bluuebunny/crossref_metadata_2025 huggingface dataset:
A study on the setting position of vibration pickups for the measurement of hand-arm transmitted vibration
Authors: Tokita Yasuo , Oda Atushi , Ohkuma Tunenobu
Published: October 2005

In order to reduce hand-arm vibration disease, it is important to know the amount of vibration transmitted from a hand-held vibration tool to the hand-arm system. There are many problems in determining the amount of exposure in practice. For example, (1) the handle vibration does not always correspond to the vibration disease. (2) If the vibration pickup is set between the handle and palm, this system will not be usable for a long time, because the vibration pickup will become an obstacle for the worker. In this study, Leq8 (8 h of equivalent hand-arm vibration level according to ISO 5349) is used to evaluate the transmitted vibration, and the back side of the hand is recommended for the placement of the vibration pickup to measure the amount of transmitted vibration without disturbing the worker during the long-time use of the measuring device. The reasons for this proposal are indicated by the results of field measurements of actual hand-held vibration tools (chain saw, breaker, etc.) and by the analytical results of the model experiment. [Work supported by Ministry of Labor of Japan.]
Errors in the measurement of power tool handle vibration from excess accelerometer weight
Authors: Keith S. E. , Brammer A. J.
Published: October 2005

ISO 7505 (Forest Machinery—Chain Saws—Measurement of hand-transmitted vibration. International Oragnization for Standardization, Geneva, 1986) specifies for the measurement of frequency-weighted handle acceleration a maximum permissible accelerometer weight of 50 g, which is comparable to that of most chain saw handles. To explore the magnitude of errors so introduced, a series of difference measurements has been conducted with a variable mass accelerometer mount. With this device the effective accelerometer weight can be cycled between 6 and 25, 50, or 100 g, in a time in which other parameters affecting handle vibration during saw operation with no load (viz: engine speed and hand grip) usually vary little. Reductions in frequency-weighted acceleration from the minimum resolvable (± 5%) to 50% were recorded when changing accelerometer weight from 6 to 50 g, depending on handle design and material. Investigation of the vibration spectra (6–1250 Hz) and transmissibility (30–700 Hz) of one handle suggests that the data are consistent with a localized mass loading, introduced by the accelerometer, reducing the amplitude of one or more flexural modes of handle vibration.
Linear and nonlinear acoustics in the time domain: Long-range pulse propagation
Authors: McDonald B. Edward
Published: October 2005

During recent years the nonlinear progressive wave equation (NPE) model has been developed as a nonlinear time domain counterpart of the linear frequency domain parabolic equation (PE) model [B. E. McDonald and W. A. Kuperman, J. Acoust. Soc. Am. 81, 1406–1417 (1987)]. It was motivated by investigation of nonlinear wave evolution, so that time domain formulation was natural. With nonlinearity absent, the NPE gives an efficient algorithm for linear broadband propagation in the time domain. (No Fourier transforms between frequency and time domains are involved.) Recent simulation results from the NPE model (given in a color movie) illustrate linear and nonlinear pulse propagation through a deep ocean convergence zone. Differences in physics between broadband linear and nonlinear propagation will be pointed out and discussed. Linear results reveal transients whose time scale as a function of range is “remembered” from the source. In corresponding nonlinear results, steepening and shock formation cause the source's time scale to be gradually “forgotten.” For the parameters of the example given, nonlinear spreading behind the shock increases the transients' time scale by roughly a factor of 3 in the farfield.
Simulated autocorrelations of a broadband signal in an ocean with a layered bottom
Authors: Ebbeson Gordon R.
Published: October 2005

It is well documented that the correlations between the multipath arrivals from a broadband acoustic source in a deep ocean can be greatly affected by the bottom sediment. As an aid to explaining these effects, the seismogram version of the SAFARI model was extended to simulate the autocorrelation of the received signal. As with the original model, the bottom environment is input in the form of a geoacoustic bottom model. However, the inputs to the extended model also include the source level, realistic ambient noise and sea state conditions, noise due to distant shipping, and the self-noise levels of the receiving array. Simulations were carried out using the environment of the Tufts Abyssal Plain of the Northeastern Pacific. The geoacoustic bottom model for this area was developed at DREP [N. R. Chapman, J. Acoust. Soc. Am. 73, 1601–1607 (1983)] and has been validated by a detailed analysis of propagation loss measurements. It was found from these simulations that reflections from both the water-sediment and sediment basement interfaces result in a time-spreading of the correlogram peaks as well as a reduction in the overall correlation magnitude.
The effects of time delay quantization on underwater ultrasonic imaging techniques
Authors: Goode A. H. , Reeves J. M. , Harrold S. O.
Published: October 2005

