bluuebunny/crossref_metadata_2025 huggingface dataset:
The two-microphone transfer function method for measuring absorption coefficients and impedance: Some limitations and precautions
Authors: Nobile Matthew A.
Published: October 2005

The transfer function (TF) method described by Chung and Blaser [J. Acoust. Soc. Am. 68, 907–921 (1980)] is an attractive alternative to the conventional, and tedious, standing wave ratio technique. Many laboratories have already implemented the new method and are obtaining useful results. However, several problems are being encountered that have not been satisfactorily addressed in the literature. This paper discusses some of these problems, points out the limitations of the TF method as it is currently implemented, and suggests some necessary precautions. In addition, the importance of calibration is stressed and demonstrated through some examples (here, unlike two-microphone intensity measurements, the relative phase match between the microphone channels is just as critical at the higher frequencies as at lower frequencies). The results of measurements with the TF method, using two impedance tubes (100- and 30-mm diameters) to cover a frequency range of 90–6300 Hz, will be discussed.
Models of the pitch and pitch strength of ripple noise
Authors: Yost William A. , Hill Richard
Published: October 2005

A model based on a peripheral weighting function rather than on critical bands is used to provide predictions for the repetition pitch and pitch strength associated with ripple noise. The peripheral weighting function along with a bandpass filter representing the middle and outer ears produces a transformation of the ripple noise spectrum. The Fourier transform (autocorrelation) of this spectrum produces predictions consistent with the data obtained on repetition pitches. This approach is discussed in relation to other recent models of the pitch of complex stimuli. In particular, the predictions of the pattern-transformation model for the pitch of complex stimuli were compared to data from ripple noise experiments. The pattern-transformation model does not appear capable of accounting for many of the pitches associated with ripple noise.
Relationship between adaptation and the percept and transformations of stop consonant voicing: Effects of the number of repetitions and intensity of adaptors
Authors: Ohde Ralph N. , Sharf Donald J.
Published: October 2005

Important issues in selective adaptation research concern the relative contribution of response related (perceptual) and stimulus related (acoustic) effects of the adaptor in the adaptive process. Two response related issues pertain to the effects of the adaptor percept and verbal transformations on adaptation. This investigation systematically examined perceptual and acoustic contributions of the adaptor on the adaptation of the voicing feature. Subjects rated the degree of voicing/voicelessness of end-point VOT adaptors, i.e., 5- and 55-ms VOT, and an acoustically neutral adaptor, i.e., 25-ms VOT, during periods of repetitions. The number of adaptor repetitions during each of ten trials was either 5, 32, or 95, and the intensity of the adapting stimulus was either 50, 70, or 90 dB SPL. The major findings were as follows: (1) No significant correlations were found between ratings of voicing percept of the adaptor and magnitude of boundary shift; (2) Increases in repetitions and relative intensity level of end-point adaptors produced significantly greater phonetic boundary shifts and generally greater affects on ratings of test stimuli; and (3) The end-point adaptors produced significant shifts in rating of boundary and nonboundary stimuli. The findings indicate that neither the adaptor percept or verbal transformations affected the magnitude of adaptation. These results and those for acoustic parameters strongly suggest an acoustic as opposed to a phonetic basis of adaptation of the voicing feature. Furthermore, the effects of end-point adaptors on boundary and nonboundary stimuli support the generalized change in feature sensitivity assumed by a fatigue model of adaptation.
Transverse vibrations of circular plates having nonuniform edge constraints
Authors: Leissa A. W. , Laura P. A. A. , Gutierrez R.H.
Published: October 2005

The free vibrations of circular plates having flexible edge supports have been studied by several researchers for the restricted case when the supports are represented by springs having constant stiffness. In the present paper a general method is presented for dealing with supports having translational and rotational flexibilities which vary in an arbitrary manner around the boundary. It is shown that the varying stiffnesses can be represented as accurately as desired by expanding them into trigonometric series in the polar angle. The exact solution in polar coordinates of the differential equation of motion for the plate is then substituted into the elastic boundary conditions. The resulting infinite characteristic determinant is solved by successive truncation. As an example the case of a plate having a simply supported edge (infinite translational stiffness) with rotational stiffness varying according to L0+L1 cos ϑ (L0 and L1 being constants) is considered. Numerical results are obtained by the method described above and also by using the Ritz method with functions which approximate both the differential equation and the boundary conditions.
Intercomparisons of laboratory determinations of airborne sound transmission loss
Authors: Jones Robert E.
Published: October 2005

