bluuebunny/crossref_metadata_2025 huggingface dataset:
Pitch of two-tone complexes comprising nonsuccessive harmonics
Authors: Houtsma A. J. M.
Published: October 2005

Most studies of the musical pitch of harmonic tone complexes have utilized signals comprising two or more successive harmonics. It is also easily observed that tones from a clarinet, whose even harmonics are heavily suppressed, evoke a strong and unambiguous pitch. The present study provides systematic data on melodic interval recognition by three musically experienced subjects, with sounds made up of two nonsuccessive harmonics. If we describe such a two-tone complex as nf0, (n + m)f0, data were obtained for the ranges 1 ⩽ n ⩽ 10, 2 ⩽ m ⩽ 4, 200 ⩽ f0 ⩽ 1000 Hz. Trends in the data are interpreted in the light of three popular models, the “optimum processor theory” (Goldstein), the “pattern transformation theory” (Wightman), and the “learning matrix theory” (Terhardt). A constraint on performance is proposed which is based on the “analytic” and “synthetic” perception modes of complex tones (Helmholtz). Apparently, when the harmonic spacing increases, the listener is more likely to operate in the analytic mode, perceiving harmonics as individual tones each having its own pitch. This degrades the tracking ability for the missing fundamental. [Work partially supported by NIH.]
“Super-random” uniform-spectrum sequences
Authors: Pollack I.
Published: October 2005

Cramer and Licklider (1957) reported short tonal-like segments while listening to unconstrained random polarity-modulated, constant-interval pulse trains. Since the long-term spectrum of such sequences is uniform, they ascribed the tonal-like segments to a short-term spectral analysis of long runs. Might we achieve “super-random” sequences by restricting long runs? Pierce, Lips, and Cheetham (1977) have developed a procedure for generating sequences which have a uniform short-term spectrum over a defined sampling period. Listening tests show that the discriminability of a uniform-spectrum sequence from an unconstrained random sequence, and the discriminability between two uniform-spectrum sequences, is more closely related to the informational constraints imposed upon the uniform sequences than to the run-length properties of the sequences. Direct estimations of “periodicity strength” verify the discriminability tests. Any statistical restriction tends to yield sequences which “sound more organized” than unconstrained random sequences. The search for “super-random” sequences goes on.
Larynx height and intraoral air pressure during voiced stop production
Authors: Riordan Carol
Published: October 2005

Transglottal airflow is necessary for voicing. During complete oral closure, however, the capacity of the vocal tract to absorb this airflow is insufficient for the voiced stop closure durations commonly observed in speech. Some additional mechanism(s) to absorb glottal airflow must be present, and several have been suggested. This paper focuses on the hypothesis that larynx lowering is one such mechanism. To test this hypothesis, simultaneous larynx height and intraoral air pressure data were recorded for intervocalic /p, b/. Differences in the findings for the voiced and voiceless stop consonants will be evaluated.
Evolution of a concert hall: Lateral reflections and the acoustical design for Wellington Town Hall
Authors: Marshall A. H. , Hyde Jerald R.
Published: October 2005

The evolution of research and acoustical design objectives which have been the basis for the preliminary design phase of the Wellington (NZ) Town Hall is reported. The undoubted success Christchurch Town Hall with four years of evaluation in intensive use, and with the progress of research together indicated the acoustical refinements and modifications to be included in the Wellington Town Hall design. There has become a growing recognition of the fact that the directional properties of early reflections are as important in the perception of sound in concert halls as their temporal distribution. Marshall's early papers on the importance of lateral reflections led to a comprehensive study on the effects of early reflections on subjective acoustical quality in concert halls by M. F. E Barron at Southampton. The study includes details of masking of both high- and low-frequency orchestral sound in lateral reflections, integration time, quantitative prediction of “Spatial Impression,” image shifting, and the difference limen. Results of other studies are presented which give complementary results in listener preference for relative levels of lateral and frontal energy. While model studies and in situ evaluation have shown that conventionally fan-shaped halls cannot usually provide the preferred levels of envelopmental energy, acoustical experience in Christchurch has shown that in halls of this type, the preferred relative levels of lateral sound are achievable. Specific design features are presented including the use of longitudinally oriented “Quadratic Residue Diffusers” which provide multiple lateral reflections. The result is an unpresidented design proposal.
Transducer arrays in medical ultrasonics
Authors: Lele P. P.
Published: October 2005

Ultrasonic instrumentation based on linear and phased array transducers is increasingly being used in clinical diagnosis and yields information not hitherto available. However, the use of ultrasonic arrays is not necessarily restricted to diagnosis. They may, in the future, play an important role in therapy as well. Fundamentals of arrays, current status of systems, and some of the ongoing research on beam shaping by apodization will be briefly reviewed. An attempt will be made to guess at the future based on the current work in various laboratories. [Work supported by NIH.]
Medical ultrasonic arrays
Authors: Rose J. L. , Meyer P. A. , Good M. S.
Published: October 2005

