bluuebunny/crossref_metadata_2025 huggingface dataset:
The discriminability of transition duration: Effects of the amplitude and duration of a following steady state
Authors: Jamieson Donald G. , Slawinska Elzbieta
Published: October 2005

When listeners are asked to discriminate changes in the duration of a brief frequency glide which is followed immediately by a longer, steady-state, pure tone, they show better performance when the duration of the glide is in the range 40–60 ms than when it is longer or shorter. This effect could result from the interaction of two psychoacoustic functions—the first, a discriminability function, which follows Weber's Law and the second, a masking function. Three experiments were undertaken to test this explanation. In experiment 1, the glide began at 400 Hz, ended at 700 Hz, and was immediately followed by a steady-state pure tone of equivalent amplitude, also at 700 Hz. Overall stimulus duration was fixed at 200 ms, while glide and steady-state values varied in 10-ms intervals, as follows: 10,190; 20,180; 30,170; 40,160; 50,150; 60,140; 70,130; 80,120; 90,110. This experiment replicated previous results [D. Jamieson and E. Slawinska, J. Acoust. Soc. Am. Suppl. 1 74, S67 (1983)] in finding the discriminability (− ln η) was a maximum and bias (ln β) a minimum, when glide duration was in the interval 40–60 ms. Stimuli for experiment 2 were identical to those for experiment 1, except that the amplitude of the steady-state pure tone was reduced 10 dB. Stimuli for experiment 3 were identical to those for experiment 1 except that the overall duration of the stimulus was fixed at 10 ms. Thus, glide and steady-state pure-tone components had the following durations in ms: 10,90; 20,80; 30,70; 40,60; 50,50; 60,40; 70,30; 80,20; 90,110. Results obtained in experiments 2 and 3 are consistent with the hypothesis that the effect seen in experiment 1 is due, in part, to the masking of the glide stimulus by the following steady-state pure tone. In both cases, the point of maximum discrimination (− ln η) shifted to shorter glide durations (20–40 ms) than was found in experiment 1.
Syllable identification by younger and older adults
Authors: Elliott L. L. , Busse L. B. , Lyons L. E.
Published: October 2005

Three synthesized syllables differing in place of articulation ([bɑ], [dɑ], and [gɑ]) were used in a three-alternative identification task. Preliminary trials established the range of stimulus levels at which each S's performance, for each syllable, varied from below 50% to nearly 100% correct. Each of the three syllables was presented at five different intensities differing in 5-dB steps. The three syllables and the five levels for each syllable were presented in intermixed random orders. Listeners (18–23 and 60–75 years old) all had normal hearing for their ages. Syllable identification thresholds (i.e., levels required for 50% correct performance) were higher for the older adults, as might be expected because of their poorer sensitivity. Older listeners also exhibited more shallow identification functions—an outcome that cannot be attributed to sensitivity differences since stimulus levels were adjusted for individual subjects. Possible explanations for these shallow slopes are considered. [Work supported, in part, by NSF.]
Intelligibility of English “voicing” produced by Arabs
Authors: Mitleb Fares
Published: October 2005

The implicit claim of traditional approaches to the phonetic and phonological aspects of second language acquisition by adults is that pronunciation difficulty occurs almost exclusively at the abstract level of segmental features. Thus the “contrastive analysis hypothesis” predicts pronunciation difficulty for second language learners when the phonetic inventory of the native language lacks a phoneme of the target language. It would predict, then, that English /p/ and /v/ should be difficult for Arabs since Arabic lacks these two phonemes. An intelligibility test, however, revealed that our native English listeners had no serious difficulty identifying “voiced” from “voiceless” obstruents in the English minimal pairs spoken by ten Arabs. Our experimental approach, which is based on the very theoretical framework of traditional approaches to the analysis of foreign accented speech, nevertheless yielded results which counter the predictions of that framework. It also verifies the results of recent research on foreign accent [F. Mitleb, J. Acoust. Soc. Am. Suppl. 1 73, S29 (1983)] that nonsegmental level differences and/or similarities between languages play an important role in second language learning. [Research supported by Yarmouk University, Irbid, Jordan.]
Perceptual assessment of the intervocalic formant transition place cues of /s/ in /sɪsi/ and /sɪsɑ/
Authors: McCasland George P.
Published: October 2005

This is an extension of the investigator's previous work [J. Acoust. Soc. Am. Suppl. 1 71, S105; and 72, S16 (1982)]. A series of spoken /sɪsi,sisɑ/ tokens was modified using a PCM speech editor. The /s/-noise segments were excised. The original intervocalic /ɪ-i,ɪ-ɑ/ contexts and the /ɪ,i,ɑ/ vowels in isolation were assembled into a perceptual identification test and low-pass filtered at 3 kHz with an attenuation slope of 48 dB/oct. As a consequence, the noise and transient energy coextensive in time but higher in frequency than the vocalic formant structures was removed more completely than before. Subjects identified the consonants heard adjacent to the vocalic segments. There was better agreement as to the labial stop frequently heard in the intervocalic /ɪ-i/ context than previously. As before, dental-alveolar stops were frequency heard in /ɪ-ɑ/. These results reflect perceptual integration over the arresting and releasing transition and silent interval duration cues. [Haskins Laboratories provided valuable assistance in the preparation of stimuli on NIGHD Contract No. NIH-71-2420.]
Voiced source extraction from an envelope of residual signals using LPC analysis
Authors: Mano Atsushi , Ozawa Shinji
Published: October 2005

