bluuebunny/crossref_metadata_2025 huggingface dataset:
Field calibration and check-out of an intensity measurement system using a simplified probe and a reference sound source
Authors: Campanella Angelo J.
Published: October 2005

There is need for a simple field check-out and calibration method for intensity measurement systems. Microphones, preamps, accompanying software, and data display constants can come in question after shipping, reassembly, and resetting. A simplified and standard method of exercising the total measurement system in a few minutes from microphone to display is presented. Microphones are mounted in the side-by-side configuration described by Per Bruel [2nd ICA (Sept. 1985)]. The sound field around an ACCULAB RSS-101 [Inter-Noise 91, 961–964] reference sound source (RSS) is sweep-surveyed [Tachibana, Inter-Noise 91, 1041–1044] over the five sides of an enclosing 1-m cube. This intensity survey method tends to cancel external noise and reflections in a wide variety of environments (shops, offices, outdoors, etc.). Instrument software is then set to display sound power in octave bands that can be directly compared with the laboratory calibration values known for the RSS-101. Sound power agreement within 1 to 2 dB will be found A-weighted and in the 250-Hz to 2-kHz octave bands with a nominal 1-in. (25-mm) microphone spacing. System performance in lower and higher frequency bands may be verified by repeating the test with appropriate microphone spacing.
Fiber-optic interferometric and intensity microphones
Authors: Zhou Chonghua , Letcher S. V. , Shukla A. , Singh Raman
Published: October 2005

Both interferometric and reflective-intensity fiber-optic microphones that exploit the fiber-optic advantages of electromagnetic-interference immunity, small size, and robustness are described. The interferometric microphone is a low-finesse Fabry–Perot sensor in which interference occurs between the light reflected from the end face of the fiber and that reflected from the membrane. The return signal is directed to a fast photodiode. The sensor has a flat frequency response from 50 Hz to 20 kHz. The phase shift is directly proportional to the amplitude of the diaphragm displacement with a broad dynamic range. The sensitivity is 2 rad/Pa and the minimum detectable pressure is less than 0.001 Pa when the phase shift is measured at quadrature. The intensity-reflection microphone simply measures the change of the intensity of light that is recaptured by the fiber after reflection from the diaphragm. It has the advantage of low cost and ease of sound reproduction. In a dynamic range of more than 60 dB, it is linear and has a flat frequency response from 50 Hz to 20 kHz. The minimum detectable pressure is 0.1 Pa. [This work is supported, in part, by NSF Grant No. MSS-9101514.]
Microphone placement in a binaural hearing manikin and its effects on IACC, reflectograms, and head-related transfer functions
Authors: Madaras Gary S. , Siebein Gary W. , Chiang Wei-hwa , Doddington Harold W. , Schwab Wilhelm K.
Published: October 2005

A binaural hearing manikin was constructed and then tested by comparing its head-related transfer functions to those of a human subject. The manikin was used to record impulse responses in eight northeastern United States concert halls as part of the recent Concert Hall Research Group Study. The placement of the manikin’s 1/2-in. Bruel & Kjaer microphones varied between two locations: inside the head (truncating the auditory canal) and outside the head (20 mm from the canal opening). For most measurements, no statistically significant differences were found between octave band IACC values for the two microphone locations. The manikin’s head-related transfer functions were measured in an anechoic chamber using both microphone positions and 324 source positions. Comparisons of reflectograms and head-related transfer functions show that the manikin’s head and the human subject’s head affect sound in similar ways. [Work supported by the National Science Foundation and Concert Hall Research Group.]
Acoustical measurements on stages of nine U.S. concert halls
Authors: Gade A. C. , Bradley John S.
Published: October 2005

A measurement tour of nine U.S. concert halls included acoustical measurements on the stage of each hall. Two teams (from the National Research Council of Canada, and the Technical University of Denmark) made measurements of the acoustical quantities suggested by Gade: the ‘‘support’’ family of measures describing the acoustical conditions for the musicians on stage. This paper compares the results from the two measurement teams and discusses the influence of hall designs. [Work partially supported by the Concert Hall Research Group.]
Statistical relations among architectural features and objective acoustical measurements of concert halls
Authors: Siebein Gary W. , Chiang Wei-hwa , Madaras Gary S. , Doddington Harold W. , Schwab Wilhelm K. , Bradley John S. , Gade A. C.
Published: October 2005