This paper describes different pulse echo imaging techniques based on time delay focused arrays for both transmission and reception. The effects on system performance of time delay quantization are examined using a versatile pulse echo imaging test facility and a corresponding computer simulation. The system currently used has B and C scan capability over a 40° sector. It is designed around 32-element linear-phased arrays, operating at 2 MHz, for both transmission and reception of short bursts of ultrasonic energy. The techniques described fall into two categories, high resolution and low resolution. The high-resolution techniques calculate the sum of the corresponding instantaneous samples of each of the 32 individually received signals, therefore allowing phase cancellation/reinforcement to occur. The low-resolution technique simply utilizes the total energy in the received signal. A brief description of the test facility is included together with the methods for using these types of imaging techniques. The sensitivity effects due to the time delay quantization are discussed for various levels of quantization, and three-dimensional sensitivity contour plots depicting system performance are included.
Biochemical considerations in presbycusis
Authors: Thalmann Isolde , Takahashi Kuniaki , Thalmann Ruediger
Published: October 2005

Virtually no direct information about biochemical changes in the peripheral auditory system due to aging is available. However, numerous predictions about biochemical changes due to characteristic degeneration patterns can be made indirectly. (1) Since glycogen and calmodulin are extremely high in outer hair cells, significant degeneration of these cells due to aging will result in lowered average levels of both substances in whole organ of Corti. Analogous considerations apply if major longitudinal gradients are present (e.g., glycogen, P-creatine, cyclic GMP). (2) Degeneration of stria vascularis, the so-called “metabolic” type of presbycusis, will result in a lowered endolymphatic potential and low endolymph K, while Na and Ca will be elevated. (3) The “inner ear conduction” type of presbycusis is associated with massive degeneration of the spiral ligament (SL). Apart from mechanical effects on basilar membrane motion, this should result in significant changes in perilymph chemistry. The normal SL exhibits extremely high carbonic anhydrase, which is thought to be essential in perilymph regulation. The SL also appears to be responsible for the markedly different kinetics of K and furosemide in scala vestibuli and tympani. The pronounced gradients of these substances between the scalae can no longer be maintained in the absence of a functional SL.
A local cochlear mechanism of endolymph volume regulation
Authors: Thalmann Ruediger , Salt Alec N. , DeMott John
Published: October 2005

It is generally believed that endolymph volume is regulated, not by the cochlea, but by the endolymphatic sac. Acute changes of endolymph volume in anesthetized guinea pigs has been induced by perfusing hypertonic medium (400 mOsm/kg H2O) for 20 min through the perilymphatic spaces. During hypertonic perfusion, changes of endocochlear potential (EP), endolymph K level, and endolymph volume (using tetramethylammonium: TMA as a volume marker) were monitored using ion-selective microelectrodes. The osmotically induced K increase appeared to be comprised of two components. The first, an increase due to the loss of water from endolymph was estimated from the recorded TMA changes. In addition, a second component existed representing a net addition of K to the endolymph. This component was closely correlated with the EP reduction during dehydration, which suggests the endolymph K increase may result from the admission of anions into endolymph. A mechanism that produces an increase in endolymph solutes when endolymph volume is reduced would allow volume changes induced by osmotic treatment to be counteracted. These data suggest that, in addition to the ionic composition of endolymph being locally regulated, endolymph volume may also be locally controlled. [Work supported by NIH and DRF.]
Fractal character of the auditory neural spike train
Authors: Teich M. C.
Published: October 2005

Long-counting-time pulse-number distributions (PNDs) were measured in cat primary auditory fibers using a broad range of fibers, stimuli, counting times T, and number of repetitions NT. Short-counting-time PNDs readily exhibit the existence of spike pairs, whereas PNDs with T≳100 ms exhibit irregular shapes in all cases, indicating the presence of spike clusters in the underlying auditory neural spike train. The count variance-to-mean ratio (or Fano factor) F(T) was relatively constant over a broad range of stimulus levels for all units measured, but increased substantially as T or NT increased. A normalized coincidence rate [g(τ) versus delay time τ], based on three physiologically plausible factors, is in accord with the Fano-factor versus counting-time function. The observed power-law growth for the Fano factor for large T[F(T) ∝ Tα] implies that g(τ) ∝ τα 1 and that the spectral density of the spike train S(f) ∝ f−α, where 0⩽α⩽1. This suggests that the auditory events exhibit fractal behavior for sufficiently large times (sufficiently low frequencies) [M. C. Teich, IEEE Trans. Biomed. Eng., in press (1989)]. This behavior leads to spike clusters in the PND. The fractal dimension for the process is D = α≈12 in the range 0.1 s⩽T⩽10 s. The firings of vestibular fibers do not exhibit fractal behavior, suggesting that the form of the auditory neural-firing pattern serves to effectively sample fractal natural noises. [Work supported by NIH and NSF.]
Contribution of auditory evoked potentials to binaural processing
Authors: McPherson David L. , Pinkston Charles , Starr Arnold
Published: October 2005