The subject of multielement ultrasonic transducers is presented, with emphasis on describing the ultrasonic transducer design techniques and time-delay pulsing schedules used to achieve real-time imaging, beam steering, and beam focusing. Transducer design techniques make use of a first-order scattering theory in fluids, which model quite nicely many basic problems in diagnostic ultrasound. The various sidelobe energy and resolution problems are explored as a function of such transducer array design parameters as gapping effects, numbers of elements, ultrasonic excitation waveform shapes and frequencies, element size, and element electronic time-delay pulsing schedules. The electronic time-delay pulsing schedules are responsible for waveform constructive and destructive interference phenomena at points of interest in the ultrasonic field; the time-delay schedules can, therefore, be used to achieve focusing and steering, in both linear array and rectangular array transducer systems. Time-delay analysis and mechanical rotation concepts are used to achieve focusing and steering in annular array transducers. A comparison of the resolution characteristics of annular and linear array transducers is discussed. Axial and lateral resolution as a function of depth and various dynamic focusing concepts are also presented. An introduction to the problems and advantages of rectangular array transducer design is outlined. Sample theoretical and experimental results are presented to highlight the principle concepts associated with the design and utilization of ultrasonic transducer arrays in the medical field.
A perceptual locus for anchoring effects in vowels
Authors: Sawusch J. R. , Nusbaum H. C.
Published: October 2005

The perceptual locus of anchoring effects in vowel identification was investigated using a series of synthetic, steady-state vowels ranging from [i] to [I]. In the control conditions, each stimulus occurred equally often. In the anchoring conditions, one of the end points occurred more often than the other stimuli. The category boundary in the anchoring condition shifted toward the more frequently occurring stimulus, relative to the control. A signal detection analysis was used to separate the effects of sensitivity from those of response bias. For the [i] anchor, each subject showed a decrease in sensitivity for the stimulus pair at the category boundary following anchoring. For the [I] anchor group, an increase in sensitivity at the category boundary was found. Changes in criterion placement were inconsistent across subjects. These data indicate that anchoring effects occur at a relatively early stage of vowel processing. Implications of the data for the nature of vowel processing will be discussed. [Research supported by University Funds and the SUNY Research Foundation.]
Detection of acoustically excited droplet shape oscillations with rainbow photometry
Authors: Marston P. L. , Apfel R. E.
Published: October 2005

Shape oscillation resonances of liquid drops levitated acoustically in water have been excited by modulating the deforming part of the acoustic radiation pressure [P. L. Marston and R. E. Apfel. J. Acoust. Soc. Am. 61, S93(A) (1977); Bull. Am. Phys. Soc. 22, 1283 (1977)]. Small deformation of drops was detected by observing the blurring of fringes produced with optical interferometry. An improved method for detecting resonance, to be described here makes use of the rapid change in the phase difference between the deformation and the pressure modulation upon sweeping the oscillation frequency through shape resonance. The deformation produces a shift in the angles of Airy's rainbow maxima associated with the scattering of monochromatic light from a single drop. The scattered optical intensity for a fixed angle was converted to a voltage and compared with a harmonic of the pressure modulation. [Work supported by ONR and Petroleum Research Fund of the American Chemical Society.]
Puzzles about sound absorption and ion pairing in sulfuric acid and MgSO4−HCl aqueous solutions
Authors: Fisher F. H.
Published: October 2005

The results reported by Kurtze and Tamm for sound absorption in MgSO4−HCl solutions and the results reported by Eigen, Kurtze, and Tamm for sound absorption in H2SO4 solutions are discussed. The puzzle is that the HSO4− ion-pairing effects seem to be radically different from those exhibited by NaSO4− and other salts. H2SO4 exhibits significant sound absorption and displays a different concentration dependence below 0.1M than most salts exhibiting sound absorption. When NaCl is added to MgSO4 solutions, the reduction in absorption can be calculated from the equation given by Kurtze and Tamm: A/A0 = [MgSO4/([MgSO4] + f[NaCl]), where A0 is the absorption in pure MgSO4 aqueous solution, A is the absorption when NaCl is added, and the brackets denote concentration of the salt within the bracket. For NaCl−MgSO4 mixtures Kurtze and Tamm found f = 0.21. For HCl−MgSO4 mixtures f was not independent of the ratios of salt concentrations. These puzzles are discussed in the framework of the differential sound-absorption technique suggested by Fisher to measure ion pairing of unsymmetrical electrolytes. [Research supported by the National Science Foundation.]
Cochlear insults from noise, drugs, and genetics
Authors: Webster D. B. , Webster M.
Published: October 2005

In recent years, cochlear damage has usually been documented by the loss of hair cells as judged from surface preparations; in some reports, small samples have also been taken for radial light or electron microscopic analyses. The present study used three groups of deaf guinea pigs: (1) deafened by kanamycin-ethacrynic acid; (2) deafened by 159-dB-SPL noise; and (3) genetically deaf waltzing guinea pigs. Ears were studied by light microscopy, using serial paraffin sections cut in the plane of the modiolus. In all animals, there is more extensive damage to the organ of Corti than simple hair cell loss, and there is a secondary loss of spiral ganglion cells. Basilar membrane, tectorial membrane, Reissner's membrane, and stria vascularis are not affected. The time course for total organ of Corti loss and spiral ganglion loss is dependent upon the type of deafening but the end results are all similar. [Supported by NIH grants NS-12510 and NS-11647.]
A physical model of basilar membrane dissipation
Authors: Allen Jont B.
Published: October 2005