A statistical analysis of architectural features and detailed objective acoustical measurements made in eight concert halls and several multi-use rooms in their concert configuration will be presented. A method for evaluating the architectural features of rooms that affect their acoustical properties was developed. Architectural features of interest include both room average values and more-detailed subdivisions of surfaces including shape, volume, height, width, and sound absorption properties of materials. Regression modeling was performed for individual source–receiver paths as well as for entire rooms. Measurements data from all three teams were used in the models to assess the sensitivity of the models to expect variations in measurements. The results were compared to the previous work of Barron, Gade, and Hook among others. [Work supported by the National Science Foundation and Concert Hall Research Group.]
Dichotic interference effects in gap detection
Authors: Moore Brian C. J. , Shailer Michael J. , Black Mary Jane
Published: October 2005

Thresholds for detecting a temporal gap in a 20-Hz-wide band of noise (the target) were measured for the target alone, and in the presence of multiple 20-Hz-wide flanking bands presented to the opposite ear. The flanking bands caused gap thresholds to increase, and this effect was greater at higher levels of the flanking bands. The impairment to gap detection was greater when the flanking bands were comodulated with the target (i.e., had the same envelope) than when they were not comodulated, except at very low and high levels of the flanking bands. A series of supplementary experiments was conducted to investigate why the difference between comodulated and noncomodulated bands was reduced at high levels. The results suggest that this was not due to inter-aural crosstalk. It may have been partly caused by: (1) a central masking effect that reduced the effective sensation level of the target band at high levels of the contralateral flanking bands; (2) reduced independence of the flanking bands owing to broadening of the auditory filters at high levels. The results are discussed in terms of perceptual grouping processes.
Auditory and visual localization performance in a sequential discrimination task
Authors: Perrott David R. , Costantino Brian , Cisneros John
Published: October 2005

Four subjects were tested in a two-alternative, forced-choice, three down-one up adaptive paradigm in which two 200-ms signals were presented sequentially with a 200-ms interstimulus interval. The subject’s task was to indicate whether the second stimulus was to the right or left of the first. Tests were conducted with 57 dB (A-weighted), 1.0-kHz high-pass noise, the minimum audible angle (MAA) task, and with lights emitting 620 nm at a luminance level of 200 mL, the minimum visible angle (MVA) task. Localization performance in the MAA task was equal to or better than that obtained in the MVA task for all regions of the frontal field with only one exception, presentations at 0° azimuth. The implications of these results are discussed.
Prewhitening for intelligibility gain in hearing aid arrays
Authors: Link Michael J. , Buckley Kevin M.
Published: October 2005

In this article, prewhitening multimicrophone data prior to use in an optimum spatial-filter preprocessor for a monaural hearing aid is considered. Considering preprocessor capabilities, interference signal spectral content and intelligibility, it is argued that prewhitening is advantageous. This advantage is illustrated via simulation of a head worn array. Also, an effective procedure for the design of (prewhitening) digital filters is presented. The effectiveness of this filter design technique is shown by presenting several prewhitening filter design examples.
Speaking fundamental frequency, intensity, and rate of adventitiously profoundly hearing-impaired adult women
Authors: Leder Steven B. , Spitzer Jaclyn B.
Published: October 2005

No objective group data on speaking fundamental frequency (F0), intensity, or rate have been reported on the speech of adventitiously profoundly hearing-impaired adult women. Results of the present study showed that F0 was significantly higher, intensity significantly greater, and speaking rate significantly slower for 12 adventitiously profoundly hearing-impaired adult women than for 20 normal-hearing control subjects of the same sex and approximate age. Based on these objective data, aural rehabilitation is recommended for the improvement and monitoring of speaking F0, intensity, and rate abnormalities in the adventitiously profoundly hearing impaired.
F0 gives voicing information even with unambiguous voice onset times
Authors: Whalen D. H. , Abramson Arthur S. , Lisker Leigh , Mody Maria
Published: October 2005