The contribution of the early (ABR), middle (MLR), and late (LLR) auditory evoked potentials (AEPs) to binaural interaction (BI) was studied in 17 normal young adults. AEPs were recorded by an electrode at C2 and referenced to a noncephalic electrode at CVII. Filtering was between 10 and 3000 Hz for both the ABR and MLR, and between 1 and 500 Hz for the LLR. Rarefaction clicks with an intensity of 60 dB above threshold for wave V of the ABR were presented at 11.1/s for both the ABR and MLR, and at 1.7/s for the LLR. Two samples of 2000 trials each were averaged for right monaural, left monaural, and binaural stimulus presentation. The binaural interaction component (BIC) was obtained by subtracting the sum of the monaural responses from the binaural response. Maximum BI in the ABR occurred at 7.35 ms with a mean BIC of 23%. Maximum BI occurred in the MLR at 39.6 ms with a mean BI of 49%. Maximum BI of the LLR occurred at 145 ms with a mean BI of 46%. This would indicate that BI is primarily medicated at the time of the generation of the N40 component of the MLR and that binaural processes constitute a major function of the auditory pathway rostral to the pons. [Work supported by NIH Grant ♯NS-11876.]
Time-frequency receptive fields for neurons of the cochlear nuclei
Authors: Clopton Ben M. , Backoff Patricia M.
Published: October 2005

Periodic segments of wideband noise, digitally synthesized, evoked highly replicable patterns in period histograms for spike activity sampled in all subdivisions of the cochlear nuclei of guinea pig. These stimuli were transformed to time-frequency surfaces of energy density, and reverse correlation between the stimulus surface and response histogram used to estimate a weighting surface, relative to the time of spike occurrence, representing a time-frequency receptive field for the unit. A region or regions of elevated energy densities on the receptive-field surface, restricted in time and frequency before spike occurrence, were observed for all units driven by the stimuli and can be interpreted as representing excitatory afferents to the unit. For some units a region or regions of energy density lower than expected were present, presumably due to inhibitory afferents. It is possible, for most units, to predict responses to tone bursts and more complex sounds from these receptive-field surfaces. This approach, an extension of the work of de Boer, Johannesma, and others [Eggermont et al., Q. Rev. Biophys. 16, 341–414 (1983)], does not depend on phaselocking and provides a mathematical base for studying neuronal networks in these nuclei. [Work supported by the NIH and NSF.]
Effects of attenuation, dispersion, and high sound-pressure levels on acoustic wave distortion in horns
Authors: Pla Frederic G. , Reethof Gerhard
Published: October 2005

High-power sound sources have received a lot of attention in the past few years due to renewed interest in industrial applications of high-intensity sounds such as the acoustic agglomeration of aerosols or combustion enhancement. Most high-power sound sources require a horn to match the source impedance to the medium where the sound is radiated. Such horns introduce distortion in the initial waveform, which can be detrimental to the agglomeration or combustion enhancement process. Boundary-layer attenuation smooths the wave shape while dispersion breaks up the symmetry of the waveform. Horn-induced dispersion is usually the dominant dispersion mechanism, resulting in strong peaks in the waveform. Finally, due to the very high acoustic levels at the horn throat, finite-amplitude effects are responsible for a significant amount of distortion at high frequencies. Simple examples of waveform distortion due to these various mechanisms are shown. The effects of sound-pressure level, horn design, and frequency on distortion are illustrated for an exponential horn and several initial wave shapes. Experimental results are presented that compare very well with theory.
Relaxation-based speech labeling
Authors: Katagiri Shigeru
Published: October 2005

It was revealed that a trained human spectrogram reader could perform accurate speech labeling, and that accuracy was based on the flexibility of his/her decision process using many kinds of spectrographic features [S. Katagiri, SP87-115, IEICE Tech. Rep. (1988)]. In this paper, a new flexible speech labeling system that simulates the trained reader capability is proposed. The main task of the system is to apply the trial-and-error process used in a human reader's labeling work. Therefore, a relaxation method is adopted here [S. Katagiri, 2-1-19, A. S. J. Spring Meeting (1988)]. The system consists of three parts: an acoustic analyzer, a verifier, and a supervisor. In the acoustic analyzer, many kinds of acoustic feature candidates, e.g., formant and pitch frequencies, are calculated. In the verifier, possible speech labels are verified. The supervisor, with a behavior principle based on the relaxation method, controls the whole system. Experimental results show that the system's performance is comparable to a human reader's performance and that very accurate labels are automatically created.
Hidden Markov models for Spanish stops
Authors: Franco Horacio
Published: October 2005

In this work, the design and evaluation of the recognition performance of hidden Markov models (HMM) for the intervocalic Spanish stops are described. The proposed HMMs were context dependent in order to account for coarticulatory effects. Continuous probability density functions were used for the output vectors, which included spectral dynamic parameters. Initial model parameter estimates were obtained by means of an ad hoc segmentation procedure for careful modeling of spectral transitions. The speech database consisted of a total of 2592 productions of the Spanish stops/p, t, k, b, d, g/ in intervocalic position with the vowels /a, i, u/ in all combinations. The speech data were produced by two male Argentine speakers. As a step to evaluate the recognition performance with running speech, three different consonants were produced embedded in VCVCVCV nonsense utterances. Results were obtained from variations of the basic models that consisted of a comparative study of the recognition performance under different degrees of context dependence, and the alternative use of spectral transition parameters. In the best condition the average consonant recognition performance, obtained in a speaker-dependent manner, was 96.61% and 96.53% for each speaker. The errors were concentrated mainly in the voiced stops, with almost perfect recognition obtained for the unvoiced stops.
Word spotting method based on HMM phoneme recognition
Authors: Kawabata Takeshi , Hanazawa Toshiyuki , Shikano Kiyohiro
Published: October 2005