In mechanical models of the cochlea it is very commonly assumed that the basilar membrane (BM) may be treated as a locally reactive impedance, namely that the pressure across the BM is proportional to the BM velocity. This impedance function typically contains three terms which may be identified as the BM stiffness K, BM mass M, and BM dissipation R. K and M have been directly measured and modeled. The damping term R, however, has remained unexplained from both an experimental and a theoretical point of view. We will present a theory of damping in the cochlea that appears to be consistent with known cochlear morphology and with other known bits of relevant information, (i.e., damping as determined by modeling). Finally, we will discuss a possible mechanism which would allow R to be signal level dependent (nonlinear mechanics) as discussed by D. O. Kim and C. Molnar [The Nervous System (Raven, New York, 1975); J. L. Hall, J. Acoust. Soc. Am. 61, 802–810 (1977).]
Temporal cueing effects in backward change masking
Authors: Pastore R. E. , Barnett M. L.
Published: October 2005

In a simple monaural yes-no paradigm observers were asked to detect a 20-ms segment of a 3000-Hz sinusoid under two types of masking conditions both with and without contralateral temporal cues. Backward change masking was produced by a postsignal change in the frequency centroid of the masking noise in which both spectrum level of the noise within the critical band of the signal and its total power remained constant. The change masking was produced by an alternation between two masking noises which were classified as “high” and “low,” corresponding to the perceived pitch of each. Relative to the continuous (no change) conditions, results show a differential masking effect from backward change masking out to a 300-ms delay between signal offset and change onset in the noncued experimental conditions. A timing cue was then provided by a gap in an otherwise continuous, uncorrelated noise in the contralateral ear. The gap cue led the signal by 5 ms and was switched off simultaneously with signal offset. Results show the cue to be effective in reducing backward change masking, thus implicate temporal uncertainty as a factor in backward change masking. These results parallel those reported at previous meetings for standard backward masking conditions [J. S. Puleo and R. E. Pastore, J. Acoust. Soc. Am. 60, S49(A) (1976)]. [Research supported by a grant from NINCDS.]
Prediction versus reality: An evaluation of the NRC traffic noise model
Authors: Halliwell R. E.
Published: October 2005

In order to evaluate the NRC traffic noise prediction model a program of taffic noise measurements has been undertaken. An array of ten microphones, sampled simultaneously, is used to provide information on the propagation of traffic noise, in terms of Leq and other statistical descriptors, for a variety of traffic speeds, car-truck mixes, and site configurations. At present, eight sites with free-flowing traffic have been studied. Five of these were open sites with most of the traffic stream visible from the measuring positions, and three were located in residential streets perpendicular to a freeway. The attenuation with distance of Leq is presented as a function of the effective receiver height, and compared to that predicted by the NRC model.
Categorical perception of synthetic vowels in /d/-vowel-/d/ context and in isolation
Authors: Larkey L. S.
Published: October 2005

Identification and discrimination experiments using series of isolated vowels have shown that these vowels, unlike many consonant series, are not perceived categorically. Recent studies, however, suggest that naturally spoken isolated vowels are identified less reliably than vowels in context. In this study three series of synthetic three-formant syllables varying in ten steps from /I/ to /ε/ or /dId/to /dεd/ were constructed to investigate the following questions: (1) Are vowels in syllable context perceived in a more categorical manner than isolated vowels of similar length? (2) Are isolated vowels of short duration perceived in a more categorical manner than vowels of longer duration? (3) To what extent are the identification and discrimination of isolated vowels of short duration comparable to that of vowels in syllable context? Perception was investigated using standard identification and oddity discrimination procedures. Further investigations are planned using unbiased measures of discrimination. [Supported by grants from NICHD and NIMH.]
Recognition of dichotic melodies by musicians and nonmusicians
Authors: Zatorre Robert J.
Published: October 2005

It has been claimed that musicians process melodies by analyzing the internal relations of the constituent parts, a characteristic left-hemisphere strategy. Nonmusicians, however, must rely on a more holistic and therefore right-hemisphere approach, these strategies changing as a function of training. This hypothesis was tested by giving dichotic recognition tasks to 24 trained musicians and 24 nonmusicians. The stimuli were three types of six-note pure-tone melodies: diatonic melodic, diatonic random and chromatic random; previously rated by musicians as to melodic content. There was also a dichotic speech task as a control. All melodies showed a clear left-ear advantage for both musicians and nonmusicians. Also, a right-ear advantage was found for the speech. Although musicians generally scored higher on both tasks, the magnitude of the laterality differences was equivalent for all groups. The pattern of individual differences likewise does not reveal any support for the hypothesis of left-hemisphere processing of melodies. The results imply that melodies are recognized more by the right hemisphere regardless of specific training. [Supported by NICED.]
On measuring auditory temporal resolution with pseudorandom noise
Authors: Smiarowski Richard A.
Published: October 2005