The voiced/voiceless distinction for English utterance-initial stop consonants is primarily realized as differences in the voice onset time (VOT), which is largely signaled by the time between the stop burst and the onset of voicing. The voicing of stops has also been shown to affect the vowel’s F0 after release, with voiceless stops being associated with higher F0. When the VOT is ambiguous, these F0 ‘‘perturbations’’ have been shown to affect voicing judgments. This is to be expected of what can be considered a redundant feature, that is, that it should carry a distinction in cases where the primary feature is neutralized. However, when the voicing judgments were made as quickly as possible, an inappropriate F0 was found to slow response time even for unambiguous VOTs. This was true both of F0 contours and level F0 differences. These results reinforce the plausibility of tonogenesis, and they add further weight to the claim that listeners make full use of the signal given to them, even when overt labeling would seem to indicate otherwise.
From touch to string vibrations. III: String motion and spectra
Authors: Askenfelt Anders , Jansson Erik V.
Published: October 2005

This article describes an experimental study of the wave motion on the piano string and the corresponding spectra. The string motion was detected using an electrodynamic method. The measurements include typical initial string waveforms and spectra in the bass, mid, and treble at three dynamic levels (p–mf–f ). The interaction between hammer and string is also analyzed, including multiple contacts in the bass. The influence of hammer properties is investigated by shifting hammers and adjusting stiffness (‘‘voicing’’). A comparison between the string and bridge vibrations is presented, including the influence of the resonant tail lengths of the string (‘‘duplex scale’’). A brief account of the interaction between damper and string closes the study.
Habitat-induced degradation of sound signals: Quantifying the effects of communication sounds and bird location on blur ratio, excess attenuation, and signal-to-noise ratio in blackbird song
Authors: Dabelsteen Torben , Larsen Ole Næsbye , Pedersen Simon Boel
Published: October 2005

The habitat-induced degradation of the full song of the blackbird (Turdus merula) was quantified by measuring excess attenuation, reduction of the signal-to-noise ratio, and blur ratio, the latter measure representing the degree of blurring of amplitude and frequency patterns over time. All three measures were calculated from changes of the amplitude functions (i.e., envelopes) of the degraded songs using a new technique which allowed a compensation for the contribution of the background noise to the amplitude values. Representative songs were broadcast in a deciduous forest without leaves and rerecorded. Speakers and microphones were placed at typical blackbird emitter and receiver positions. Analyses showed that the three degradation measures were mutually correlated, and that they varied with log distance. Their variation suggests that the broadcast song could be detected across more than four, and discriminated across more than two territories. The song’s high-pitched twitter sounds were degraded more rapidly than its low-pitched motif sounds. Motif sounds with a constant frequency projected best. The effect of microphone height was pronounced, especially on motif sounds, whereas the effect of speaker height was negligible. Degradation was inversely proportional to microphone height. Changing the reception site from a low to a high position reduced the degradation by the same amount as by approaching the sound source across one-half or one-whole territory. This suggests that the main reason for a male to sing from a high perch is to improve the singer’s ability to hear responses to its songs, rather than to maximize the transmission distance. The difference in degradation between low and high microphone heights may explain why females, which tend to perch on low brush, disregard certain degradable components of the song.
Otoacoustic emissions from normal-hearing and hearing-impaired subjects: Distortion product responses
Authors: Gorga Michael P. , Neely Stephen T. , Bergman Brenda , Beauchaine Kathryn L. , Kaminski Jan R. , Peters Jo , Jesteadt Walt
Published: October 2005