This paper shows a new method to extract voiced source information to construct a high quality vocoder. As voiced source information, we can directly extract peak points and amplitudes of pitch pulses by estimating the envelope characteristic of the amplitudes of the time-domain residual signals. Then we create the following algorithm adapting LPC analysis. First, the squared time-domain residual signals are regarded as the “pseudo-frequency-domain power spectra.” Next, these signals are symmetrized and the inverse FFT is performed. Through these computations, the autocorrelation coefficients of the “pseudo-frequency-domain power spectra” are obtained. Then, LPC analysis is performed on these coefficients, and linear prediction coefficients ai are obtained. Next, the envelope signals of the magnitude spectra are computed by performing an FFT on the ai. Finally, the envelope characteristic of the time-domain residual signals regarded as the power spectra is obtained. From these envelope signals, we can very easily extract exact peak points and amplitudes of pitch pulses during voiced segments. By using this method, a little change of pitch pulses during transition segments between voiced and unvoiced segments can be exactly extracted. We can synthesize a high quality voice using a PARCOR vocoder.
Synthesis by rule of Japanese
Authors: Klatt Dennis , Aoki Chieko
Published: October 2005

Based on an analysis of a corpus of recorded CVCV syllables and several paragraphs [C. Aoki, D. Klatt, and H. Kawasaki, “Acoustic-Phonetic Analysis of Japanese,” J. Acoust. Soc. Am. Suppl. 1 75, S60 (1984)], as well as a study of the prior literature on analysis and synthesis of Japanese, we have formulated a set of synthesis rules within the general framework used in DECtalk. Input to the system must be specified phonemically. The program is divided into several subprograms that (1) parse this string into phoneroes and associated structural/accent features, (2) apply phonological rules to select appropriate allophones or delete segments, (3) assign a duration to each segment, (4) specify onset times and strength of pitch rises and falls, (5) compute 17 time functions to control a formant synthesizer on the basis of stored tables of phonetic target values and smoothing time constants, and, finally, (6) compute a waveform from the control parameter specification. The program has been optimized by systematic spectral/waveform comparisons between the synthetic output and recordings of a selected model speaker. The oral presentation will emphasize differences between the English and Japanese synthesis systems. A demonstration will be played. [Work supported in part by an NIH grant.]
Enhancement of the nonlinear scattering of crossed sound beams in the presence of turbulence due to a concentration of bubbles in the interaction region
Authors: Korman Murray S.
Published: October 2005

Experiments [M. S. Korman, Ph.D. thesis, Brown University (1981)] involving the interaction of mutually perpendicular crossed ultrasonic beams (f1 = 2.05, f2 = 1.95 MHz) interacting in the presence of turbulence in water show that a radiated sum frequency component is detected (outside the interaction region) only in the presence of turbulence. Lighthill's equation, using a cubic quadrupole source term, ρuiuj, describes the scattering. Experimental results for Doppler shift and spectral broadening versus angle are explained by using theoretical results of Kraichnan and Ishimaru for turbulence. However, predictions for intensity versus angle are only in fair agreement with theory. The detection of radiation at subharmonics (fs1 = f1/2, fs2 = f2/2, 3 fs1, 3fs2) shows evidence of small air bubbles in the turbulent flow. Aerodynamic sound generation by turbulent water with air bubbles can be represented by an equivalent distribution of monopole, dipole, and quadrupole sources [D. G. Crighton and J. E. Ffowes Williams, J. Fluid Mech. 36(3), 585–603 (1969)]. Small concentrations allow for a considerable increase in acoustic power output. Source terms involving two primary waves, turbulence. and bubble concentration β are investigated. Dominant equivalent source mechanisms responsible for “enhanced” nonlinear scattering are discussed. [Work supported by NRL (Physical Acoustics Branch) and USNA.]
Mie scattering as a technique for the sizing of air bubbles
Authors: Hansen Gary M.
Published: October 2005

A technique has been developed which allows the average radius of a bubble, in a fluid subjected to a standing wave acoustic field, to be accurately measured without disturbing the bubble. Results are presented which show that radius measurements of single air bubbles in water can be achieved by shining a He-Ne laser on the bubble and measuring the light scattered at 80 deg. Light intensity measurements at a scattering angle of 55 deg were used to calibrate a photodiode detection system and to demonstrate the correlation of data with Mie scattering theory. The photodiode was then moved to 80 deg for actual radius measurements. This technique was accurate to within 3% for bubbles with radii less than 80μ. [Work supported in part by ONR.]
Measured wall pressure cross spectra due to a turbulent boundary layer
Authors: Narayan Nilabh , Plunkett Robert
Published: October 2005