A new technique for detecting and locating key words in continuous speech using (hidden Markov model) (HMM) phoneme recognition is proposed. HMM word models are composed of HMM phone models trained on an isolated word database. Because the speaking rates for isolated words and continuous speech are different, phoneme spectra and durations change considerably. An HMM consists of several states and arcs. Each arc has output probabilities for each VQ code. In order to cope with the spectral changes, the output probabilities are smoothed with the probabilities of their spectral neighbor codes. In order to cope with the duration changes, HMM state duration parameters are shifted according to a second-order duration calibration curve. The calibration curve is obtained from a speaking rate ratio of continuous speech to isolated words. The word detection rate for 8 key words in 25 sentences uttered by one speaker was 98.4%. Accurate word spotting is accomplished using the HMM output probability smoothing technique and the state duration control mechanism taking the speaking rate into account.
Estimating the acoustic characteristics of speech from visual speech signals
Authors: Yuhas Ben P.
Published: October 2005

Speech articulation produces acoustic and visual signals. When the acoustic signal is degraded by noise, the visual signal can provide compensatory information [W. H. Sumby and I. Pollack, J. Acoust. Soc. Am. 26, 212–215 (1954)]. The aim of the work to be presented is to explore the extent to which the transfer function of the vocal tract filter can be estimated from the visual speech signal. The approach was to use a neural network to obtain the short-term power spectral envelope of the acoustic signal given the corresponding visual image as input. The visual images were taken from laser disc recordings of speaker's faces [Bernstein et al., J. Acoust. Soc. Am. Suppl. 1 82, S22 (1987)]. A small box centered at the mouth was extracted and spatially subsampled. The output of the neural net was the acoustic spectral envelope of the corresponding acoustic signal obtained via the cepstral method. After being trained on 95 tokens of vowels and dipthongs, the network was tested on 24 images it had not seen. The net was able to estimate the spectral envelope associated with these images more accurately than more traditional approaches that used stored templates. [Work supported by AFOSR Contract No. 86-0246.]
NNet: A natural voice neural network speech synthesizer
Authors: Villarreal James A. , McIntire Gary , Anderson Claire W. , Anderson Wallace L.
Published: October 2005

Reproduced speech extracted from actual voice recordings of a few selected individuals representing a wide variance of age, speech dialect (English), and sex was used as input values to a four-layer (input, hidden, output, and state) generalized backward propagation artificial neural network system (ANS) using a linear prediction coding (LPC) model to produce more natural sounding artificial speech. The speech phonetic research environment (SPIRE) MIT-developed software package running on a Symbolics computer was utilized in making the original voice recordings. The linear predictive analysis of the LPC model efficiently represented the speech signals in terms of slowly varying parameters. It converted the combined spectral contributions of the glottal flow within a pitch period, the vocal tract, and the radiation of the lips into a single recursive (all-pole) time-varying filter. The transfer function of the filter involved the use of gain coefficients that, when fed into the system, were found to destabilize it. The LPC model was modified to use impulse characteristics of the system rather than the individual discrete coefficients. Preliminary data indicated close output to input matches, artificial speech to natural speech, indicating the ability of the artificial neural network to learn the timing, pitch fluctuations, and connectivity between individual sounds, and the speaking habits unique to a person.
Calculation of sound radiation from an aperture of a building by means of Kirchhoff's diffraction formula
Authors: Hara Nobuo , Furue Yoshihiro
Published: October 2005

When a noise source is located in a very large room as compared to the wavelength concerned, the integral equation method to calculate the sound field is not applicable because of limitations in computer capacity. In this case, an approximate approach based on Kirchhoff's diffraction formula becomes more practical. This formula requires the velocity potential and its normal derivative at the aperture, which are unknown. These unknowns may be assumed to consist of the direct wave from the source and the diffused ones. The latter are assumed to be plane waves with the same amplitude but random phase. The relation between the amplitude of the direct wave and the reflected ones is determined by geometrical acoustics. Thus the radiated sound field can be readily evaluated from the aperture by Kirchhoff's formula. The numerical results for a scale model of a reverberation chamber with a rectangular aperture are shown with measurements. The agreement between the calculations and the measurements is satisfactory.
Fiber-optic sensors for vibro-acoustic measurements
Authors: Garrett S. L.
Published: October 2005