An analysis of transmission loss (TL) data from seven north american laboratories was organized by ASTM Subcommittee E33.05 to study ASTM E 90 accuracy performance for single- and double-panel partitions. Several mass law expressions are reviewed and compared with TL data from seven laboratories for four different limp-mass panel materials. It is noted that the laboratory TL data are not as linear with frequency as generally assumed, and examples are found where TL variation from mass law exceeds that suggested by E 90 precision requirements. In addition, an interlaboratory difference of 2 dB was found for some laboratories. For double-panel walls, TL data from two laboratories for similar constructions (STC ≳50) showed unexpectedly large variation in TL (up to eight points) and in STC (six points). Using double-panel theory, these TL differences are explained on the basis of interlaboratory differences in mass law and flanking (perhaps due to the test frame liners). A TL performance test series is illustrated with recent data from two laboratories and suggested as a basis for improving laboratory TL accuracy.
A theory of bubbly liquids
Authors: Drumheller Douglas S. , Bedford A.
Published: October 2005

A theory of bubbly liquids is developed in which the liquid and the bubbles are treated as separate continua. A variational procedure is used to obtain the equations of motion. The theory is linearized and applied to plane harmonic wave propagation. The dispersion and attenuation solutions are compared with experimental data and with other theoretical predictions. The theory is reduced to single continuum form, and results of previous single continuum theories are recovered.
Acoustic high-frequency scattering by elastic cylinders
Authors: Dickey J. W. , Überall H.
Published: October 2005

Acoustic scattering from an infinite elastic cylinder in a fluid is described by using the Sommerfeld–Watson transformation. The wavenumbers for the surface and transmitted waves are determined by poles or saddle points, respectively, of the scattering amplitude in the complex wavenumber plane. Dispersion and attenuation curves for the dominant surface wave modes have been obtained from the pole positions and given in our previous work. The present study gives the amplitudes of all the known contributions to the scattered field by evaluating the corresponding residues and the saddle-point amplitudes. The results indicate that the individual surface wave modes (i.e., Franz, Rayleigh, Stoneley, and Whispering Gallery), remain comparable to the transmitted-wave contributions to the scattered field (all these being plotted versus scattering angle at several frequencies) through frequencies at least as high as ka=200. This indicates that sound scattering from an elastic body does not tend to become predominantly geometrical-diffractive in the short-wavelength limit. Finally, some surface wave modes are plotted versus frequency, and are shown to contain a series of narrow resonances corresponding to the excitation of eigenvibrations of the elastic cylinder.
Ear dominance in perception of dichotic chords: Pure tones versus tone glides
Authors: Whittaker Ronald G. , Porter Robert J. , Cullen John K.
Published: October 2005

Pitch judgments for dichotic chords composed of two pure tones often show a bias in favor of the chord-component going to one ear. This ’’ear-advantage for pitch’’ (EAP) varies between subjects, but is very stable within subject and is thought to reflect differences in spectral ’’sensitivity’’ of the two auditory pathways. The present study explored this hypothesis by examining pitch judgments of complexes composed of tones dichotically paired with frequency-varying signals. The direction and strength of EAP was first established using pure tones, then rising tone glides were introduced into the channel going to the nondominant ear. Since a glide possesses less energy at a given frequency than a pure tone of equal duration, an increase in EAP was expected. An increase in EAP was consistently observed for only one subject; three subjects showed small, variable effects; and six subjects displayed a decrease in EAP. The results suggested that factors other than relative spectral sensitivity affect observed EAP.
Comments on ’’A rapid approximation to optimal array processing for the case of strong localized interferences’’ [J. Acoust. Soc. Am. 65, 456–462 (1979)]
Authors: Bordelon Derrill J.
Published: October 2005

Repeated application of Bartlett’s formula extended to complex matrices is employed to exhibit a representation of the inverse of the spectral matrix of signal, noise and interferences free of any matrix inversions.
Location of structurally similar areas in chinchilla cochleas of different lengths
Authors: Bohne Barbara A. , Carr Charles D.
Published: October 2005

In chinchillas, the length of the organ of Corti was found to vary by as much as 26% of the mean (18.31 mm). In specimens of different lengths, areas in which the basilar membrane and organ of Corti have the same physical dimensions were found to be located at the same percentage distance along the cochlear partition. Average densities of the inner and outer hair cells were noted to decrease with increasing cochlear length. However, the rate of decrease in density was less than the rate of increase in length. These findings indicate that long cochleas have a greater amount of the organ of Corti (and hence more sensory cells) in a given frequency region.
Aberrant tonotopic organization in the inner ear damaged by kanamycin
Authors: Robertson D. , Johnstone B. M.
Published: October 2005

Spiral ganglion cell recordings were obtained from normal and kanamycin-damaged guinea pig cochleas. Results show that kanamycin poisoning, which causes loss of outer hair cells, leads to a loss of sharp, sensitive tuning and, in addition, a shift to lower frequencies in the tuning curves from a damaged region. The normal tonoptic pattern of organization is therefore greatly disrupted in the damaged cochleas.
Band elimination processor for an experimental parametric acoustic receiving array
Authors: Rohde D. F. , Goldsberry T. G. , Olsen W. S. , Reeves C. R.
Published: October 2005