Two-point wall pressure cross spectra were measured in a 190 × 190 mm water tunnel with a free stream velocity of 10 m/s. The frequency range was 0 to 2.5 kHz, the separation was varied axially over a range of 4 to 180 mm and transversely from 4 to 50 min. It appeared to be possible to separate the levels of the convective field, the direct grazing incidence acoustic field and the reverberent acoustic field by comparing various cross spectra. The direct pressure field decays much more rapidly in the transverse direction than it does in the axial one. The grazing incidence acoustic field decays slowly at about the same rate axially and transversely. The reverberent field is constant in the test section and at least 10 dB lower than the grazing field for this facility. The convective phase velocity decreased monotonically with frequency from 100 Hz to 2.5 kHz. [Work supported by ONR, SRO program in Hydroacoustics.]
Spectral estimation of the low-wavenumber region of a turbulent boundary layer: Maximum likelihood method
Authors: Wakefield Gregory H. , Kaveh M.
Published: October 2005

The large dynamic range exhibited by the frequency-wavenumber spectrum of a turbulent boundary layer (TBL) poses a serious problem for estimating the low-wavenumber region of that spectrum. From the standpoint of sidelobe leakage in beamforming procedures, the convective and sonic regions of the TBL are strong sources of interference which bias the estimate at low wavenumbers. By increasing the number of sensors in the measurement array, the sidelobe-leakage problem can be minimized. However, this solution is difficult to implement experimentally and to justify theoretically in view of the globally nonhomogeneous nature of the TBL. As an alternative, certain data adaptive procedures can be used which do not have similar problems with sidelobe leakage when the number of sensors is small, but do require that the frequency-wavenumber spectrum fall within a specified class of spectra. The performance of one such procedure, the maximum likelihood method (MLM), was evaluated under two criteria: spectral fidelity and sensitivity to sampling error. In general, the MLM was superior to the array beamformer in estimating the frequency-wavenumber spectrum based on simulated observations from a linear array of sensors. [Sponsored by the Office of Naval Research Selected Research Opportunity in Hydroacoustics.]
Comparative examination of tip vortex and surface cavitation noise
Authors: Higuchi Hiroshi , Forbes Scott , Arndt Roger
Published: October 2005

Ship propeller noise is a problem of current interest. One of the major sound sources is due to tip vortex cavitation. This phenomenon is currently under investigation utilizing elliptically loaded hydrofoils placed in a 190 × 190-mm high-speed water tunnel. Results from two different hydrofoils were compared: one with a symmetrical (NACA 66-012) cross section and another with a cambered (NACA 66-415) cross section. As predicted, surface cavitation would occur without tip vortex cavitation on the former and tip vortex cavitation could be obtained without surface cavitation on the latter. This permits a quantitative comparison of surface and tip vortex cavitation noise under controlled conditions. The test environment is highly reverberant. Because of this, the acoustic signature of an idealized cavitation noise source in the water tunnel was modeled on the computer. After some experimentation with positioning the sound source in the computer model, the actual time traces of individual bubble implosions in a reverberant field could be closely reproduced. Utilizing an auto-regressive smoothing routine, the spectrum of the computer generated signal with reverberation was identical to the measured data except for a proportionality constant. This important finding lends support to acoustic research in water tunnels. [Work supported by ONR, SRO program in Hydroacoustics.]
Turbulent pressure fluctuation measurements with conventional piezoelectric and miniature piezoresistive transducers
Authors: Galib Thomas A. , Zadina Anthony
Published: October 2005

Turbulent boundary layer pressure fluctuation measurements for longitudinal flow over a cylinder were made with piezoelectric and piezoresistive pressure transducers. The smaller piezoresistive transducer yielded results which did not require correction for transducer size. This data compared favorably with classic results, to which Corcos' corrections were applied. The data from the piezoelectric transducers, when corrected as proposed by Corcos, showed excellent agreement with classic results. In addition, corrected piezoelectric transducer data collapsed to a single nondimensional spectral density curve for a length of Reynolds number range of 9.6 × 106 to 1.27 × 107. The spatial frequency response of the piezoresistive transducer was far superior to the piezoelectric ones (fδ */U∞ = 0.93 vs 0.3). In addition, much less attenuation of the low-frequency portion of the spectrum was realized with the piezoresistive transducer. Nondimensional rms pressure fluctuation levels, and convection velocities determined from the data, also show good agreement with previous results.
Measurements of the acoustic field of a turbulent vortex ring
Authors: Clark Joseph A.
Published: October 2005

Turbulent vortex rings are one of the few examples of nearly stable flow structures in high Reynold's number flows. An investigation of their pressure and flow field characteristics could contribute to our understanding of the processes that couple (dilitational) acoustic fields and rotational flow fields. For this study an electromagnetic method for generating vortex rings of highly repeatable form was developed. The propagating rings were visualized by means of laser sheet illumination and aerosol seeding of the flow. The velocity field associated with a vortex ring was measured by multiple pulse modulation of the illuminating laser beam and direct photographing of the observed particle motions. The pressure field associated with the ring was measured with an array of membranes. Numerical data processing methods were employed to distinguish the acoustic component of the total flow field at various distances from the turbulent core of the vortex ring. Some possible implications of the results for low-wavenumber components of turbulent boundary layer flow will also be discussed. [Research sponsored by ONR.]
An introduction to accreditation and certification
Authors: Peppin Richard J.
Published: October 2005