The minimum perceptible separation between two successive bursts of broad-band noise has been used as a measure of auditory temporal resolution by Plomp [J. Acoust. Soc. Am. 36, 277–282 (1964)] and Smiarowski and Carhart [J. Acoust. Soc. Am. 57, 1169–1174 (1975)]. In these earlier studies, the noise bursts were produced by gating random noise. The present experiment included an additional condition wherein reproducible noise bursts were generated with a pseudorandom noise source. We found no systematic difference in the magnitude of the gap-detection thresholds obtained with the two types of noise. However, the random-noise condition did require a greater number of trials to converge on the target value in the transformed up-down threshold procedure. This effect may stem from the fact that bursts of random noise are considerably variable in their afterringing qualities, whereas a burst of pseudorandom noise is a constant and reproducible waveform. The latter type of noise appears to be a useful tool for reducing variability in listener performance during adaptive testing of auditory temporal resolution. [Supported by V.A. Medical Research Project 7990-03.]
Vowel segment duration as a factor in CV identification
Authors: Castellanos F. X. , Whittaker R. G. , Porter R. J.
Published: October 2005

A previous paper reported the effect of second-formant competition in a consonant-vowel (CV) identification task [R. G. Whittaker, and R. G. Porter, Jr., J. Acoust. Soc. Am. 60, S119(A) ( 1976)]. In the present follow-on study, identical stimulus elements consisting of synthetic two-formant targets representing /ba, da, ga/were paired with the isolated second formants (F2) from each CV in a modified “split-formant” design. The vowel portion of each F2 was segmented to create seven different signals consisting of the F2 transition plus a vowel segment of between 0 and 250 ms. These elements were presented dichotically and monotically with the CV target stimuli at levels equivalent to those of natural speech. The effectiveness of the vowel-segment duration, as measured by F2 intrusions into CV target identification, increases as duration increases from 0 to 100 ms and becomes asymptotic thereafter. The “transition only” condition has a measurable effect, and the rate of increase from there to 100-ms duration appears to be a linear function of vowel-segment duration. In the pairings where conditions for “fused” responses were optimal, the percentage of such judgments was low (14%–20% overall). [Supported in part by NIH (USPHS Grant NS- 11647) and The Kresge Foundation.]
A Rhythmic Model of English
Authors: Vanderslice Ralph
Published: October 2005

The subjective impression that English tends to be spoken rhythmically-that is, with recurrent alternations of stronger and weaker elements-is apparently due chiefly to the rhythmic recession of accents: potentially accentable prenuclear heavy syllables tend to be realized as unaccented if flanked closely by neighboring accented syllables. Accent, though it usually augments the duration of the affected syllable, is primarily manifested by a pitch obtrusion. Despite the myth of isochronism, it takes longer to get from one accented syllable to the next if several unaccented syllables intervene, even if all are light (hence, cet, par., short), than if only one or none does. The durational factors needed, e.g., to synthesize natural-sounding speech (leaving aside such paralinguistic effects as pauses, hesitations, and abrupt changes of speech rate) are (a) inherent syllable duration, depending on phonematic makeup; syllable weight (heavy ones are longer, cet. par.); accent (accented syllables are longer, cet. par.); and location in sense group (terminal syllables are drawled). Rhythmic can be used loosely to refer to differences in any of these factors, but in the strict sense English is better described as arrhythmic.
Wave propagation in strongly curved musical instrument tubes
Authors: Keefe Douglas , Benade Arthur , McCracken George
Published: October 2005

Nederveen (1969) calculated the wave impedance Z and phase velocity v in a curved pipe. These are altered relative to a straight of the same cross section by the amounts φ = +Δv/v and ψ = −ΔZ/Z. We have shown that Rostafinski's exact result (1972) reduces to Nederveen's in the long wavelength limit. We have performed experiments to separately measure φ and ψ along with the junction discontinuity impedance Zδ = jω(pc/πɑ2)δ between curved and straight pipes. Two curved pipes were used, each composed of five butted semitorus sections of tube diameter 2a = 17.5 mm, and length 2S = 39.9 mm with midline radius of curvature R = 12.7 mm (baritone crooks). One pipe was formed as a continuous 212 turn helix, the other as a sinuous duct of alternating curvatures. Resonance frequencies were measured with these tubes connected in various configurations to segments of straight tubing of the same ID. Preliminary data shows φ ≃ 0.058, ψ ≃ 0.059, δ ≃ .029a. Our values are smaller than Nederveen's prediction, ψ = −φ = 0.089, whereas similar experiments by Brindley (1973) showed larger values than theory. Work continues to resolve the discrepancies.
Pitch adjustments in clarinet playing
Authors: Lockwood James C.
Published: October 2005