Optical fiber sensors of sound and vibration offer potential advantages over conventional (electronic) sensing techniques in applications where there is strong electromagnetic interference, large sensor/receiver separations, or restrictive (remote) power requirements. The rapid acceptance of optical fibers as a communications medium over the past decade has stimulated much research in fiber-optic sensing but has generated very few commercially available vibro-acoustic sensors of the intensity [J. W. Berthold, “Overview of fiber-optic intensity sensors for industry,” in Fiber Optic and Laser Sensors V, Proc. SPIE 838, 2–8 (1987)] or the interferometric [C. M. Davis, “Fiber optic sensors: An overview,” Opt. Eng. 24(2), 347–351 (1985)] types. This presentation will review the basic fiber-optic sensor types (amplitude/phase and intrinsic/extrinsic), present examples of microphone, hydrophone, and accelerometer designs, and speculate on the factors that will influence their commercialization over the next decade. Issues of minimum detectable signals, electro-optic demodulation, and multisensor array multiplexing will be addressed as time permits.
Piezoelectric acoustic transducers for electronic telephone sets
Authors: Ohga Juro
Published: October 2005

The development of piezoelectric ceramic transducers for the transmitter, receiver, and tone-ringer sounder of recent electronic telephone sets is reviewed. As the first development stage, the working attenuation (WA) was studied as a measure to compare efficiencies of various types of electroacoustic transducers. The conclusions were that WA for electroacoustic transducers, which was defined in a similar way to WA for electrical circuits, could be estimated easily from the specific response for a microphone, and that WA was almost independent from transducer size. Then, it was shown by comparison of WAs that the efficiency of the simplest piezoelectric ceramic transducer using a unimorph diaphragm was as high as that of ordinary electromagnetic or electrodynamic ones. Moreover, a piezoelectric transmitter and receiver needed the same resonant frequency of the diaphragm, because both diaphragms should work in the stiffness-controlled region. Fortunately, their optimum resonant frequency was close to that of the tone-ringer sounder. Therefore, the main problem of the second development stage was how to design acoustical circuits for the piezoelectric transmitter, receiver, and tone-ringer sounder using the same diaphragms. This was solved by introducing a circuit with four degrees of freedom. As a related problem, simplification of transducer structure was also studied.
Pulse discharge sound source and its application to acoustic measurement
Authors: Shibayama Hideo , Kido Ken'iti
Published: October 2005

This paper describes a pulse discharge sound source without any diaphragm, and its application for acoustic measurements. The waveform of the sound pulses radiated from this source is simple and reproducible. The power spectra of these pulses have no zero points up to 90 kHz. The duration of the sound pulse is about 50 uμs, and is dependent on the gap length between the discharge electrodes. The directivity function of the sound source is the same as that of a line source that is equal to the gap length. The sound generating spot is small, and the sound source requires no baffle board. As a result, no reflection from the sound source itself is produced. The timing of sound generation is stable and controllable. As examples of acoustic measurements utilizing the sound source, the estimated results of acoustic impedance characteristics of materials and their frequency responses of diffraction are shown. The estimated results agreed well with the calculated ones. A measuring system can be considerably simplified by use of the pulse discharge sound source and the digital signal processing technique. Measuring time can also be reduced.
Review of scanning laser acoustic microscopy—Imaging applications
Authors: Kessler Lawrence W.
Published: October 2005

The scanning laser acoustic microscope (SLAM) was first developed in the early 1970s as a laboratory curiosity to make high-resolution images of materials in the expectation that microstructure could be studied in a new way. Since that time the techniques have developed into practical and commercially available instruments that have found their way into industry by solving very important problems in quality control. Stimulated by the evolution of new materials and processes, for example, fine ceramics, composites, and the ever-changing field of microelectronics, SLAM has often played an important role as an analytical tool, alongside other analytical instruments to “see inside” products nondestructively, in true real-time and with higher resolution than had been possible with conventional ultrasonic imaging.
Recent progress in quantitative measurements by acoustic microscopy
Authors: Kushibiki Jun-ichi
Published: October 2005

The line-focus-beam (LFB) acoustic microscopy developed at Tohoku University in 1981 is expected to provide a new technology for quantitative material analyses in scientific and industrial fields. The material characterization method through the so-called V(z) curve analysis can measure the acoustic properties of materials, viz., velocity and attenuation, of leaky surface acoustic waves (LSAWs) excited on the boundary between samples and its reference coupling liquid of distilled water. The most remarkable features of the method are the following: the proper anisotropy detection of the material properties, very high measurement accuracy, and nondestructive and noncontracting evaluation. In this paper, some of the promising applications of the acoustic microscopy will be reviewed concerning the following subjects: the characterization of the acoustic anisotropy and inhomogeneity of materials; the structural analysis of polycrystalline materials; measurement of film thickness; and determination of the elastic constants of bulk and thin-film materials. Present and future problems will also be discussed.
Application of variational principles to the evaluation of axisymmetric surface pressure and displacement along a harmonically vibrating elastic shell
Authors: Chen Pei-Tai , Ginsberg Jerry H.
Published: October 2005