In the parametric acoustic receiving array (PARRAY), a low-frequency acoustic wave modulates a locally generated high-frequency acoustic carrier wave. The information in the low-frequency signal wave then appears as low-level, modulation sidebands of the carrier. The function of the receiver electronics is to suppress the high level carrier while simultaneously amplifying the low level near sideband signals. A band elimination processor has been developed that is capable of detecting signals with carrier to sideband ratios approaching 180 dB. Cascaded crystal filters provide the high order of carrier suppression. Additional circuitry is incorporated to permit independent observation of the upper and lower sidebands. The band elimination processor is described in detail and test data for sideband signals from 40 to 800 Hz centered about the 65-kHz carrier are presented.
Integrated-impulse method measuring sound decay without using impulses
Authors: Schroeder M. R.
Published: October 2005

A method of measuring linear-system responses (such as room responses) is described using ’’maximum-length’’ pseudorandom noise as the test signal. In this manner, high signal-to-noise ratios can be achieved, even for measurements in noisy environments and for low-power test signals. Pseudorandom noise has also been used successfully as the test signal in the ’’integrated-impulse’’ method of measuring sound decay and reverberation time. Thus, the need to radiate a short pulse of high peak energy for impulse type measurements is completely avoided. Improvements in signal-to-noise ratios are equal to the period length of the pseudorandom noise, typically 40 dB in room acoustical applications. The necessary digital processing to realize these gains in signal-to-noise ratio and accuracy of response can be performed on available minicomputers. Apart from maximum-length sequences, another type of periodic binary sequency, called the ’’Legendre sequence,’’ can be used as a test signal. Like maximum-length sequences, Legendre sequences have flat power spectra, but their discrete Fourier components have only two phase angles (?±90°), thus simplifying their digital representation (for storage and transmission). In fact, the discrete Fourier transform of a Legendre sequence is equal (within a constant factor) to the sequence itself! Legendre sequences exist for all period lengths equal to a prime number of the form 4k−1, where k is an integer. Thus, there are many more period lengths to choose from than for maximum-length sequences.
Acoustic reflection and radiation by thick fluid-loaded composite plates
Authors: Rudgers Anthony J.
Published: October 2005

The structural response function is found theoretically for a thick composite plate of infinite extent, which is composed of a thick coating layer bonded at an interface to a thick substrate layer, and which is loaded on one side by a fluid. The Timoshenko–Mindlin thick-plate equations are used in each of the two layers to describe the flexural waves that are generated, owing either to a plane acoustic wave in the fluid impinging upon the plate, or else to a point-force excitation of the plate. Two ideal types of interfacial bonding are considered: the ’’welded’’ bond, for which continguous plate elements on either side of the interface move in complete unison, and the ’’perfectly slipping’’ bond, for which such plate elements move in unison in the direction normal to the interface, but move completely independently in the direction parallel to the interface. From the analytic form of the structural response function of a composite plate, one can draw the following conclusions. First, there is, in general, no homogeneous plate that is equivalent to a thick bilaminar composite plate of the type considered. That is, a thick composite plate cannot be modeled by means of a simple thick plate with some sort of artificial or ’’average’’ material constants. Second, the analysis shows that the nature of the bond between the substrate layer and coating layer plays a significant role in determining the acoustic reflection and radiation characteristics of the composite plate.
A study of frequency transitions in cat vocalizations
Authors: Pick G. F.
Published: October 2005

In order to provide a starting point for neurophysiologists who seek neurons in the cat sensitive to frequency transitions, the 170 frequency transitions present in 36 cat vocalizations were analyzed in terms of octave rate of change and duration. The results are presented as histograms and scattergrams.
Psychophysical tuning curves and auditory thresholds after hair cell damage in the chinchilla
Authors: Ryan Allen , Dallos Peter , McGee Therese
Published: October 2005

Chinchillas were treated with kanamycin sulfate (150–200 mg/kg/day) to produce high-frequency hearing loss extending to about 4.0 kHz. Thresholds and psychophysical tuning curves (PTCs) were obtained before and after treatment, utilizing a shuttlebox avoidance procedure, and cochlear hair cells were evaluated under phase contrast microscopy. Hair cell loss resulting from kanamycin treatment varied from restricted lesions of the outer hair cells (OHCs) in the cochlear base, with no loss of inner hair cells (IHCs), to more extensive lesions involving both OHCs and IHCs. Threshold shift of at least 40 dB was always associated with OHC loss. PTCs obtained from frequency regions exhibiting 40–50 dB of threshold shift were normal in shape. With threshold shift in excess of 50 dB, PTCs were progressively distorted, with truncation of the tip segment and in some cases increased sensitivity of the tail segment. The results suggest that the threshold of optimally functional IHCs after kanamycin-induced OHC loss is about 40 dB higher than normal. Threshold shift in excess of 40 dB may represent IHC damage. IHCs are capable of transducing the fine-frequency information necessary for generating normally sharp PTCs in the absence of OHCs. However, with threshold shift in excess of approximately 50 dB, this frequency resolution is increasingly compromised.
Stimulated acoustic emissions from the human ear
Authors: Wit H. P. , Ritsma R. J.
Published: October 2005