A mathematical model of the clarinet presented by Wilson and Beavers [J. Acoust. Soc. Am. 56, 653–658 (1974)] is used to examine the effects of reed damping, resonance frequency, and aperture on the playing frequency of the clarinet. It is found that an increase in damping or aperture alone has the effect of flattening the pitch. An increase in the reed frequency tends to sharpen the pitch, but the change is too small to be significant in pitch control. Both damping and reed aperture changes may be effective in pitch adjustments made while playing. This conclusion is in agreement with Backus [J. Acoust. Soc. Am. 35, 305–313 (1963)]. It is suggested that the single effect that dominates pitch control in a given case depends on the shape of the mouthpiece lay. A lay that remains practically flat against the reed out to the lip position and then opens rapidly lends itself to damping control through variation of the position and firmness of the lip. A lay that leaves the reed early, so that there is already a significant gap at the lip position, lends itself to reed aperture control through variation of the lip pressure.
An experimental study of handbell overtones
Authors: Schultz W. W. , Frarey J. L.
Published: October 2005

Handbell choirs are rapidly gaining popularity in this country. It could only be a matter of time after joining a handbell choir that two physicists with a well-equipped vibration analysis lab at their disposal subjected the bells to test. This paper presents the results of analysis of the overtones produced by the handbell. There seemed little agreement with some early literature that suggested that the frequency of bell partials should be 1:2: 2.4: 3:4. In some bells, the fundamental tone was completely missing and the ratio was 2:4:6 indicating a single family of modes while in others it was strongly present and the ratios were 0.4:1:1.6:2 (if 2 is assigned to the labeled tone of the bell) indicating two separate mode families. Spectra are compared for different loudness settings of the bell as well as at impact and later during the after tone.
Practical considerations for atmospheric absorption losses of filtered bands of noise
Authors: Sutherland L. C. , Bass H. E.
Published: October 2005

A new American National Standard to be issued shortly on calculation of the absorption of sound in the atmosphere (Proposed ANSI Standard S1.26) provides a specific method for computing absorption loss coefficients for pure tones and outlines general methods for evaluating absorption losses of filtered bands of noise. This paper considers this latter aspect in more detail and outlines practical considerations of the problem with emphasis on analysis of aircraft noise. Typical aircraft source noise spectra are used along with accurate models for absorption loss and filter response characteristics to predict absorption losses at large distances. It is shown that substantial differences begin to occur between actual band losses and values computed at the center frequency of a filter band when the product of distance, in km. and frequency, in kHz, exceeds about 5 to 15 for octave and one third octave bands respectively. However, frequency weighted overall noise levels are not nearly as sensitive to such errors. The paper also briefly considers the more difficult problem of synthesizing a true source spectrum from measured spectra at a distant receiver. In this case, errors in measured levels due to filter response characteristics require a trial and error approach to accurately compute source spectra.
Interference between direct and ground-reflected waves over grass in the presence of atmospheric turbulence
Authors: Daigle G. A.
Published: October 2005

In a previous paper [G. A. Daigle. J. E. Piercy and T. F. W. Embleton, J. Acoust. Soc. S14–S15. (1977)] measurements of the interference spectrum between direct and ground-reflected waves, at distances up to 150 ft over a flat asphalt surface, were shown to be in satisfactory agreement with a theory that included a known degree of partial covariance between the phases of the interfering waves. The amount of covariance was calculated from measurements of wind and temperature fluctuations in the atmosphere. This theory has now been extended to include propagation over ground of finite acoustic impedance and provides an explanation of the measurements of Parkin and Scholes over grass [J. Sound Vib. 2, 353–374 (1965)] to a distance of 1100 m. Above 300 Hz the sound levels reported by Parkin and Scholes are up to 10 dB higher than predicted by coherent acoustic theory. Although the fluctuations of wind and temperature were not reported, agreement is obtained by assuming standard deviations in phase velocity of about 1 in 1000 and correlation lengths of about 1 m—typical of values measured recently in Ottawa. Covariance of sound amplitude is relatively more important at long distances—at short distances its effect is negligible compared with that of phase covariance; phase-amplitude covariance is negligible at all distances.
An auditory model of vowel normalization
Authors: Greenberg Steven
Published: October 2005

Normalization algorithms which seek to reduce the dispersion of phonetically similar vowels produced by vocal tracts of unequal size exclusively on the basis of formant frequency data ignore a number of perceptually relevant aspects of the acoustic signal such as fundamental frequency, formant bandwidth, and spectral rolloff. These so-called secondary characteristics of the speech signal interact with certain nonlinear properties of the inner and middle ear to produce an output function not predictable from formant frequencies alone. The model proposed here uses F1, F2, F3. and F0 as input to derive an output from a set of transformations based on a set of (a) acoustic parameters (e.g., formant bandwidth, spectral rolloff) and (b) nonlinear properties associated with the inner and middle ear (e.g., critical bands, mechanical sensitivity). The operational characteristics of the model suggest that differences in vocal tract size can, in large part, be offset by appropriate modification of fundamental frequency.
Bias in the judgment of the vowel loudness
Authors: Chuang Chiu-Kuang , Wang William S.-Y.
Published: October 2005