An earlier paper [J. H. Ginsberg and P. T. Chen, J. Acoust. Soc. Am. Suppl. 1 82, S1 (1987)] employed assumed modes to describe the displacement and pressure along an elastic plate. The amplitudes of those modes were obtained by simultaneously satisfying Hamilton's principle for the structure and a variational principle for the pressure distribution along a vibrating body. Here, the method is extended to an elastic shell structure in the form of an arbitrary body of revolution. The energy expressions for the shell are based on Love's assumptions, while the stationary quantity for the surface pressure is derived from the Kirchhoff-Helmholtz principle. Two important facets of the derivation are the treatment of singularities in the latter and the reciprocal nature of the coupling between the pressure and surface displacement. A numerical example compares the results obtained from the variational formulation with Hayek's analytical solution [J. Acoust. Soc. Am. 40, 342–348 (1966)] for the radiation of a spherical shell in an acoustic medium under a harmonic, concentrated force. [Work supported by the Office of Naval Research, Code 1132-F.]
Nondestructive measurement of texture and nine elastic constants of rolled metal sheets by EMAT
Authors: Kawashima Katsuhiro
Published: October 2005

All the lowest-order independent coefficients W400, W420, and W440 for the crystallographic orientation distribution function were obtained nondestructively for thin rolled steel sheets by measuring ultrasonic wave speeds and making use of the known values of the single-crystal elastic constants. All the ultrasonic wave speeds were measured by EMAT so that no acoustic coupling medium was necessary, and the measurements were done quickly and with very good reproducibility. Pole figures were calculated using the obtained coefficients W400, W420, and W440 and the results were quite similar to the pole figures obtained by the x-ray diffraction method. It is well known that a rolled steel sheet can be modeled as an orthorhombic continuum having nine different elastic constants. It is also known that only six of the nine constants are independent when the sheet is a polycrystal of cubic crystallites and assuming that the macroscopic properties can be predicted by an averaging scheme. All these nine constants were calculated using the obtained W400, W420, W440, and the known single-crystal elastic constants. Young's moduli in the plane of the rolled steel sheet were also calculated and compared with the measured plastic strain ratio. Good relations were obtained between the average Young's modulus E and the average plastic strain ratio r.
Electromagnetically generated acoustic determination of delamination
Authors: Imaino Wayne
Published: October 2005

Previous work has demonstrated a technique for acoustically detecting localized delamination of metallized patterns on insulating substrates [W. Imaino et al., in Proceedings of the IEEE Ultrasonics Symposium, edited by B. R. McAvoy (IEEE, New York, 1986), p. 1065]. Employing a high-spatial resolution permeable EMAT, to preferentially excite the metallization, the acoustic coupling between the metal foil and substrate may be probed. Scanning, then provides a map of delaminations. To extend and generalize these results, the detailed generation mechanism and acoustic response of the substrate has been studied. A computer program developed previously, [W. Imaino, J. Acoust. Soc. Am. Suppl. 1 80, S7 (1986)] is used to model the acoustic source and a finite element calculation provides a detailed description of the acoustic behavior. The effect of source size relative to defect dimensions and acoustic properties of the substrate have been studied. The determination of the localized coupling is complicated by structural resonances of the substrate, which make single frequency measurements unfavorable. However, the analysis shows that the spectral acoustic behavior provides an indication of metallization to substrate coupling. A signal processing algorithm based on this analysis has been formulated and will be described.
Ultrasonic properties in excessively fatty rat livers
Authors: O'Brien William D. , Greenleaf James F. , Erdman John W.
Published: October 2005

The orotic acid dietary “liver lipid” model has successfully been used to study ultrasonic properties in excessively fatty rat livers [O'Brien et al., J. Acoust. Soc. Am. 83, 1159–1166 (1988)]. The model was used again in order to evaluate more fully ultrasound properties within a more condensed (2%–10%) lipid concentration range. The fatty livers were first evaluated in the intact animal (in situ) with an ultrasonic imaging system (≅ 5 MHz) and then the excised (in vitro) livers were evaluated over the 1- to 100-MHz frequency range. Water, lipid, total protein, and collagen were evaluated for each liver. Significant (p < 0.001) correlations of ultrasonic properties (absorption, all attenuation, speed, and Markovian entropy) against tissue properties (water and lipid) were obtained, thus suggesting that the independent measure of texture may relate to basic ultrasonic propagation properties. [Work supported by NIH Grant CS36O29.]
Low-frequency search unit for ultrasonic testing of concrete
Authors: Imoto Kanji , Ohta Koji , Watanabe Takeharu
Published: October 2005

A new low-frequency search unit for ultrasonic testing has been developed. A conventional low-frequency search unit involves many high-harmonic components in its signal. This is one of the reasons why it is difficult to test concrete by the ultrasonic pulse echo method. Characteristics of a search unit have been analyzed by a computer. This analysis was based on the assumption that each of the components of the search unit, such as the piezoelectric transducer and the resonator will form a four-port terminal network. The design criteria for a search unit have been determined by this analysis. The model for analyzing the search unit characteristics and the results of analysis are shown. A low-frequency search unit has also been produced. The characteristics of this search unit are compared with analytical results. Ultrasonic testing of concrete has been improved by using this search unit.
High-fidelity hearing aid amplifier
Authors: Killion Mead C.
Published: October 2005