The influence of stimulus frequency upon the magnitude of stimulated acoustic emissions from within the human ear was studied. Stimuli of higher frequency generate much smaller emissions (responses) than stimuli of lower frequency at the same stimulus level. For low response levels the relation between stimulus level and response level is approximately linear. High response levels rise approximately as the cube root of stimulus level. A tuning curve could be derived by suppressing emissions with a second steady tone.
Predicting musical pitch from component frequency ratios
Authors: Piszczalski Martin , Galler Bernard A.
Published: October 2005

A computer algorithm has been developed for music applications that automatically estimates pitch from any small number of frequencies, not necessarily of equal amplitude. Low-integer ratios are sought for each pair of frequencies. If an actual frequency-pair ratio closely approximates a low-integer ratio, the matching integers of this latter ’’ideal harmonic ratio’’ are accepted as possible harmonic numbers of the two actual frequencies in the tone. Hence, by comparing components to each other we can infer their likely harmonic numbers; if this is done successfully, then estimating the fundamental becomes a straightforward task. Therefore, the efficacy of the method lies in the cumulative effect of several component pair evaluations, all substantiating consistent harmonic assignments to the participating components. The method has proven to be useful in extracting pitch from spectral peaks taken from natural musical sounds, and it closely approximates some results of earlier experimental measurements of pitch perception. The method operates well with no fundamental present and for nonsuccessive partials (as well as inharmonic partials), and it is ’’phase insensitive.’’ Prior knowledge regarding the source of the signal (e.g., the musical instrument being played) is not required.
Scattering from oceanic microstructure: Detection with a large aperture array
Authors: Kaye G. T. , Anderson V. C.
Published: October 2005

A large aperture, high-frequency array with sound source has been used to detect returns from sound velocity microstructure. Returns were infinitely clipped and numerous beams were formed by steering for curved wave fronts. The processing was a unique application of digital array phasing that allowed discrimination between returns from reflectors and discrete point scatterers. Microstructure reflections were found to be highly directional with beamformer outputs dropping more than 15 dB as the angle of incidence varied to 2° off from normal. Detection of microstructure returns with conventional echosounders will be masked by biological reverberation for almost all typical oceanic thermoclines.
Reflection loss and mode attenuation in a Pekeris model
Authors: Weston D. E. , Tindle C. T.
Published: October 2005

The loss on reflection of a homogeneous acoustic wave from the boundary of an absorbing fluid is derived in several ways, including a calculation based on the attenuation along the beam displacement. For a so-called inhomogeneous plane wave there is an extra term, and the overall effect can be a loss or a gain. The paper considers several approaches to mode attenuation in an isovelocity or Perkeris channel with a lossy lower medium. We compare the horizontal attenuation of a mode with the horizontal attenuation of a ray due to multiple reflection losses. For exact agreement at all angles it is necessary either to include the beam displacement in the cycle distance, or to use the reflection coefficient appropriate to inhomogeneous waves, but these latter provisos are important only for shallow water and modes near cutoff. Beam displacement plays a part in both the reflection loss and the cycle distance calculations, and these aspects may be generalized.
Influence of atmospheric absorption on the propagation of bands of noise
Authors: Sutherland L. C. , Bass H. E.
Published: October 2005

As a band of noise propagates through the atmosphere, atmospheric absorption attenuates the higher frequencies in the band more rapidly than the lower frequencies. For large propagation distances, atmospheric absorption shapes the spectrum such that most of the acoustic energy is in the lower part of the band. When this occurs, the effective atmospheric absorption coefficient for this band of noise is less than when the energy is evenly distributed over the band. When a real filter is used to examine the received spectrum, the problem becomes even more complex since the filter does not cut off completely at the nominal lower frequency limit for the band. The equations governing the attenuation and spectrum analysis of noise under these conditions have been derived and a numerical integration process has been used to obtain quantitative results. It is shown that predicting observed values of atmospheric absorption loss for bands of noise involves two correction factors which are added to single-frequency absorption loss to define the effective attenuation for the band of noise. The first correction factor ΔI accounts for the difference between the attenuation predicted over a given path in a homogeneous atmosphere for a pure tone at the geometric center frequency of an ideal filter band, and the actual attenuation over the same path for the same ideal band of noise. The second correction factor ΔF accounts for the difference between the band level measured at one point with a practical filter with finite transmission outside the nominal filter pass band and the true band level measured at the same point with an ideal (rectangular) filter. Representative calculations show that the sum of these corrections become substantial when the product of distance, in km, and the square of frequency, in kHz, exceeds about 4 to 40 for octave and one-third octave band filters, respectively. Although the net effect of such corrections will generally be small for overall frequency weighted-noise levels (e.g., perceived noise level), the errors in diagnosing spectral detail in noise signatures of distant sources, such as aircraft, can be important.
Comments on ’’Acoustic Impedance Measurement using sine excitation and known volume velocity technique’’ [J. Acoust. Soc. Am. 64, 995–1003 (1978)]
Authors: Pratt R. L. , Elliott S. J. , Bowsher J. M.
Published: October 2005