As a continuation of our studies on the psychophysical bias in the judgment of vowel pitch (Chuang and Wang. ASA 60.1 and 62.1) and duration (Wang et al. ASA 60.1), this study was aimed at quantitative investigation of perceptual bias in the loudness dimension. The experimental design and the four vowels [i,e,u, and ɑ] were the same as previous studies with [ɑ] as the reference. The perceptual loudness bias differences were obtained by equalizing the syllabic speech power of the four test vowels. The mean loudness difference, based upon data gathered from 12 listeners, indicated that in reference to [ɑ], [i] was 7.1 dB, [e] was 4.5 dB, and [u] was 3.8 dB louder, respectively. These results give quantitative support to the studies of Ladefoged (1961) and Allen (1971) which showed qualitatively that with speech power normalization, the high vowels [i] and [u] were judged louder than the lower vowel [ɑ]. Furthermore, these perceptually obtained loudness differences are found to be inversely matched to the reported intensity difference of sustained vowels (Lehiste and Peterson, 1958). The negative correlation between the production intensity and the perceptual judgment of loudness in congruent with results reported in our previous pitch and duration studies. [Work supported by NSF grant BNS 76-00017 and partially supported by NS 13274.]
Flow noise in cylindrical hydrophones
Authors: Strasberg M.
Published: October 2005

The noise at frequencies below 10 Hz due to water flow across a rigid cylindrical hydrophone can be estimated from measurements of wind noise detected by microphones inside cylindrical wind screens. For both air and water flows, the flow noise at these low frequencies will not depend on the fluid compressibility provided the cylinder is smaller than the acoustic wavelength. Accordingly, the low-frequency sound pressure p̄ in proportional bands (e.g., 13-octave or octave bands) should obey the simple scaling relation (p̄/12ρU02 = F(fD/U0), where ρ is the density of the fluid (either air or water), U0 the flow velocity perpendicular to the cylinder axis, f the center frequency of the band, D the cylinder diameter, and the form of the function F may depend on the Reynolds number. This relation is used to predict flow noise in water from old data on flow noise inside cylindrical wind screens reported in Bolt Beranek and Newman Report 255 (30 August 1954). The estimates are in reasonable agreement with recent observations of noise below 10 Hz made with a cylindrical hydrophone al sea in low-speed currents.
The intelligibility of whitened and amplitude compressed speech
Authors: Young Lamar L. , Goodman Jeannette
Published: October 2005

Substantial research has demonstrated that amplitude compression can be imposed on speech without degrading intelligibility. Nonetheless, there remain aspects of compression which should be investigated. Consequently, we completed an experiment which investigated (1) the effects of whitening speech prior to amplitude compression, (2) the use of a moderate compression ratio (3:1) versus one that imposed a more drastic reduction on the dynamic range of speech (10:1), and (3) the effects of amplitude compression on speech intelligibility when more than one speaker was present. A competing speech message was electrically mixed with CNC discrimination words and the composite signal whitened and amplitude compressed. Discrimination functions were then obtained for the CNC material by varying the intensity of the competition to yield signal-to-competition ratios of 0, +8, and +12 dB. Persons with normal hearing served as subjects. The results indicate that whitening the speech and the use of the 10:1 compression ratio did not alter intelligibility. Amplitude compression did consistently lower speech intelligibility when more than one talker was present, although this reduction was not substantial and was statistically significant only for the +8 dB signal-to-competition ratio. [Research completed at Auditory Research Laboratories, Northwestern University and supported by Veterans Administration.]
Tone spacing: evidence from bilingual speakers
Authors: Maddieson Ian
Published: October 2005

Two competing hypotheses suggest that level tones in a tone language may be separated in phonetic tone space by either the maximum possible interval or a more-or-less standard interval merely adequate for their identification. It is difficult to determine if either of these hypotheses is correct from comparison of F0 measurements in the literature. Such measurements are not controlled for individual speaker variables and are subject to variation resulting from differences in the elicitation and measurement procedures used. To eliminate such variation, an experiment was conducted using individuals who spoke (at least) two African tone languages, one of which had only two level tones while the other had three. Subjects produced utterances from their two languages consisting of tonally contrastive items in a carrier phrase. As far as practicable, matching segments were employed in the two languages. Recordings of each speaker were subjected to a standardized measurement procedure to determine the F0 of the contrasting level tones in each language. The interval between tones was then calculated to determine which of the hypotheses above better describes the relationship between the spacing of the tones in the two languages spoken by each speaker. [Work supported by NSF.]
Radial trajectories in the F2 × F2 plane as place invariants
Authors: Dalby Jonathan , Port Robert
Published: October 2005

