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Top 20 Most Read Articles
February 2010
The 20 articles with the most full-text downloads during the month, in descending order.
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The sweet spot of a hollow baseball or softball bat J. Acoust. Soc. Am. Volume 116, Issue 4, pp. 2602-2602 (2004); (1 page)
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The ‘‘sweet spot’’ of a baseball bat is often defined as a region approximately 5–7 in. from the barrel end of the bat where the performance (as measured by a batted‐ball speed) is the greatest and the vibrational sensation (the sting felt in the hands) is minimized. In terms of the vibrational properties of a bat, the sweet spot region includes the nodes of the first two bending vibrational modes and is often also identified as being related to the center‐of‐percussion (COP). This paper will discuss recent research which shows that the COP has no bearing on performance and little, if any, on the perception of feel. We will also discuss the contribution of the bending vibrational modes to the location of the sweet spot and compare results for a wide variety of wood, metal and composite baseball and softball bats. We will attempt to correlate the sweet spot defined in terms of performance and that defined in terms of vibrational sensation. Finally, we will discuss the contribution of the hoop mode, unique to hollow bats, which correlates strongly to performance, and which may also influence the perception of feel. |
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Advantages of a Two‐Transducer Frequency Analysis Technique to Characterize Discontinuities J. Acoust. Soc. Am. Volume 52, Issue 1A, pp. 162-162 (1972); (1 page) Online Publication Date: 12 Aug 2005
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In an earlier paper [J. Acoust. Soc. Amer. 48, 102(A) (1970)], we have shown that the frequency analysis of a broad‐band ultrasonic echo may be used to determine the size and orientation of flat circular reflectors. A single transducer was used as both source and receiver. The one‐transducer technique is not completely satisfactory for the case of discontinuities having both random shape and orientation. Recently, we have used a second transducer to receive ultransonic pulses scattered by a reflector. Defects in materials were simulated by irregularly shaped metal shims suspended with random orientation in water. The received frequency spectrum when analyzed shows characteristic maxima and minima, which depend on the shape, size, and orientation of the reflectors. A model—which assumes an interference pattern formation of the scattered spherical wavelets at the receiver—relates the spacing between frequency maxima and the positions of the two transducers to the size and orientation of the “discontinuities.” The measurement and the analysis is greatly simplified with the use of the second transducer, and size determination of the discontinuities can usually be made within 10%–20%. [Research sponsored by the U. S. Atomic Energy Commission under contract with the Union Carbide Corporation.] |
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Crosstalk mitigation using adaptive time reversal J. Acoust. Soc. Am. Volume 127, Issue 2, pp. EL19-EL22 (2010); (4 pages) Online Publication Date: 08 Jan 2010
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Although the spatial focusing property of the conventional time reversal approach facilitates multiuser communications, there always is residual crosstalk between users. A recent paper [
Kim and Shin, J. Acoust. Soc. Am. 115, 600–606 (2004)
] proposed an adaptive active time reversal approach for simultaneous multiple focusing with minimal interference. This letter applies the adaptive approach to passive time reversal, multiuser communications for additional suppression of crosstalk among users. Experimental data at 3.5 kHz with a 1-kHz bandwidth demonstrate as much as 6.5-dB improvement per user in terms of output signal-to-noise ratio for three-user communications over a 20-km range in 120-m deep shallow water.
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A simple method to account for size effects in the transfer matrix method J. Acoust. Soc. Am. Volume 127, Issue 2, pp. EL30-EL36 (2010); (7 pages) Online Publication Date: 12 Jan 2010
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The transfer matrix method based is extensively used and well-validated for predicting the transmission loss of multilayer structures. However, this method leads to poor results at low frequencies due to the infinite extent assumption it is based on. This paper presents an efficient implementation of a Rayleigh-integral based method to account for the finite size effects. The accuracy of the method is illustrated by various examples.
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New Fellows of the Acoustical Society of America J. Acoust. Soc. Am. Volume 127, Issue 2, pp. 1166-1166 (2010); (1 page) Online Publication Date: 05 Feb 2010
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The neural code for interaural time difference in human auditory cortex J. Acoust. Soc. Am. Volume 127, Issue 2, pp. EL60-EL65 (2010); (6 pages) Online Publication Date: 19 Jan 2010
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A magnetoencephalography study was conducted to reveal the neural code of interaural time difference (ITD) in the human cortex. Widely used crosscorrelator models predict that the code consists of narrow receptive fields distributed to all ITDs. The present findings are, however, more in line with a neural code formed by two opponent neural populations: one tuned to the left and the other to the right hemifield. The results are consistent with models of ITD extraction in the auditory brainstem of small mammals and, therefore, suggest that similar computational principles underlie human sound source localization.