Distortion product otoacoustic emissions (DPOAE) were measured in normal-hearing and hearing-impaired human subjects. Analyses based on decision theory were used to evaluate DPOAE test performance. Specifically, relative operating characteristic (ROC) curves were constructed and the areas under these curves were used to estimate the extent to which normal and impaired ears could be correctly identified by these measures. DPOAE amplitude and DPOAE/noise measurements were able to distinguish between normal and impaired subjects at 4000, 8000, and, to a lesser extent, at 2000 Hz. The ability of these measures to distinguish between groups decreased, however, as frequency and audiometric criterion used to separate normal and hearing-impaired ears decreased. At 500 Hz, performance was no better than chance, regardless of the audiometric criterion for normal hearing. Cumulative distributions of misses (hearing-impaired ears incorrectly identified as normal hearing) and false alarms (normal-hearing ears identified as hearing impaired) were constructed and used to evaluate test performance for a range of hit rates (i.e., the percentage of correctly identified hearing-impaired ears). Depending on the desired hit rate, criterion values of −5 to −12 dB SPL for DPOAE amplitudes and 8 to 15 dB for DPOAE/noise accurately distinguished normal-hearing ears from those with thresholds greater than 20 dB HL for the two frequencies at which performance was best (4000 and 8000 Hz). It would appear that DPOAE measurements can be used to accurately identify the presence of high-frequency hearing loss, but are not accurate predictors of hearing status at lower frequencies, at least for the conditions of the present measurements.
Comodulation masking release as a function of type of signal, gated or continuous masking, monaural or dichotic presentation of flanking bands, and center frequency
Authors: Fantini Deborah A. , Moore Brian C. J. , Schooneveldt Gregory P.
Published: October 2005

Thresholds were measured for detecting a signal centered in a narrow-band noise (NBN) masker (on-frequency band, OFB), for the OFB alone, and with two flanking bands (FBs) added to the OFB, one centered above and one below the OFB. The FBs were either correlated with the OFB or were independent and were presented either to the same ear as the signal plus OFB (monaural condition) or to the opposite ear (dichotic condition). The OFB and FBs were either gated with the signal, or were presented continuously. Three signal types were used: a pure tone; an NBN uncorrelated with the OFB; and an NBN correlated with the OFB. The signal was centered at 0.5, 2, or 6 kHz. Comodulation masking release was estimated either as the difference between threshold with the OFB alone and with the OFB plus correlated FBs [CMR(R−C)], or as the difference between thresholds using correlated and uncorrelated FBs [CMR(U−C)]. Although there were marked individual differences, positive CMR(R−C) values were found in all conditions for all three signal types. CMR(U−C) values were often larger than those for CMR(R−C), reflecting the fact that the uncorrelated FBs tended to produce interference effects, especially for the gated maskers, and at 6 kHz. Values of CMR were larger and more consistent across subjects for continuous than for gated maskers. For continuous maskers, the values of CMR tended to be smallest for the correlated-NBN signal. Results are discussed in terms of available cues and in terms of perceptual grouping mechanisms.
Applications of optimal time-domain beamforming
Authors: Collins Michael D. , Berkson Jonathan M. , Kuperman W. A. , Makris Nicholas C. , Perkins John S.
Published: October 2005

Applications to ocean acoustic data from a towed array and to speech processing are presented for an improved optimal time-domain beamformer, which involves optimizing over all possible source bearings and time series for multiple sources using simulated annealing. The convergence of the parameter search is accelerated by accepting time series perturbations only when the energy decreases. A comparison with the conventional delay-and-sum beamformer illustrates that the optimal beamformer handles larger receiver spacing and larger source-to-receiver ratio. Periodic ambiguities are essentially eliminated by using irregular receiver spacing and the improved search algorithm. Weak sources are handled with fractional beamforming. Noise cancellation is possible if the parameters of the noise are included in the search space. Two-dimensional localization is performed for nearby sources.
Stable marching schemes based on elliptic models of wave propagation
Authors: Knightly George H. , St. Mary Donald F.
Published: October 2005

A marching numerical scheme is applied to a far-field elliptic model of underwater wave propagation. General stability conditions are derived for the scheme in the case of varying parameter functions. Several examples are presented to demonstrate the capacity of the method to detect backscattered energy. In these examples the elliptic equation is cast as an initial value problem with assumed correct initial data. The ill-posed nature of initial value problems for elliptic problems is discussed.
On the crossing points of Lamb wave velocity dispersion curves
Authors: Zhu Qigao , Mayer Walter G.
Published: October 2005