This production study explored the effectiveness of F2 × F2 formant trajectories normalized for the vowel steady-state to differentiate the place of articulation of initial voiced stops. A list of CVd syllables (/b/, /d/, /g/, followed by 11 English vowels) was read once by five speakers and five times by one speaker. Measurements of F2 and Fa were made from wide-band spectrograms at onset of the formants and at the vowel target. The rise or fall in F2 and F2 relative to the vowel steady-state was computed and the two transitions were plotted as a single trajectory toward the normalized vowel target in the F2 × F3 plane. The resulting displays reveal that (1) for some vowels (e.g., /a,ʌ,ɔ/) the angle in the plane at which the trajectories approached the target differentiated the three stops; (2) for others (eg., /o,u/) the extent of the trajectory as well as angle was needed for differentiation; (3) for front vowels the trajectories of /b/ and /g/ were consistently different in angle or extent but /d/ trajectories sometimes overlapped one or both; (4) intraspeaker variability was not noticeably smaller than interspeaker variability.
Parellel processing of auditory and phonetic information: A reassessment using speed-accuracy tradeoff functions
Authors: Wood Charles C.
Published: October 2005

In previous experiments I have used “redundancy gains” in choice reaction time (RT) as evidence that the processing of auditory and phonetic information does not occur in a strict serial fashion, but that some form of shared, simultaneous, or parallel processing must be involved [C. C. Wood, Percept. Psychophys. 15, 501–508 (1974)]. The present experiments reassessed this interpretation using speed-accuracy tradeoff functions as dependent variables instead of mean RT [C. C. Wood and J. R. Jennings, Percept. Psychophys. 19, 92–101 (1976)]. Speed-accuracy tradeoff functions (a) permit response criterion effects to be assessed in RT tasks much as signal detection theory permits such effects to be assessed in detection tasks, and (b) provide estimates of the rate of growth of information over time following stimulus onset. The results of these experiments confirm and extend previous evidence of redundancy gains for fundamental frequency and place of articulation in synthetic consonant-vowel (CV) syllables. However, similar tradeoff functions and redundancy gains of similar magnitude were obtained when the CV syllables were reduced to isolated first and second formant transitions. Thus, we cannot rule out the possibility that redundancy gains for the full CV syllables are based upon only an auditory analysis of the formant transitions instead of phonetic feature identification. Some ways to approach this question experimentally will be discussed.
The modulation transfer function as a link between room acoustics and speech intelligibility
Authors: Houtgast T. , Steeneken H. J. M.
Published: October 2005

Generally speaking, the important perceptual information in a signal is carried by the modulation of the envelope (more precisely, of the envelopes after critical-band filtering, to account for auditory spectral analysis). In relation to speech transmission in enclosures this leads to the notion that loss of intelligibility is related to loss of the original envelope modulations. The degree of modulation loss caused by an enclosure (reverberation, interfering noise) can be quantified objectively by the modulation transfer function (MTF). Experimentally, for a variety of conditions, the MTF was found to correlate well with intelligibility scores obtained with talkers and listeners. The MTF, as a powerful predictor of speed intelligibility, can be obtained either from direct measurements in concrete situation, or from calculations in the design stage of a room or auditorium.
Noise assessment program in Allentown, Pennsylvania
Authors: Shampan Jack , Anderson Pat , Everett Jeff , Iobst Jeff
Published: October 2005

As part of the EPA Quiet Communities Program, the City of Allentown, Pennsylvania, administered a comprehensive noise assessment study to define the scope of their noise problems in development of a noise control program. This assessment effort is the first field test of the methodologies developed by EPA, for uniform application by interested communities in the evaluation of their noise problems, identification of noise control needs, and understanding of the public's reaction to the noise environment and their support for local noise initiatives. Emphasis is placed upon local administration using available local manpower. The assessment program consisted of manual monitoring in over 100 site locations, a social survey of over 500 households, and the integrated analysis of the respondent population to their specific noise environments. The methodological designs of these survey assessment techniques, the resulting findings, and the proposed application of the results in development of responsive local noise control strategies for the City of Allentown will be discussed.
Improvement in speech quality due to time incoherence of quantization noise in separate frequency bands
Authors: Mermelstein Paul
Published: October 2005

New speech coding techniques such as sub-band or splitband coding use independent quantizers to encode the speech information in separate frequency bands. This work is intended to improve our understanding of the perceptual effects of independent quantizers operating on separate frequency bands. The subjective preference of multiplicative white noise introduced independently into two 2-kHz wide speech bands has been compared to that of fullband 0–4 kHz noise of equal overall energy. In terms of both the just noticeable noise level, as well as the noise level perceptually equivalent to full-band noise at signal-to-noise ratios of 20 and 30 dB, introducing multiplicative noise independently into the two bands achieves an SNR gain of 2–5 dB. These results suggest that a significant advantage results from splitting the quantization noise into the separate bands because the time-coherence of the noise components is thereby to a large extent destroyed. This advantage is additional to that derived from the lower signal-to-noise ratio tolerance in the 2–4 kHz frequency band compared to the 0–2-kHz band. [Research supported by Grant A0105 from the National Research Council, Canada.]
Light diffraction by ultrasound in liquid helium-4
Authors: Smith C. W. , Carey R. F. , Rooney J. A.
Published: October 2005