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Effects of stimulus duration on amplitude modulation processing with cochlear implants J. Acoust. Soc. Am. Volume 127, Issue 2, pp. EL23-EL29 (2010); (7 pages) Online Publication Date: 11 Jan 2010
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This study investigated temporal integration processes underlying cochlear implant (CI) users’ amplitude modulation processing. Thresholds for modulation detection (AMDTs) and modulation frequency discrimination (AMFDTs) were measured for 50-, 100-, and 200-Hz modulation frequencies with stimulus durations from 50 to 400 ms in eight adult CI users. The results showed significant interactions between modulation frequency and stimulus duration for AMDTs and AMFDTs. The data suggest that temporal integration limits CI users’ sensitivity to low temporal pitch over short durations, and that temporal integration over longer durations may not enhance CI users’ sensitivity to high temporal pitch.
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Speech levels in meeting rooms and the probability of speech privacy problems J. Acoust. Soc. Am. Volume 127, Issue 2, pp. 815-822 (2010); (8 pages) Online Publication Date: 05 Feb 2010
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Speech levels were measured in a large number of meetings and meeting rooms to better understand their influence on the speech privacy of closed meeting rooms. The effects of room size and number of occupants on average speech levels, for meetings with and without sound amplification, were investigated. The characteristics of the statistical variations of speech levels were determined in terms of speech levels measured over 10 s intervals at locations inside, but near the periphery of the meeting rooms. A procedure for predicting the probability of speech being audible or intelligible at points outside meeting rooms is proposed. It is based on the statistics of meeting room speech levels, in combination with the sound insulation characteristics of the room and the ambient noise levels at locations outside the room.
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An acoustic comparison of nonlinguistic sounds to sentences spoken in American English. J. Acoust. Soc. Am. Volume 124, Issue 4, pp. 2454-2454 (2008); (1 page) Online Publication Date: 15 Oct 2008
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Acoustic characteristics of nonlinguistic (nonspeech) sounds (NLSs) were measured for duration and spectral variation, and compared to acoustic characteristics of spoken sentences (TIMIT database). The NLS, included samples produced by animal, human, mechanical, and natural sources. The acoustic comparison examined stoplike onsets, fricativelike intervals, vowel‐like intervals, and syllablelike variations in amplitude. The NLSs were identified by two groups of listeners: listeners with normal hearing and users of cochlear implants. Results of the listening tests have been reported previously by Inverso et al. (2007) and Inverso (2008). All of the NLSs were identified accurately by listeners with normal hearing, but not by the users of cochlear implants. The current analysis focuses on the ways in which the NLSs are similar to and other ways in which they are different from sentences spoken by a variety of talkers. It was found that speechlike variation in amplitude, both in terms of duration and event onset∕offset, was a strong cue for listeners with cochlear implants; that is, NLSs that contained distinct events that were similar in duration and amplitude variation to syllables in speech were identified more accurately than ones that did not. |
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A non-linear three-dimensional model for quantifying microbubble dynamics J. Acoust. Soc. Am. Volume 127, Issue 2, pp. EL80-EL86 (2010); (7 pages) Online Publication Date: 25 Jan 2010
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A three-dimensional non-linear model for simulating microbubble response to acoustic insonation is presented. A 1 μm radius microbubble stimulated using positive and inverted 2.4 MHz pulses produced radius-time curves that matched (error <10%) with the experimental observation. A bound 2.3 μm radius microbubble insonated using 2.25 MHz 6 cycle pulse was observed to oscillate with max/min oscillations 45% lower than that of the free microbubble, this correlated ( ∼ 10% error) with the observations of Garbin et al. [
Appl. Phys. Lett. 90, 114103 (2007)
]. The adherent microbubble oscillated asymmetrically in the plan view and symmetrically in the elevation view, consistent with the previous experimental results.
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A mathematical model of vowel identification by users of cochlear implants J. Acoust. Soc. Am. Volume 127, Issue 2, pp. 1069-1083 (2010); (15 pages) Online Publication Date: 05 Feb 2010
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A simple mathematical model is presented that predicts vowel identification by cochlear implant users based on these listeners’ resolving power for the mean locations of first, second, and/or third formant energies along the implanted electrode array. This psychophysically based model provides hypotheses about the mechanism cochlear implant users employ to encode and process the input auditory signal to extract information relevant for identifying steady-state vowels. Using one free parameter, the model predicts most of the patterns of vowel confusions made by users of different cochlear implant devices and stimulation strategies, and who show widely different levels of speech perception (from near chance to near perfect). Furthermore, the model can predict results from the literature, such as
Skinner, et al. [(1995). Ann. Otol. Rhinol. Laryngol. 104, 307–311
] frequency mapping study, and the general trend in the vowel results of Zeng and Galvin’s [
(1999). Ear Hear. 20, 60–74
] studies of output electrical dynamic range reduction. The implementation of the model presented here is specific to vowel identification by cochlear implant users, but the framework of the model is more general. Computational models such as the one presented here can be useful for advancing knowledge about speech perception in hearing impaired populations, and for providing a guide for clinical research and clinical practice.