The authors defend the accuracy and reliability of their measurements against the criticisms of Singh and Schary.
Is pitch a learned attribute of sounds? Two points in support of Terhardt’s pitch theory
Authors: Divenyi Pierre L.
Published: October 2005

Terhardt [J. Acoust. Soc. Am. 55, 1061–1069 (1974)] postulated a pitch perception model wherein a learning stage constitutes an integral part: it is only repeated exposure to patterns of spectral pitch that will generate the percept of virtual pitch (i.e., the residue). Two examples, one clinical and one musical, are cited to support the idea that perception of the pitch of complex tones represents a case of pattern perception which is acquired with experience.
Intensity discrimination with gated and continuous sinusoids
Authors: Green David M. , Nachmias Jacob , Kearney Joseph K. , Jeffress Lloyd A.
Published: October 2005

Detectability of sinusoidal signals at several durations was measured in the presence of sinusoidal maskers of the same frequency. Maskers were either continuous or gated on and off with the signal. For all signal durations and both types of maskers, ΔI/I varied with I raised to the −0.1 power, approximately. Psychometric functions were two or three times steeper with continuous than with gated maskers. Signal duration had only a small effect that varied with the masking condition.
Comments on ’’Array gain of a broad side vertical array in shallow water’’ [J. Acoust. Soc. Am. 65, 148–161 (1979)]
Authors: Clay C. S.
Published: October 2005

Buckingham uses a simple unshaded vertical array in a waveguide. The simple unshaded vertical array is an inefficient receiver because it does not match the eigenfunctions of the waveguide. For his noise sources he uses free space functions instead of the eigenfunctions of the waveguide. This letter clarifies these aspects of his paper and gives a procedure for designing an optimum array processing system. The analysis shows that the main advantage of using a vertical array is to filter the different modes and when the mode spectrums of the signal transmission and noise are the same there may be little or no gain of the signal-to-noise power ratio. The procedure also gives upper and lower limits of the signal-to-noise ratio for vertical arrays in waveguide transmission.
Use of the ’’acoustic menu’’ in assessing human response to audible (corona) noise from electric transmission lines
Authors: Molino John A. , Zerdy Gerald A. , Lerner Neil D. , Harwood Dane L.
Published: October 2005

A behavioral preference procedure, the ’’acoustic menu,’’ was used to assess human aversion to the sounds from two samples of transmission line audible noise and from samples of other environmental noises. The audible (corona) noise produced by extra-high voltage (EHV), overhead transmission lines was tape-recorded during moist weather. Reproduced samples of corona noise, other common environmental noises, and artificial reference stimuli were compared in two experiments. The two corona noise samples were found to be equally preferred to: (1) a 1000 Hz octave band of noise about 11 dB SPL higher than the corona noises and (2) a collection of other environmental noises about 8 dB SPL higher than the corona noises. The corona noise samples were more aversive than the ambient sounds (including rainfall) occurring near rural transmission lines and were roughly equivalent in aversiveness to the noise from a room air conditioner (indoor recording). Knowledge of the source of the corona noise (from photographs and a brief description) did not affect its aversiveness. Of the simple frequency-weighting scales, the A-weighted sound level reduced the difference between the two corona noise samples and the set of other environmental sounds from about 8 dB SPL to about 3 dB. The D-weighted sound level further reduced this discrepancy to about 2 dB. These results were obtained with only two samples of corona noise; other samples may give different results.
The nonlinearity parameter of saturated marine sediments
Authors: Hovem Jens M.
Published: October 2005

The sources that may account for the acoustic nonlinearity of marine sediments are discussed based on the theory for wave propagation in porous media. It is concluded that the nonlinearity parameter B/A can be expressed as the weighted sum of contributions from the nonlinear stress– strain relations of the pore fluid, the mineral grains, and the skeleton frame. Calculations of B/A for sands show the nonlinearity of the fluid to be most important with a dependence on porosity in good agreement with earlier experimental results. Some consideration is given to distortion of the finite amplitude waves and the effect of this on acoustic propagation.
Arrangement for precision reciprocity calibration of condenser microphones
Authors: Wong G. S. K. , Embleton T. F. W.
Published: October 2005

This paper describes a precision ac null-detecting arrangement that has a resolution of better than 0.001 dB for reciprocity calibration of condenser microphones. Systematic errors are analyzed in depth. The uncertainty of the absolute open-circuit sensitivity is predicted to be less than 0.005 dB, which is approximately an order of magnitude improvement over conventional calibration arrangements. With international standardization, a comparison accuracy of a few thousandths of a decibel can be achieved.
Frequency domain techniques for speech coding
Authors: Crochiere R. E. , Tribolet J. M.
Published: October 2005