In a brief overview, this paper discusses the basics of accreditation and certification. Since either implies the acceptance or approval of a laboratory, a service, or a product, by some agency according to some criteria, the entire procedure and credibility rests with the adequacy of the approval process. In addition, the accreditation procedure requires a lack of bias, and a reasonable sum of money for application, and other fees. These and other factors will be presented.
The multifacets of laboratory accreditation at the Riverbank Acoustical Laboratories
Authors: Kopec John W.
Published: October 2005

In 1983 Riverbank Acoustical Laboratories (RAL™) became the first independent acoustical laboratory to be accredited by the National Bureau of Standards, National Voluntary Laboratory Accreditation Program (NVLAP). The various aspects centered around time, labor, and expenses as compared to the advantages of laboratory accreditation will be discussed.
Acoustical Laboratory Accreditation: An independent laboratory's viewpoint
Authors: Guernsey R. M.
Published: October 2005

The accreditation of laboratories is a means to establish minimum acceptance limits by a purchaser of services. As desirable as this may seem there are many laboratories that do not participate in the program. This paper will review one laboratory's status and future plans concerning NVLAP accreditation.
Laboratory accreditation: An acoustical consultant's viewpoint
Authors: Lubman David
Published: October 2005

This paper will present the views of one consultant, who also heads the Laboratory Accreditation Committee of the National Council of Acoustical Consultants, towards the U.S. National Voluntary Laboratory Accreditation Program. While this program can provide significant benefits to the professional community, it unfortunately has some drawbacks that make it almost prohibitive for the involvement of small consulting businesses.
Industry's needs for laboratory accreditation and standards traceability
Authors: Lang William W.
Published: October 2005

Companies which export to Western Europe and other sensitive marketing areas are encountering an increasing demand for publication and verification of the noise levels of their products. Smaller firms are likely to engage independent testing laboratories or consulting firms for assistance. Larger firms tend to have their own acoustical facilities (and staffs). The principal need is for noise level data obtained according to the appropriate international (and national) standards which are suitable for inclusion in product literature. To satisfy this need, the manufacturer (or independent testing laboratory) must ensure that the test facility to be used is qualified for measurements according to the test method selected, must consider the precision and accuracy expectations of the method selected, and must ensure that the test results are traceable to national standards laboratories. The sample size selected by the manufacturer must be sufficient to meet the reporting requirements that have been established.
Acoustic reflectivity of the human middle ear at high frequencies
Authors: Stinson Michael R.
Published: October 2005

A key parameter for network modeling of the human middle ear is the acoustic energy reflection coefficient, or reflectance, at the eardrum. At high frequencies (above 5 kHz) the reflectance is the only unambiguous indicator of acoustic behavior. Using a technique described recently [J. Acoust. Soc. Am. Suppl. 1 75, S11 (1984)] eardrum reflectance can be determined by measuring the spatial variation of acoustic phase along the ear canal. The main advantage of this technique is the small amount of space required for the measurements, so only shallow penetration of the ear canal by a probe is required. One problem, at times, with the preliminary work was involuntary movement of a subject's head. This has been largely overcome with the use of an improved head brace and microphone manipulator. This new system will be described and experimental results presented for several subjects. As with the earlier study, high eardrum reflectances, between 65% and 85%, are found above 10 kHz.
Mechanical, microphonic, and neural responses to low-frequency tones in the base of the chinchilla cochlea
Authors: Ruggero M. A. , Robles L. , Rich N. C.
Published: October 2005

Low-frequency stimuli (30–1000 Hz tones) have been used to correlate the mechanical responses of the 8–9 kHz place of the chinchilla basilar membrane with the cochlear microphonics recorded at the round window and the responses of auditory nerve fibers with appropriate characteristic frequency. At the lowest frequencies maximum displacement of the basilar membrane toward scala tympani occurs in near synchrony with maximum rarefaction at the eardrum; at higher frequencies the response phases gradually lag rarefaction, reaching −240 deg at 1000 Hz. At most frequencies (30–1000 Hz) near-threshold neural responses, once corrected for neural travel time and synaptic delays, are synchronous with maximal scala tympani displacement and with maximal negativity of the round window microphonics; at some frequencies (60–150 Hz) neural excitation appears to lead both mechanical vibration and microphonics by as much as 60 deg. These results confirm our earlier conclusions [M. A. Ruggero and N. C. Rich, J. Acoust. Soc. Am. 73, 2096–2108 (1983)] which were based on the inference of mechanical responses from cochlear microphonic data. [Work supported by NINCDS and NSF.]
Enhancement of pitch-relevant discharge synchronization to two-component signals in single cochlear nerve fibers of the cat
Authors: Greenberg Steven , Geisler C. Daniel , Deng L.
Published: October 2005