If the claims of many mildly hearing-impaired individuals that they “do not need a hearing aid most of the time” are true, the appropriate hearing aid design appears self-evident: the hearing aid should be designed to do nothing (be acoustically transparent) most of the time. A low-power-consumption bipolar linear (!) custom integrated circuit hearing aid amplifier has been designed to permit that result while providing the gain and high-frequency emphasis usually required to make quiet sounds audible. The goal is to amplify only the quiet sounds the wearer is missing, while neither standing in the way of his apparently adequate hearing for high-level sounds, nor—worse yet—amplifying sounds that are already loud enough. The resulting level-dependent frequency response characteristic is opposite that of the presently popular “ASP” circuits, which appear to be solving a problem created by the typical hearing aid design. [Work supported in part by NIA.]
Masked thresholds for hearing-impaired listeners
Authors: Formby C.
Published: October 2005

Masked thresholds for sinusoids between 1000–500 Hz in (1) broadband noise, (2) low-pass filtered noise, and (3) high-pass filtered noise were measured by adaptive forced-choice procedure for hearing-impaired listeners [acute Meniere disease (n = 3), noise-induced hearing loss (n = 7), and eighth-nerve tumor (n = 2)]. Thesholds in the filtered noises were measured as a function of masker cutoff frequency. Masker spectrum levels (55–70 dB SPL) were listener dependent and were equal for the three masker conditions. Masked thresholds for the broadband masker were near normal except for the tumor patients, who showed abnormally large (by 10–15 dB) masked thresholds at 1000 Hz. Auditory filters were estimated by plotting the masked thresholds for the low-pass and high-pass marker conditions at the marker cutoff frequencies. Linear functions were fitted to the masked thresholds to derive the auditory filter's half-power frequencies and bandwidth. The auditory filters were characterized by shifts in the center frequency of the filter below signal frequency and, often, by broad tuning. [Work supported by NIH.]
Binaural detection and discrimination for listeners with high-frequency sensorineural hearing losses
Authors: Koehnke I. , Colburn H. S. , Owen G. A.
Published: October 2005

As part of a survey study of binaural performance, NoSπ detection thresholds and just-noticeable differences (jnd's) in interaural time delay (ITD) and interaural intensity difference (IID) were measured for listeners with high-frequency sensorineural hearing losses. Thresholds and jnd's were obtained with 1/3-oct noise bursts at 500 and 4000 Hz for a variety of interaural reference conditions, including ITDs and IIDs up to 600 μs and 46 dB. The present data for high-frequency hearing loss subjects show poorer than normal binaural performance even at 500 Hz, a region where auditory thresholds are normal; however, binaural detection and discrimination performance at 4000 Hz is generally poorer than at 500 Hz. This is in agreement with previously reported results [e.g., J. Koehnke and H. S. Colburn, J. Acoust. Soc. Am. Suppl. 1 81, S27 (1987)]. In cases with unilateral losses studied to date, the compensation for interaural threshold differences through appropriate choice of the reference interaural intensity differences does not improve binaural performance. [Work supported by NIH.]
Characteristics of electrically induced otoacoustic emission
Authors: Moriyama Toshio , Hosokawa Yutaka , Minami Sadao , Murata Keiichi
Published: October 2005

Acoustic emissions were detected using the fast Fourier analysis of sounds in an ear canal during injection of alternating currents at different frequencies into the scala media of guinea pigs anaesthetized with urethane (1.2 g/kg i.p.). The amplitude of the distortion products relative to the fundamentals in the emitted sound was about ten times as large as that in the cochlear current, which was already slightly distorted, probably due to nonlinearity at the metal electrode surface. This discrepancy implies a larger additional nonlinearity in the cochlea. During COCB stimulation or after moderately intense sound exposure, the emitted distortion products were suppressed reversibly, in contrast to almost no change in the fundamental emissions. All the emissions were suppressed reversibly after temporary anoxia or furosemide administration, and they disappeared completely after severe acoustic trauma or death. These results suggest that the normal metabolic activity of the cochlea is required for generation of the emissions, and that the electromechanical transduction is less vulnerable than the cochlear nonlinearity.
Longitudinal analysis of spontaneous otoacoustic emissions in audiometrically normal adults
Authors: Haggerty Holly Susan
Published: October 2005