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J. Acoust. Soc. Am. Volume 127, Issue 2, pp. 943-954 (2010); (12 pages) Online Publication Date: 05 Feb 2010
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The neural mechanisms underlying the ability of human listeners to recognize speech in the presence of background noise are still imperfectly understood. However, there is mounting evidence that the medial olivocochlear system plays an important role, via efferents that exert a suppressive effect on the response of the basilar membrane. The current paper presents a computer modeling study that investigates the possible role of this activity on speech intelligibility in noise. A model of auditory efferent processing [
Ferry, R. T., and Meddis, R. (2007). J. Acoust. Soc. Am. 122, 3519–3526
] is used to provide acoustic features for a statistical automatic speech recognition system, thus allowing the effects of efferent activity on speech intelligibility to be quantified. Performance of the “basic” model (without efferent activity) on a connected digit recognition task is good when the speech is uncorrupted by noise but falls when noise is present. However, recognition performance is much improved when efferent activity is applied. Furthermore, optimal performance is obtained when the amount of efferent activity is proportional to the noise level. The results obtained are consistent with the suggestion that efferent suppression causes a “release from adaptation” in the auditory-nerve response to noisy speech, which enhances its intelligibility.
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Efficient numerical modeling of traffic noise J. Acoust. Soc. Am. Volume 127, Issue 2, pp. 796-803 (2010); (8 pages) Online Publication Date: 05 Feb 2010
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An optimized method is presented for the numerical evaluation of the sound field generated by an incoherent line source, which is commonly used to model road and rail traffic noise. Two different solutions for the numerical integration over the line source are distinguished, a point source solution and a line source solution. With proper segmentation of the line source, both solutions yield accurate results. Special attention is paid to receiver positions close to the (infinite) line through the (finite) line source. At these positions, conventional methods give numerical errors, which occur frequently in calculations of large-scale noise maps of cities, employing automatically generated geographical input data. The problems are avoided by using the optimized method presented here. The method is based on a combination of angular segmentation and linear segmentation of the line source and can be used to minimize the number of point-to-point calculations for noise mapping.
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Ray-based blind deconvolution in ocean sound channels J. Acoust. Soc. Am. Volume 127, Issue 2, pp. EL42-EL47 (2010); (6 pages) Online Publication Date: 13 Jan 2010
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This letter describes a ray-based blind deconvolution technique for ocean sound channels that produces broadband estimates of the source-to-array impulse response and the original source waveform from array-measured signals corrupted by (unknown) multipath propagation. The technique merely requires elementary knowledge of array geometry and sound speed at the array location. It is based on identifying a ray arrival direction to separate source waveform and acoustic-propagation phase contributions to the received signals. This technique successfully decoded underwater telecommunication sequences in the bandwidth 3–4 kHz that were broadcast 4 km in a 120-m-deep ocean sound channel without a-priori knowledge of sound channel characteristics.
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J. Acoust. Soc. Am. Volume 127, Issue 2, pp. 841-849 (2010); (9 pages) Online Publication Date: 05 Feb 2010
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This paper presents a revised theory for predicting the sound insulation of double leaf cavity walls that removes an approximation, which is usually made when deriving the sound insulation of a double leaf cavity wall above the critical frequencies of the wall leaves due to the airborne transmission across the wall cavity. This revised theory is also used as a correction below the critical frequencies of the wall leaves instead of a correction due to Sewell [
(1970). J. Sound Vib. 12, 21–32
]. It is found necessary to include the “stud” borne transmission of the window frames when modeling wide air gap double glazed windows. A minimum value of stud transmission is introduced for use with resilient connections such as steel studs. Empirical equations are derived for predicting the effective sound absorption coefficient of wall cavities without sound absorbing material. The theory is compared with experimental results for double glazed windows and gypsum plasterboard cavity walls with and without sound absorbing material in their cavities. The overall mean, standard deviation, maximum, and minimum of the differences between experiment and theory are −0.6 dB, 3.1 dB, 10.9 dB at 1250 Hz, and −14.9 dB at 160 Hz, respectively.