Frequency domain techniques for speech coding have recently received considerable attention. The basic concept of these methods is to divide the speech into frequency components by a filter bank (subband coding) or by a suitable transform (transform coding) and then encode them using adaptive PCM. Four basic operations are involved in the design of these coders: (1) the type of transform or filter bank (analysis/synthesis), (2) the adaptive quantizer design (quantization theory), (3) the choice of bit allocation used by the quantizers (noise shaping and auditory masking), and (4) the control of the step-size of the quantizers (spectral estimation). This paper briefly reviews the basic aspects of the design of these four operations particularly as they apply to low bit-rate adaptive transform coding.
Diffraction of ultrasonic waves by penny-shaped cracks in metals: Theory and experiment
Authors: Achenbach Jan D. , Adler Laszlo , Lewis D. Kent , McMaken Harry
Published: October 2005

In this paper an analytical solution to the diffraction of elastic waves by penny-shaped cracks in metals is compared with experimental observations. The analysis, which is based on elastodynamic ray theory, is valid for the region of ka ≳ 1. A digitized spectrum analysis system is described which measures the frequency components of the waves diffracted from a 2500-μ-radius crack in diffusion bonded titanium. The amplitude spectra show good agreement between experiment and theory. The theoretically predicted periodicity of the diffracted spectra provides a simple formula for the inverse problem. Application of this formula to the experimental measurements determines the crack size with excellent accuracy.
Direct and inverse problems in radiation of sound from discrete random sources on two coaxial rings
Authors: Maestrello Lucio
Published: October 2005

An analytical model consisting of two ring sources of sound is developed to study the direct radiation in terms of correlation, coherence, and phase and also to aid in solving the inverse-radiation problem of determining the noise source in terms of farfield measurements. The rings consist of discrete sources which are either monopoles or quadrupoles with Gaussian autocorrelations. Only adjacent sources, both within and between the rings, are correlated. Results show that from the farfield information one can determine when the sources are compact or noncompact with respect to the acoustic wavelength and distinguish between the types of sources. In addition, from the inverse-radiation approach one can recover the center of mass, the location and separation distance of the ring, and the respective diameters.
A note on the derivation of the wave equation in an inhomogeneous ocean
Authors: Goodman R. R. , Farwell R. W.
Published: October 2005

Several derivations of the wave equation in an inhomogeneous medium appear in the literature. A term ∇p1⋅∇lnρ0 is found in all of them. Inhomogeneities in the sea are due to both gravity and other effects. The reference state characterized by ρ0(xyz) gives rise to two different limits, one due to gravity alone and the other due to inhomogeneities ignoring gravity. Both lead to the same form of the wave equation, but different arguments are required to demonstrate that ∇p1⋅∇lnρ0 can be ignored. A seldom cited paper by Bergmann [J. Acoust. Soc. Am. 17, 329–333 (1946)] is used.
Evaluation of a segmental SNR measure as an indicator of the quality of ADPCM coded speech
Authors: Mermelstein Paul
Published: October 2005

Two objective measures of speech quality, the traditional signal to quantization noise ratio (SNR) and a segmental SNR, are compared from the point of view of their ability to predict subjective preference ratings of PCM and ADPCM coded speech. The SNR shows an 8-dB bias for PCM coding, i.e., the SNR of the PCM has to exceed the SNR of ADPCM by 8 dB before the two are judged to be equally preferable. In contrast, the segmental SNR shows no such bias. The segmental SNR is considered a better perceptual model since it evaluates the quantization noise with respect to the energy in each underlying speech segment.
Formant frequency patterns in Russian VCV utterances
Authors: Purcell Edward T.
Published: October 2005

Previous studies have sought to establish the significance of various parameters in the determination of the patternings of Russian formant frequency trajectories in vowel-consonant-vowel (VCV) syllables. In the present study, 9600 measurements of first and second formant frequency were made on Russian VCV nonsense words. Variance for speakers was controlled through regression. Stepwise multiple regression was employed to determine the relative contributions of six predictors to the explanation of the patternings of first and second formant frequency in the test words. The six predictors included the height of the first and second vowels, the front/back dimensionality of the first and second vowels, the place of articulation of the consonant, and the palatalization or nonpalatalization of the consonant. Results of the regression analyses are summarized. The regression coefficients for the useful predictors of the various dependent variables comprise a model of formant frequency patterns in Russian VCV utterances.
A system for evaluating auditory function from 8000–20 000 Hz
Authors: Fausti Stephen A. , Frey Richard H. , Erickson Deborah A. , Rappaport B. Z. , Cleary Edward J. , Brummett Robert E.
Published: October 2005