We have employed light diffraction produced by first sound to study nonlinear acoustic phenomena in liquid helium-4 above and below the superfluid transition. PZT-4 thickness mode transducers resonant at 1 or 3 MHz were mounted at either end of a standing wave cavity which is immersed in liquid helium. A laser beam is directed orthogonal to the standing wave produced by one of the transducers with a second transducer acting as a reflector and receiver. The resulting diffraction patterns associated with the Debye-Sears effect were focused projected on a screen and photographed. Assymmetries in these patterns are indicative of nonlinear phenomena. For example, the subharmonic response of the liquid helium is clearly evident in the diffraction patterns above the threshold amplitude for subharmonic production. Distortion of the pattern followed by its complete disruption results as the threshold for collapse cavitation is approached and exceeded. Photographic records of these effects will be presented. [This work sponsored in part by the Air Force office of Scientific Research.]
Flexural wave baffling by a viscoelastic material
Authors: Ko S. H.
Published: October 2005

The effectiveness of a viscoelastic baffle against flexural waves is studied. The physical model is composed of an infinite, elastic plate with vacuum on one side and an adjacent plane viscoelastic baffle on the other side. The viscoelastic baffle is covered by another infinite, elastic plate with a semi-infinite fluid medium lying beyond this plate. The viscoelastic baffle confined between two infinite elastic plates is designed for reducing the pressure fluctuations in the semi-infinite fluid medium resulting from flexural waves produced in the plate. The viscoelastic baffle is a rubberlike material that has a loss factor associated with the shear modulus. The present model is compared to the case where the viscoelastic baffle is replaced by a lossless, nondispersive fluid baffle. Effects of various parameters on the flexural wave baffling are discussed. A comparison between the results for flexural and acoustic wave baffling will also be presented. [Work supported by Naval Material Command.]
The interaural pathway and auditory localization: Cochlear microphonic measures in free field sound
Authors: Rosowski John J. , Saunders James C.
Published: October 2005

Cochlear microphonic 1.0 μV thresholds and phase were measured in pigeons exposed to free-field sound stimulation of varying horizontal azimuths. The pigeons were anesthetized and mounted on a rotating platform at a distance of one meter from a loudspeaker. The relative azimuth of the speaker to the binaural axis of the pigeon could be controlled remotely. The amplitude and phase responses of the CM to tone sources of varying direction were determined. One ear canal was then filled with plasticine and the measures were repeated. Comparison of the control and ear blocked measures showed differences in angular sensitivity of the CM which were related to sound conduction through the interaural pathway. The effects of interaural sound conduction on the azimuthal sensitivity of the CM were small but in the proper direction to increase the amplitude difference between the two ears. These results suggest that the combination of interaural sound conduction and the diffraction of sound about the head enable the pigeon to localize sounds in their best frequency range. [Work supported in part by NSF.]
The prediction of the damping effectiveness of multiple constrained layer damping treatments
Authors: Miles R. N.
Published: October 2005

An accurate, closed form solution is presented to the problem of determining the damping effectiveness of a damping treatment on a beam consisting of an arbitrary number of alternate viscoelastic and elastic layers. It was found that the sensitivity of the damping effectiveness to temperature variations may be controlled by using dissimilar viscoelastic layers.
Vibration of cylindrical shells with and without ring-stiffening
Authors: Eichelberger E. C.
Published: October 2005

The dynamical response of two cylindrical shells, one with and one without ring-stiffening, was investigated. Particular attention was placed on the use of ring-stiffening to provide velocity insertion loss at the point of excitation. The driving-point admittance was measured and compared with predictions using the mean-value characteristic admittance method [E. Skudrzyk, J. Acoust. Soc. Am. 51, 98 (1972)]. This method predicts the geometric-mean of the response of a vibrator with respect to frequency; and is valid at low as well as high frequencies. The basic computations involve the mode masses and mode density. The measurements were in good agreement with the theory, and demonstrate that mean-value characteristic admittance method is a practical means of predicting the vibrational response of shell structures to point excitation. [Work performed at the Physics Department of The Pennsylvania State University and supported by the Office of Naval Research under contract number 474.]
Word recognition and the lexicon
Authors: Jakimik Jola
Published: October 2005

How the listener uses his knowledge of the words in his language during word recognition was investigated using a listening for mispronunciations task. The critical mispronunciations involved changing the initial consonant of the second syllable of a word to produce a nonword. The amount of constraint on the word's identity provided by its first syllable was manipulated. One set of words had first syllables which begin many words in English, e.g., composer, impossible; a second set had infrequent first syllables, e.g., vampire, whisper. Listeners heard these mispronunciations, along with other items, in the phrase “Say __________ again” and in sentences. Times to detect the mispronunciations were about 100 ms faster for words with infrequent, i.e., more constraining first syllables. In addition, detection times reflected the point at which the mispronunciation became a nonword. These results imply that information provided by the beginning of the word is crucial in word recognition, and that determining the identity of a word involves a sequential interpretation of the input.