The second-by-second frequency stability of a spontaneous otoacoustic emission is purportedly high [W. Bialek and H. P. Wit, Phys. Lett. A 104, 173–178 (1984)], although minute-by-minute recordings have shown less stability. Additionally, while spontaneous emissions have been described as stable over years in a few subjects, diurnal and monthly variations have been reported for a few others [W. Fritze, Arch. Otorhinolaryngol. 238, 189–196 (1983); H. P. Wit, Hear. Res. 18, 197–199 (1985)]. To partially clarify this issue, spontaneous emissions were recorded from 11 normal-hearing adults. Diurnal, weekly, and monthly stability and variation of frequency amplitude were analyzed over the course of 2 months. Preliminary results indicate impressive intrasubject frequency and intensity variations. Final results are discussed with respect to theoretical modeling of active processes in the cochlea. [Work supported by the Knowles Research Foundation.]
Relations between frequency selectivity, temporal resolution, and speech intelligibility in sensorineural hearing-impaired subjects
Authors: Yonemoto Kiyoshi , Kurauchi Noriko , Hamada Hareo , Miura Tanetoshi
Published: October 2005

Frequency selectivity (the shape of the auditory filter), temporal resolution (the ability of gap detection), and speech intelligibility were measured for hearing-impaired subjects, and the relationships among these were investigated. The subjects were aged 18–49 years and had sensorineural hearing loss. Their average hearing threshold level was above 70 dB HTL. The shape of the auditory filter was measured from pure-tone threshold values under various maskers of notched broadband noise. Gap detection thresholds were measured using octave-band noise as a stimulus. A speech intelligibility test was conducted using two-syllable female speech. The results showed no correlation between hearing threshold level and any one of the following three parameters: auditory filter shape, gap detection threshold, and speech intelligibility score. However, the present subjects with severe hearing loss also showed a strong correlation between the sharpness of auditory filter and the speech intelligibility score, and a weak correlation between the gap detection threshold and the speech intelligibility score.
Evaluation of auditory enhancement effect in normal and hearing-impaired listeners
Authors: Thibodeau Linda M.
Published: October 2005

The auditory enhancement effect was evaluated in normal and hearing-impaired listeners using a forward masking paradigm similar to that used by Viemeister and Bacon [J. Acoust. Soc. Am. 71, 1502–1507 (1982)]. Masked thresholds for a 2000-Hz probe were compared between a standard condition in which the masker was a four-component harmonic complex including 2000 Hz, and an enhancing condition in which a three-component harmonic complex not including 2000 Hz preceded the four-component masker. The spectrum level of the masker was held constant at 85 dB SPL. For the normal listeners, the thresholds in the enhancing condition were approximately 30 dB greater than in the standard condition. For the hearing-impaired listeners, the enhancing condition resulted in only an 8- to 18-dB shift in threshold relative to the standard condition. One explanation for the enhancement effect in this case involves the adaptation of suppression in components surrounding the unadapted component such that there is actually an increase in intensity in the unadapted region. Suppression effects were also evaluated in both groups by measuring forward masked thresholds for a 2000-Hz probe as a function of a variable frequency suppressor tone that was added to a 2000-Hz masker. There was evidence of suppression for the normal listeners, particularly when the suppressor component was above the masker in frequency. However, for the hearing-impaired listeners, there was limited evidence of suppression as there was little change in masked threshold as a function of suppressor frequency.
Melody recognition by an Atlantic bottlenose dolphin (Tursiops truncatus)
Authors: Ralston James V. , Herman Louis M. , Williams Humphrey N. S. , Gory John D. , Jerger Kristin K.
Published: October 2005

Previous studies with monkeys and songbirds indicate a limited capability to recognize interval or contour information in melodic sequences of discrete sinusoidal tones. In the present experiments, two different types of melodic sequences, either four-tone sequences with constant pitch or four-tone sequences with descending pitch, were projected underwater to a 12-year-old adult female Atlantic bottlenose dolphin. The initial training attempted to teach the dolphin to press specific paddles after presentation of stimuli. Instead, the dolphin began to whistle spontaneously after descending-pitch sequences and remained silent after constant-pitch sequences. Over the same trials, she responded at chance with paddle presses. The paddles were removed, and vocal responses were thereafter judged in real-time by “blind” listeners. In subsequent tests, the dolphin successfully transferred the vocal responses to several pitch-transposed stimuli drawn from within, as well as a full octave above, the pitch range of the original training stimuli. These results provide the first evidence of octave generalization in a nonhuman listener and suggest a robust form of interval or melody recognition in this dolphin species. Data will also be presented from a second experiment contrasting rising and falling pitch sequences. [Work suppored by ONR.]
Mysticete whale sounds and human speech
Authors: Chu Kevin , Mackay R. Stuart
Published: October 2005

Toothed whales produce complex sounds, in some cases with two or more simultaneous sources in the head [R. S. Mackay and H. Liaw, Science 212, 676–678 (1981)] but mysticete vocalizations are said to be simple. Frequency analysis of recordings of humpback whales Megaptera novaeangliae on Silver Bank, Dominican Republic, produced sound spectrograms where certain pairs of components could both be present but with one increasing in frequency while the other decreased, or one alone could appear, or both could change in the same direction. This action is familiar in the formants of human speech, and some similarity of modulation mechanism is possible (though whales have no vocal cords). It does not seem possible that these recordings involved two animals singing in unison because timing remained the same for wide shifts of receiving point, which changed the relative distance to members of any source pair.