A system for the measurement of auditory function from 8000–20 000 Hz is described. This system introduces advances in: (a) maximum power output, (b) signal fidelity, and (c) transducer characteristics. Two case studies are presented to illustrate the clinical information gained from the measurement of high-frequency auditory sensitivity, which is not readily apparent in conventional threshold assessment.
A neural-counting model incorporating refractoriness and spread of excitation. I. Application to intensity discrimination
Authors: Teich Malvin Carl , Lachs Gerard
Published: October 2005

We consider in detail a new mathematical neural-counting model that is remarkably successful in predicting the correct detection law for pure-tone intensity discrimination, while leaving Weber’s law intact for other commonly encountered stimuli. It incorporates, in rather simple form, two well-known effects that become more marked in the peripheral auditory system as stimulus intensity is increased: (1) the spread of excitation along the basilar membrane arising from the tuned-filter characteristics of individual primary afferent fibers and (2) the saturation of neural counts due to refractoriness. For sufficiently high values of intensity, the slope of the intensity-discrimination curve is calculated from a simplified (crude saturation) model to be 1−1/4N, where N is the number of poles associated with the tuned-filter characteristic of the individual neural channels. Since 1?N<∞, the slope of this curve is bounded by 3/4 and 1 and provides a theoretical basis for the ’’near miss’’ to Weber’s law.
Two theoretical results suggesting a method for calibrating ultrasonic transducers by measuring the total nearfield force
Authors: Miller Eric B. , Yaghjian Arthur D.
Published: October 2005

Theory and preliminary experiments are outlined relating to a nearfield method of evaluating electroacoustic transducers. The theoretical results are conveniently organized into two theorems: (1) The total complex force on all parallel infinite planes to one side of a transducer and perpendicular to an arbitrary direction, has a constant magnitude equal to the magnitude of the farfield pressure in that same direction multiplied by the wavelength. (2) The output voltage of a baffled, reciprocal, plane-piston receiver is proportional to the total incident force normal to its face. These two theorems suggest a compact and relatively simple method for evaluating directive or moderately directive ultrasonic transducers. It is demonstrated experimentally that there is good agreement between the farfield pattern of a directive transducer in the main beam region, measured by a large, plane-piston transducer in the nearfield, and the pattern measured directly with a small probe in the approximate farfield. Further, it is experimentally demonstrated that, as a piston receiver is moved axially, with respect to a transmitter, no discernible change in the output voltage of the receiver is detected, provided the nearfield beam of the transmitter is intercepted by the receiver (even though the nearfield of either transducer differs significantly from that of a plane wave).
Topics on speech waveform coding activities in Japan
Authors: Tomozawa Atsushi , Ochiai Kazuo
Published: October 2005

Current activities on speech waveform coding technique developments in Japan are reviewed. The following topics of the activities on ADPCM and ADM are discussed. (1) An ADPCM code sequence optimization method by Fushikida for an ADPCM with an adaptive quantization and fixed predictor. The method uses a technique reminiscent of dynamic programming and optimizes code sequence. It is shown that about 3 dB SNR improvement is obtainable for speech at 32 kbps (4 bits/sample). (2) Another ADPCM technique by Araseki and Ochiai. This technique is suitable for use at the bit rate less than 32 kbps. Two adaptive predictors connected in a cascaded form are used. One of the predictors has an ordinary small number of prediction taps. However, the other predictor possesses a considerably larger number of predictor taps and is used to predict the speech waveform which is one pitch period apart. (3) Performance improvement techniques and LSI implementation by Tomozawa and Niwa for a discretely variable slope 32 kbps ADM (DVSD). Yatsuzuka’s residual encoder and Hosokawa and Yamashita’s adaptive predictive DM are also discussed briefly.
Optimizing digital speech coders by exploiting masking properties of the human ear
Authors: Schroeder M. R. , Atal B. S. , Hall J. L.
Published: October 2005

In any speech coding system that adds noise to the speech signal, the primary goal should not be to reduce the noise power as much as possible, but to make the noise inaudible or to minimize its subjective loudness. ’’Hiding’’ the noise under the signal spectrum is feasible because of human auditory masking: sounds whose spectrum falls near the masking threshold of another sound are either completely masked by the other sound or reduced in loudness. In speech coding applications, the ’’other sound’’ is, of course, the speech signal itself. In this paper we report new results of masking and loudness reduction of noise and describe the design principles of speech coding systems exploiting auditory masking.
The effect of radiation loading on the equivalent circuit for a piezoelectric polymer transducer
Authors: Hudimac Albert A.
Published: October 2005