Most low-frequency (<4 kHz) cochlear nerve fibers whose characteristic frequencies lie between the primaries of a two (isoamplitude)-component signal synchronize to a half-wave rectified version of the input waveform [S. Greenberg, C. Daniel Geisler, and L. Deng, J. Acoust. Soc. Am. Suppl. 1 75, S20 (1984)]. The autocorrelation of these fibers' timing patterns contain maxima at intervals corresponding to the signal's low pitch and additional maxima at intervals pertaining to the signal spectrum and envelope modulation pattern of discharge activity. In contrast, 25%–30% of low-frequency cochlear fibers synchronize to a restricted portion of the input waveform's modulation period. Discharge activity is either absent or significantly depressed during much of the modulation period even when the signal level is 20–30 dB above rate threshold. This pattern of discharge depression has the effect of emphasizing those maxima in the autocorrelation corresponding to the low pitch of the signal. The spontaneous discharge rate of fibers exhibiting this pattern of discharge depression is invariably lower than 10 spikes/s. Thus, low and medium spontaneous rate fibers may play a special role in the processing of pitch in the cochlear nerve. [Research supported by NIH.]
Spatio-temporal patterns of speech evoked activity in the auditory nerve
Authors: Shamma Shihab
Published: October 2005

We examine the speech evoked responses of large populations of auditory-nerve fibers in the cat [M. Miller and M. Sachs, J. Acoust. Soc. Am. Suppl. 1 70, S9 (1981)]. Three-dimensional pictures of the spatio-temporal patterns of activity are constructed by spatially ordering the display of the fibers' PST response histograms according to the CF of each fiber. The distribution of the relative phases of synchronized activity emerges as an important feature in the response patterns. Specifically, each strong low-order harmonic component of the stimulus tends to synchronize the activity of a relatively broad segment of fibers in a pattern which mirrors the underlying traveling wave. The sphere of influence of each such harmonic can be roughly subdivided into two regions: (1) a region basal to the point of resonance where the fiber PST histograms accumulate only small delays (or phase shifts) relative to each other reflecting the fast speed of propagation of the traveling wave. (2) A region at or very near the point of resonance where the responses exhibit drastic relative phase shifts owing to the sudden slow down of the traveling wave and the consequent rapid acumulation of phase shifts. These rapid phase shifts thus appear to disrupt an otherwise spatially uniform response pattern at approximately the CFs corresponding to the low harmonic components (< 1.5–2 kHz) and thus may play a helpful role in the processing and extraction of the speech signal parameters by the central auditory system.
A progressive time domain wave equation for nonlinear acoustic
Authors: McDonald B. Edward , Kuperman W. A.
Published: October 2005

A nonlinear time domain counterpart of the linear frequency domain parabolic wave equation (PE) has been derived for investigation of pulse propagation in a refracting medium, including caustics. Assuming nearly unidirectional propagation, equations of hydrodynamics yield a quadratic correction term for the linear second-order wave equation. Perturbation analysis about unidirectional wave propagation yields a first-order nonlinear progressive wave equation (NPE) cast in a wave following frame. This equation explicitly separates terms for the physical processes of refraction, nonlinear steepening, radial spreading, and diffraction. Self-refraction is manifest through a continuous interaction between steepening and diffraction terms. When the wave is taken to be linear and monochromatic, the NPE reduces to the familiar PE, within appropriate assumptions. A new numerical algorithm of the flux correction type has been constructed for integration of the NPE. Successful applications to date include: (1) development of initially smooth pulses into aging N wave shocks, (2) wideband linear pulse propagation in a slab, and (3) the evolution of a nonlinear N wave incident on a caustic. [Work supported by ONR].
The application of stepwise coupled modes to an ocean waveguide with periodic bottom roughness
Authors: Evans Richard B. , Gilbert Kenneth E.
Published: October 2005

The method of stepwise coupled modes can be used to compute the solution (including backscatter) of the elliptic wave equation for acoustic propagation in a cylindrically symmetric range-dependent ocean. We review the method and extend it to apply to periodic depth variations of the water-sediment interface. The extension, which is based on an eigenvalue decomposition of the generalized propagator matrix for one period, allows efficient computation of the acoustic field in a ocean waveguide of infinite horizontal extent. Numerical examples are given which compare propagation loss in a rough-bottom waveguide to that in a waveguide with a smooth bottom. It is shown that propagation loss is significantly grater for the rough waveguide. The increased loss is attributed to scattering of energy out of the waveguide. [Work supported by NORDA.]
Numerical considerations in ray tracing and ray expansions of the acoustics wavefield
Authors: Brown Michael G.
Published: October 2005

In separable geometries (range-independent sound speed profile, flat bottom) the reduced wave equation may be solved using the WKB technique. The resulting integral expression is a plane wave or ray expansion the acoustic wavefield. In nonseparable geometries the Maslov technique provides an expression of the same form. Use of the stationary phase technique reduces these integral expressions to standard ray theory. We focus our attention on numerical considerations associated with: (1) computing the geometric ray information which appears in these integral wavefield expressions; and (2) evaluating the integrals. Many commonly occurring problems associated with (1) (ray tracing) may be avoided by appropriately parametrizing the model and rewriting the travel time in terms of the phase integral, τ (sometimes referred to as the delay or intercept time). Problems associated with (2) are particularly troublesome when caustics are present. These may be avoided entirely by transforming to the time domain. [Work supported by ONR.]
A new look at acoustic refraction by mesoscale ocean features with Hamiltonian ray tracing
Authors: Georges T. M. , Jones R. M. , Riley J. P.
Published: October 2005

