was introduced to mitigate discontinuity between neighboring segments and was achieved by shifting the initial position of the pointer in the ADPCM decoding.