The radiation from an infinite hexagonal plane array of of circular, distended piezoelectric polymer film elements of a transducer is considered. The behavior of each element is approximated by that of an element radiating into a circular tube with rigid walls. The ratio of the area of the film element to the area of the cross section of the tube, the packing factor, greatly affects the behavior. The displacement of the film is expressed in terms of normal modes. Each mode has a kind of impedance which depends not only on the transducer parameters but also on the acoustic coupling between modes. The equivalent circuit of an element has the parallel arrangement of the modal impedances on the mechanical side of electromechanical transformer. Computations at the lower end of the spectrum demonstrate that the motional resistance is constant (except possibly for a stray antiresonance) and varies directly with packing factor; also, that the motional reactance is smaller over large frequency intervals. A single unit, with back-to-back distended sheets, is also treated.
Can intrinsic vowel F be explained by source/tract coupling?
Authors: Ewan William G. , Ohala John J.
Published: October 2005

There is extensive evidence that in the same phonetic environment the voice fundamental frequency (F0) of vowels varies directly with vowel ’’height.’’ This F0 difference between vowels could be caused by acoustic interaction between the first vowel formant and the vibrating vocal folds. Since higher vowels have lower first formants than low vowels the acoustic interaction should be greatest for high vowels whose first formant frequencies are closer in frequency to F0. Ten speakers were used to see if acoustic interaction could cause the F0 differences. The consonant [m] was recorded in the utterances [umu] and [ama]. Although the formant structure of [m] in [umu] and [ama] should not differ significantly, the F0 of each [m] allophone was significantly different. However, the F0 of each [m] allophone did not differ significantly from the F0 of the following vowel. These results did not support acoustic interaction. However, it is quite reasonable to conclude that the F0 variation of [m] was caused by coarticulatory anticipation of the tongue and jaw for the following vowel. Another experiment is offered in order to help explain the physical causes of intrinsic vowel F0. In this experiment F0 lowering was found at the beginning of vowels following Arabic pharyngeal approximants. This finding indicates that the F0 of pharyngeal constricting vowels, e.g., [] and [a], might be lowered as a result of similar articulatory movements, viz. tongue compression and active pharyngeal constriction.
Unmasking produced by combination tones
Authors: Tyler Richard S. , Small Arnold M. , Abbas Paul J.
Published: October 2005

This experiment investigated some interactions between combination tones (CT) and unmasking. Data were obtained from nine normal-hearing listeners in a forward-masking paradigm in which a 2000-Hz, 35-dB SPL masker preceded a 2000-Hz test signal. An adaptive, three-interval forced-choice procedure was employed to estimate the threshold for the test signal. This measurement provided a baseline. When a second sinusoid, the suppressor, of variable frequency and 55 dB SPL was presented concurrently with the masker, the amount of masking could be less than in the baseline condition. This two-tone unmasking was observed when the suppressor was approximately 2300 Hz. In the cubic difference tone (CDT) condition, two higher-frequency sinusoids of 80 dB SPL (with the lower frequency fixed at 2800 Hz) were added to the masker. In the difference tone (DT) condition, two higher-frequency sinusoids of 100 dB SPL (with the lower frequency fixed at 3500 Hz) were added to the masker. None of these sinusoids produced unmasking when presented individually, but when the CDT and DT were approximately 2300 Hz, unmasking was observed. The unmasking produced by the CTs was similar in its frequency characteristics to that of the single, suppressor tone. These results indicate that unmasking can be produced by CTs, just as if CTs were present in the stimulus. These data indirectly support the notion that CTs may be present in the form of traveling waves propagated along the basilar membrane.
Automated dynamic Young’s modulus and loss factor measurements
Authors: Madigosky Walter M. , Lee Gilbert F.
Published: October 2005

A progressive wave apparatus featuring an automated data processor is described. The apparatus accurately determines the propagation constants of an extensional acoustic wave by exciting one end of a strip of material while the other end is suspended under constant tension. The apparatus is capable of making measurements in the frequency range of 100 Hz to 40 kHz, depending on the extensional wave velocity and attenuation in the material. As illustrations of the technique, measurements were made on three rubber compounds: a polyurethane, a neoprene, and a nitrile rubber. Extensional sound speed, attenuation, modulus, and loss factor were determined over a frequency range of 1–10 kHz and a temperature range of 4 °–47 °C. Applying the time–temperature superposition principle to the data, master curves were constructed and WLF shift constants were determined. The apparatus was found to be a fast and reliable method to determine the dynamic viscoelastic constants.
Intensive and temporal effects in pure-tone forward masking
Authors: Widin Gregory P. , Viemeister Neal F.
Published: October 2005

Intensive and temporal characteristics of forward masking were assessed using a pure-tone masker and a 1-kHz probe. For a given interval between masker and probe, the slope of the masking function tends to decrease with increasing masker frequency, but is always less than unity. When the masker frequency equals that of the probe, increasing the masker-probe interval decreases the slope of the masking function. Implications of these data for forward-masked psychophysical tuning curves are discussed. A model, based on a linear combination of exponential decays and an initial logarithmic transform of stimulus intensity, predicts the general form of the masking function for varying interval between masker and probe, and suggests time constants which decrease with increasing masker level.