Standard dispersion curves relating the phase velocity of Lamb waves to the frequency-plate thickness parameter fd indicate that in some cases symmetrical and antisymmetrical mode velocity curves cross each other. Viktorov’s equations are used to show that the crossing points are points where the dispersion curves are discontinuous so that no distinct Lamb mode exists for these particular velocity-fd combinations. Procedures are given to predict how many such points exist and where they are located in a given range of fd.
Experimental investigation of sediment effect on acoustic wave propagation in the shallow ocean
Authors: Rogers Andrew K. , Yamamoto Tokuo , Carey William
Published: October 2005

In shallow water the sediment layers have very strong effects on the propagation of acoustic waves. An effort to study the effects of the sediment has been made using 50- to 600-Hz continuous wave acoustic propagation data taken by Carey [‘‘Experimental verification and application of bottom shear modulus profile (BSMP) method,’’ Oceans ’91 Proceedings (1991)] at the Atlantic margin coring project (AMCOR) borehole 6010 off the coast of New Jersey combined with sediment properties measured at that site by Yamamoto et al. of the University of Miami Geoacoustic Laboratory using the bottom shear modulus profiler (BSMP) method. Excellent agreement was found between the model and data indicating the acceptability of BSMP sediment values as input for acoustic propagation studies. The introduction of shear or tangential stresses in the model was found to have no effect upon which modes propagated but only on their modal intensity. The higher the order of the mode the greater the penetration in the seafloor and the stronger the shear effect on the intensity. A sub-seafloor acoustic waveguide was investigated for the site and found to give possible explanation for enhanced intensity of modes that propagate strongly within that region. The intrinsic attenuation was determined using a method of matching modal intensity from model calculations with measured data. Biot theory was utilized and depth-dependent intrinsic sediment attenuation profiles were found for seven frequencies between 50 and 600 Hz.

The depth-averaged attenuation for the first mode at each frequency and the first mode that penetrates to 100 m was found. The frequency dependence of the mode attenuation was determined. The attenuation values agreed well with previous experimental acoustic reflection data taken by Mitchell and Focke [J. Acoust. Soc. Am. 67, 1582–1589 (1980)] but were much lower than values for attenuation in this frequency range determined by Hamilton’s [Geophysics 36, 266–284 (1971)] extrapolation of high-frequency laboratory and field measurements to the ≤1-kHz range. The effect of interface and volume scattering on transmission loss was determined using the acoustic model and a thin layer representation of the scattering interface. The result showed scattering to significantly contribute to energy loss beyond a kilometer or so and to have increased significance as frequency is increased.
Ultrasonic scattering from spherical shells including viscous and thermal effects
Authors: Anson L. W. , Chivers R. C.
Published: October 2005

An analysis for sound scattering and attenuation by shell structures immersed in fluids is outlined. While the form of the analysis permits easy comparison with previous results for simple spheres, it includes novel features of viscoelasticity in the shell and core materials (which may be fluid or solid), viscosity in the suspending fluid, and thermal effects. The computational procedure adopted is outlined with particular reference to the avoidance of numerical instabilities at very low and very high values of ‘‘ka.’’ The testing of the program by comparison with previous results on simpler systems is reported, and a limited selection of numerical results is given. The program has been used successfully over the range of ka values between 10−6 and 200, its robustness needs closer definition.
Scattering enhancement by supersonic resonances of cylindrical shells
Authors: Junger Miguel C.
Published: October 2005