Tomographers use networks of acoustic paths to map mesoscale ocean structure. To know where to deploy sensors and how to interpret their measurements, one must understand how the sound channel and mesoscale features refract sound in three dimensions, and how such refraction affects acoustic measurables. In particular, one would like to know how ray refraction affects acoustic measurables. In particular, one would like to know how ray refraction affects the perturbation assumptions underlying linear inversion techniques. We use a Hamiltonian ray-tracing program called HARPO to compute the refraction by continuous three-dimensional ocean models and display the results in a way that adds insight about refractive effects. By numerically integrating Hamilton's equations through continuous ocean models, we avoid the discontinuities characteristic of models that break the medium into triangular patches. We begin with simple range-independent models of the sound channel, progress through range-dependent models and finally add a realistic model of a mesoscale eddy, including its currents. Two diagrams that have not been widely used before—plots of range versus launch angle and range versus travel time for all ray-loop numbers—are useful ways to display the arrival-time and ray-focusing perturbations caused by changes in ocean structure.
Acoustic characteristics of the voice of teachers for hearing-impaired children
Authors: Arai Shuichi , Ozawa Shinji
Published: October 2005

In general, the hearing-impaired children can recognize the voices of their teachers better than those of normal persons. The purpose of this study is to make clear the difference between the teacher's voice and a normal person's. In this study, we analyze the Japanese vowels of the teacher and normal persons, and compare their acoustic characteristics. The formant frequency (F1, F2, F3) and the power of the formant frequency (P1, P2, P3) are chosen as the phonetic features, and we analyze them by using our new automatic formant extractor. As a result of the experiments, there is no difference between the teacher's voice and the normal person's on both F1 and F3. But the F2 of teacher's /i/ and /e/ is higher than the normal person's. The F2 of the teacher's /u/ is lower, namely the frequency distances between /i/-/u/ and /e/-/o/ of F2 are larger. Both P2 and P3 of teacher's are higher than those of the normal person's. It is shown by this study that misrecognitions of /i/ and /u/ and /e/ and /o/ are decreased among hearing-impaired children by the above-mentioned reason.
The 18th century solution to the keyboard temperament problem
Authors: Jorgensen Owen
Published: October 2005

The interval relationships between the tones of the scale varied in the past. Applying historical temperament systems to performance affects music interpretation. In the 20th century equal tempered system, all semitones are equally spaced; therefore, melodic and harmonic effects in various keys are identical except for being higher or lower in pitch. Well-tempered 18th century semitones are unequally spaced. This results in melodies and harmonies having varying effects when they are transposed. Whereas the 20th century temperament divides the ditonic comma into 12 equal parts, the 18th century temperaments divide the ditonic comma into two to six parts equally or unequally in such a manner that the most used keys produce harmonies close to just intonation while the least used keys produce harmonies that are highly tempered but still musically acceptable. The 18th century tradition determined that major thirds should change in size progressively throughout the circle.
The evaluation of hearing protector performance in the U. K.
Authors: Wheeler Peter D.
Published: October 2005

Physical performance data for a number of earmuffs has been acquired, in accordance with British Standard 6344, Part 1, which was issued early in 1984. The new standard sets out requirements for the materials used in ear muffs in terms of nontoxicity and nonflammability. The headband has to be adjustable to fit more than 90% of the range of head sizes encountered in the population, for width, height, and shell rotation, and there are also requirements for the headband tension and other features concerned with the comfort of the wearer. The remainder of the requirements are concerned with a test sequence in which the attenuation of a sample pair of ear muffs is measured on an artificial head prior to exposure to a vibration test, a drop test, and a headband flexing test representing 1000 fittings of the ear muff. The attenuation of the sample protector is remeasured again after this exposure, and in order to comply with BS 6344 the maximum change in attenuation thus observed must be less than a defined value commensurate with the variation in attenuation likely to occur in real life between individual wearers. The relationship between the new Standard and the BS 5108 REAT test, and test protocols for subjective measurement of attenuation will be discussed.
Effect of hearing protective devices on auditory communication in noise
Authors: Abel Sharon M. , Kunov Hans
Published: October 2005

This paper reviews the literature on the effects of hearing protection on the perception of warning signals and speech in noise. The particular focus is the comparison of results for normal and hearing-impaired listeners. To illustrate the problem the authors describe one of their own experiments on the binaural detection of one-third octave bands of noise superimposed on backgrounds of high-level, steady-state or intermittent noise taped in a mining industry. Variables evaluated include personal noise exposure history, configuration of the audiogram and the wearing of foam insert protectors. With protectors inserted, normal observers show an increase in threshold of 40 dB in quiet, and an advantage of 3 to 5 dB in noise. Those with noise-induced hearing loss are virtually deaf (i.e., detection thresholds in excess of 100 dBA) under all conditions. Configuration of the audiogram and noise exposure history are not significant factors. These effects are limited to the frequency of hearing loss and depend on whether the hearing loss exceeds the background noise. A model of the detection process for these conditions is developed and applied to the data. [Work supported by Ontario Ministry of Labour, Grant No. 113-R.]
Hearing protection utilization: Survey results across the USA
Authors: Royster Larry H. , Royster Julia Doswell
Published: October 2005