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J. Acoust. Soc. Am. Volume 127, Issue 2, pp. 1124-1134 (2010); (11 pages) Online Publication Date: 05 Feb 2010
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In this paper, techniques for time-frequency analysis and investigation of bat echolocation calls are studied. Particularly, enhanced resolution techniques are developed and/or used in this specific context for the first time. When compared to traditional time-frequency representation methods, the proposed techniques are more capable of showing previously unseen features in the structure of bat echolocation calls. It should be emphasized that although the study is focused on bat echolocation recordings, the results are more general and applicable to many other types of signal.
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J. Acoust. Soc. Am. Volume 127, Issue 2, pp. 784-795 (2010); (12 pages) Online Publication Date: 05 Feb 2010
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Enclosures are commonly used to reduce the sound exposure of workers to the noise radiated by machinery. Some acoustic predictive tools ranging from simple analytical tools to sophisticated numerical deterministic models are available to estimate the enclosure acoustical performance. However, simple analytical models are usually valid in limited frequency ranges because of underlying assumptions whereas numerical models are commonly limited to low frequencies. This paper presents a general and simple model for predicting the acoustic performance of large free-standing enclosures which is capable of taking into account the complexity of the enclosure configuration and covering a large frequency range. It is based on the statistical energy analysis (SEA) framework. The sound field inside the enclosure is calculated using the method of image sources. Sound transmission across the various elements of the enclosure is considered in the SEA formalism. The model is evaluated by comparison with existing methods and experimental results. The effect of several parameters such as enclosure geometry, panel materials, presence of noise control treatments, location of the source inside the enclosure, and presence of an opening has been investigated. The comparisons between the model and the experimental results show a good agreement for most of the tested configurations.
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The scattering of sound by a long cylinder above an impedance boundary J. Acoust. Soc. Am. Volume 127, Issue 2, pp. 664-674 (2010); (11 pages) Online Publication Date: 05 Feb 2010
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The classical problem of sound scattering by an acoustically hard cylinder due to a point monopole and a line airborne source is extended in the present study. The solution to the homogeneous Helmholtz equation is expressed in a cylindrical coordinate system and represented by an expansion of Fourier integrals. Incorporating the image source method and the Bessel function addition theorem, the analytical solution is derived for the prediction of multiple scattering of sound by a hard cylinder placed above a ground surface of finite impedance. The total sound field can be expressed as a sum of four components: the incident field, the reflected wave, and the scattered fields from the cylinder and its image. The total far-field scattered potential was evaluated asymptotically by the method of stationary phase. Experimental measurements by using a point source were conducted in an anechoic chamber to validate the theoretical formulations. The numerical predictions of using a point source model give good agreements with all the experimental data but there are obvious discrepancies in the spectral magnitudes between the calculation and experimental results when a line source model is used to simulate the scattering problem due to a point source excitation.
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Recognition of interrupted sentences under conditions of spectral degradation J. Acoust. Soc. Am. Volume 127, Issue 2, pp. EL37-EL41 (2010); (5 pages) Online Publication Date: 13 Jan 2010
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Cochlear implant (CI) and normally hearing (NH) listeners’ recognition of periodically interrupted sentences was investigated. CI listeners’ scores declined drastically when the sentences were interrupted. The NH listeners showed a significant decline in performance with increasing spectral degradation using CI-simulated, noise-band-vocoded speech. It is inferred that the success of top-down processes necessary for the perceptual reconstruction of interrupted speech is limited by even mild degradations of the bottom-up information stream (16 and 24 band processing). A hypothesis that the natural voice-pitch variations in speech would help in the perceptual reconstruction of the sentences was not supported by experimental results.
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Boundary element analyses for sound transmission loss of panels J. Acoust. Soc. Am. Volume 127, Issue 2, pp. 829-840 (2010); (12 pages) Online Publication Date: 05 Feb 2010
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The sound transmission characteristics of an aluminum panel and two composite sandwich panels were investigated by using two boundary element analyses. The effect of air loading on the structural behavior of the panels is included in one boundary element analysis, by using a light-fluid approximation for the eigenmode series to evaluate the structural response. In the other boundary element analysis, the air loading is treated as an added mass. The effect of the modal energy loss factor on the sound transmission loss of the panels was investigated. Both boundary element analyses were used to study the sound transmission loss of symmetric sandwich panels excited by a random incidence acoustic field. A classical wave impedance analysis was also used to make sound transmission loss predictions for the two foam-filled honeycomb sandwich panels. Comparisons between predictions of sound transmission loss for the two foam-filled honeycomb sandwich panels excited by a random incidence acoustic field obtained from the wave impedance analysis, the two boundary element analyses, and experimental measurements are presented.
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