Information about the utilization of hearing protection devices (HPDs) has been obtained through on-site surveys at 219 industrial facilities representative of all regions of the USA. The primary objectives of the study were the identification of problems related to HPD use and the development of a reference manual to guide industrial personnel in achieving maximum benefit from HPDs. The data collected concerned the types of HPD utilization problems encountered. HPD selections offered, criteria considered in choosing HPDs, procedures for fitting/issuing/reissuing, and company policies for furnishing HPDs. User problems identified during the surveys include discomfort, attenuation of warning signals or communication, HPD design limitations, canal irritation, external otitis, wax impaction, and misuse or abuse of HPDs. Problem incidence could be reduced significantly by improved procedures for selection, fitting, issuing, and reissuing HPDs. Administrative problems that were observed include the use of inadequate and/or incorrect information by management in making HPD related decisions, the lack of an in-house key individual to coordinate the overall hearing conservation program (HCP), the breakdown of communciation between various professionals involved in the HCP, and excessive reliance on outside assistance for all HCP related activities.
Evaluation of two “active” hearing protective devices
Authors: Maxwell D. W. , Williams C. E. , Robertson R. M. , Thomas G. B.
Published: October 2005

A variety of laboratory measurements was obtained on two commercial “active” hearing protectors to determine their performance effectiveness with respect to cutoff level, speech intelligibility, and sound attenuation. One-third octave band insertion loss measurements obtained on both a manikin and one of the investigators for three noise level conditions (85, 95, and 110 dBA), in both “on” and “off” positions, indicated that the two devices have a lower and more gradual cutoff level for attenuating noise than levels published by their manufacturers. Speech intelligibility measurements (Tri-work Modified Rhyme Test) obtained for eight speech/noise conditions (four noise levels and two speech-to-noise ratios) revealed generally superior performance by one of the devices. Real-ear attenuation measurements obtained in accordance with ANSI S3.19-1974 resulted in noise reduction ratings of 19.1 and 20.4. [Work sponsored by the Naval Medical Research and Development Command.]
An objective procedure for determining the attenuation characteristics of earmuff hearing protectors
Authors: Lesiecki William , Bilger Robert C.
Published: October 2005

The standard method for determining the attenuation characteristics of personal hearing protectors is the real-ear at threshold method [ANSI S3. 19–1974]. A possible limitation on this method is that the maximum levels reached in this procedure are significantly lower than the noise levels in which these hearing protectors are to be used. An objective measurement procedure in which electret microphones (Knowles BT-1759) were placed at the entrance to the occluded ear was used to measure the attenuation characteristics of earmuffs. The subject was seated in a sound field and E-A-R plugs were inserted so that the end of the plug was even with the tragus, and the electret microphones were taped to the E-A-R plugs. Wideband noise, delivered through a single speaker, was presented at 65, 80, 95, and 110 dB SPL for the uncovered ears and then for the covered ears. Measures were obtained on four subjects for two pairs of each of three brands of earmuffs. Attenuations obtained for frequencies below 2000 Hz with the 65-dB noise were significantly less than those measured for higher levels. If this low-level, low-frequency effect is not artifactual, then the NRR's derived with the real-ear-at-threshold method might underestimate attenuations obtained in actual use. This method, based on computer averaged FFTs has proved to be a fast and reliable method of measuring attenuation afforded by earmuffs. [This work was supported by Grant No. NS 17273.]
The wavenumber spectral response of submerged, ribbed plates
Authors: Garrelick Joel M.
Published: October 2005

The wave vector filter (WVF) concept offers a promising new diagnostic tool for analyzing both structure-borne and fluidborne noise fields. For instance, as with intensity measurements, WVF measurements allow one to distinguish between radiated and nearfield, or incompressible, noise components. In this paper some of the issues that arise in implementing the WVF concept, e.g., spatial nonstationarity, are analyzed in the context of a harmonically line driven rib stiffened, submerged, infinite plate. The plate is assumed to be thin and the ribs are modeled as locally reacting. First the response of this spatially infinite system is determined in both the spatial and wavenumber domains for a few representative geometries. And this is interpreted as the output from a “perfect” WVF. By response is meant plating strain or acceleration or near- or farfield pressure. Next these fields are analytically sampled over a finite domain thus introducing the issues of spatial nonstationarity and other finite filter effects. Results are compared for few and many ribs and various structural loss factors. The effects of structural resonances are discussed.
Measuring vibration on plates and shells with two-dimensional arrays
Authors: Chandiramani K. L.
Published: October 2005

Outputs of the elements of a line or planar array are generally phased or beamformed so as to pick out a particular direction or trace-wavenumber-component of wave propagation. When applied to the measurement of a one- or two-dimensional noise field, this conventional beamforming, in essence, measures directly the one- or two-dimensional wavenumber spectrum of the noise field. However, there are situations where the components of a two-dimensional noise field that need to be picked out are not spectrally pure in wavenumber vector. Examples are: (i) With a two-dimensional array of sensors, we may wish to pick out resonant plate vibration at all wavenumber vectors of magnitude kp, and reject nonresonant plate vibration at all other wavenumber vectors. (ii) Modification of the above procedure for measuring resonant vibration on an orthotropic plate, or on a shell, where curvature-induced effects may lead to anisotropic vibration at low frequencies. Examples such as these are discussed.
Measurement of the nearfield intensity of vibrating ribbed plates by underwater nearfield imaging
Authors: Eschenberg K. E. , Hayek S. I. , Wilson G. L.
Published: October 2005

The vibration of a free plate submerged in an acoustic medium is investigated analytically and experimentally. The plate's vibration and the resulting acoustic field were analyzed by finite element method, resulting in the prediction of the resonances and the surface pressure and intensity for three submerged plates: one without ribs, one with one rib, and one with two unequal ribs. In the measurement, the ARL System for Underwater Nearfield Imaging (SUNI) was used to record an acoustic pressure hologram within the nearfield of each plate with the plate being excited by a shaker at specified locations. The recorded hologram was then digitally processed to calculate and display the plate's surface pressure, velocity, and acoustic intensity as well as the acoustic intensity map within a wavelength of the plate's surface. The farfield directivity of the acoustic radiation of the plate was also obtained. [Work supported by the Ship Acoustic Department of the DTNSRDC/Carderock.]
Finite difference nearfield errors for a baffled vibrating piston
Authors: Mielnicka-Pate Anna L.
Published: October 2005

Two-microphone finite difference errors for velocity, intensity, and radiated power were analyzed in the near- and farfield of a vibrating piston in a baffle. Since a closed form solution is not available for nearfield radiation, numerical solutions were used to obtain the sound pressure, velocity, and intensity. Newton's method was used for the numerical integration. The finite difference errors were calculated for two different intensity probe locations. These locations were in a plane parallel to the surface of the piston and on a sphere centered around the piston. The results were analyzed in terms of sound intensity and sound power measurements. In addition, experimental data were measured for comparison purposes.
Network-based isolated digit recognition using vector quantization
Authors: Bush Marcia A. , Kopec Gary E. , Hamilton Marie E.
Published: October 2005

This talk will describe a network-based system for speaker-independent, isolated-digit (one-nine, oh, and zero) recognition and will discuss the results of an extensive series of system tuning and evaluation experiments. The digits are modeled by pronunciation networks whose ares represent classes of acoustic-phonetic segments. Each are is associated with a matcher for rating an input speech interval as an example of the corresponding segment class. The matchers are based on vector quantization of LPC spectra. Recognition involves finding minimum quantization distortion paths through the networks by dynamic programming. The system has been tested using nearly 6000 tokens of speech by 250 talkers, including a subset of a large database developed by Texas Instruments [G. Leonard, Proc. 1984 IEEE ICASSP]. The best recognizer configurations achieved accuracies of 97–99%. Performance over 21 geographically defined talker groups included in the TI database will be discussed.
Perceptually motivated spectral comparisons for automatic speech recognition
Authors: Hieronymus James L. , Gengel Roy W.
Published: October 2005

Early experiments on amplitude discrimination by Flanagan [J. Acoust. Soc. Am. 27, 1223–1225 (1955); J. Speech Hear. Disord. 22, 205–212 (1957)] suggest that second formant amplitude in synthetic vowels may vary ± 4 dB without a noticeable difference. Klatt (ICASSP 82, 1278–1281) conducted experiments changing the spectral tilt of vowels. This changed the “quality” but not the identity of the vowel. These results contrast sharply with spectral comparisons used in automatic speech recognizers—which are often based on Enclidean or Itakura distances. We used a dynamic time warp based speech recognizer to study several perceptually based spectral comparison measures. One of these measures matches the lowest frequency spectral peak in the template within ± 1.5 dB. For each subsequent peak within the template spectrum, each peak in the unknown spectrum which is within a perceptual frequency band is scored as good match if it is within ± 4 dB of the template peak. Another measure tilts the spectrum so that F1 and F2 are approximately equal amplitude and then uses a Enclidean distance measure. The results of these experiments will be reported.
Automatic design of connected speech data bases
Authors: Fisher William M.
Published: October 2005

A computerized procedure to specify the set of utterances to be used in testing syntactically-controlled speech recognizers is presented. The problem is one of constrained optimization. A feasible solution is a set of sentences such that: (1) each sentence is grammatical; (2) some minimum coverage of the speech elements is achieved; and (3) speech recording system constraints are met. The solution algorithm should produce a feasible solution that minimizes the use of resources or maximizes a quality measure such as the variety of phonetic environments in which each speech element occurs. The procedure to be presented adds to an initially null set of sentences until a feasible solution is achieved. At each cycle, a word that has not yet been sufficiently covered by the sentence set is picked and several different sentences including that word are generated. Sentences violating recording system constraints are ignored, and the best of the remaining candidate sentences is added to the sentence set. If some words are still not sufficiently covered, the procedure recycles.