Off-beam backscattering peaks associated with supersonic resonances of a simply supported unstiffened shell are evaluated by means of a relatively unfamiliar powerful analytical technique conceived by H. Lamb [London Math. Soc. Proc. 32, 11–20 (1900)] and further developed by P. W. Smith, Jr. [J. Acoust. Soc. Am. 34, 640–646 (1962)]. Boundary conditions leading to a simple solution have been selected because they have only a minor effect on coupling between the fluid and supersonic resonances. Cross-section peaks are effectively the same for the shear and compression families of shell resonances. Frequency dependence is associated only with a sin θi factor, θi being the angle of incidence measured from the cylindrical axis, associated with trace matching of the incident acoustic wave number and modal structural wave numbers. The level of sin θi for shells in water varies by no more than 1 dB with frequency. Within this range, the target strength maximum contributed by a shell resonance is TSr=20 log(2L)−9.4 dB±0.6 dB, where 2L is the shell length. Elastic parameters only affect the range of angles of incidence θi matched to the resonant mode’s configuration. The level is independent of the cylinder radius, though of course resonance frequencies are not.
Reply to: ‘‘Comment on ‘Temporal modulation transfer functions in patients with cochlear implants’ ’’ [J. Acoust. Soc. Am. 9
3, 1649–1650 (1993)]
Authors: Shannon Robert V.
Published: October 2005

Kohlrausch [J. Acoust. Soc. Am. 93, xxxx (1993)] points out that modulation detection improves as a function of sinusoidal carrier level in normal-hearing listeners as well as in listeners with cochlear implants. While his point is well-taken, additional factors complicate the comparison between acoustic and electric stimulation: (1) the lack of spectral resolution in implant listeners, and (2) amplitude transformations necessary to properly compare acoustic and electric stimulation.
The fourth-order cumulant spectrum of a filtered linear frequency-modulated source in motion
Authors: Dwyer Roger F.
Published: October 2005

The goal of a sonar system is to extract information from received pressure signals emanating from a source. Typical methods employed are usually first-order or second-order functions of the received pressure. This paper develops an analytical method to extract information from the received pressure using the fourth-order cumulant spectrum. The analysis begins with a far-field solution of the wave equation for an acoustic point source in motion. The source is modeled as a filtered linear frequency-modulated signal. The objective is to extract the impulse response of the source characterized by the filter. But due to the motion of the source the response of the extracted filter is modified by the Doppler component. For the given source model, a scaled Doppler component, and both the phase and a scaled magnitude of the Doppler modified filter, can be extracted directly from the fourth-order cumulant spectrum assuming a large time bandwidth product.
The modified window correlation technique for estimation of arbitrary time delays in multichannel, multievent systems
Authors: Jan Ben-Chung , Henriquez Juan A. , Riemer Terry E. , Trahan Russell E.
Published: October 2005

In a previous paper, the so-called window-correlation technique [Callison et al., J. Acoust. Soc. Am. 81, 1000–1006 (1987)], a method for estimating arbitrary time delays in multichannel, multievent systems, was presented. This method windows a portion of the data record in both channels of a two-channel system, and computes the cross correlation of the data segments. However, the CPU time consumed by this technique becomes prohibitively long for most real time applications. This paper introduces a modification to the above method called the modified window-correlation technique (MWCT) that reduces the CPU time considerably by windowing only one channel and cross correlating the data record in the frequency domain. The simulation results for a two-channel, two-event system show that the MWCT can estimate event delays very accurately down to a signal-to-noise ratio (SNR=mean-square value of signal/mean-square value of noise) of 0 dB.
Analyses of Mössbauer mechanical measurements indicate that the cochlea is mechanically active
Authors: Brass David , Kemp D. T.
Published: October 2005

Using the results of Mössbauer measurements, mechanical activity in the cochlea was tested for by comparing the measured basilar membrane (BM) transverse velocity amplitude with that calculated for a lossless mechanically passive system, derived from the measured BM velocity phase. If the cochlea is considered to be a lossless mechanically passive system, then the transverse velocity amplitude can be calculated from the group velocity and the relative variation of stiffness along the BM. The group velocity can be derived from the Mössbauer phase measurements, and the relative variation of stiffness along the BM can be derived from the frequency map of the cochlea. Making some general assumptions, the actual transverse velocity amplitudes are then compared from the Mössbauer amplitude data with those derived from the Mössbauer phase data, to determine if there is a significant transverse velocity gain. This operation was performed on several sets of Mössbauer data. From Mössbauer data showing sharp tuning, differences were found of up to 40 dB between the actual transverse velocity amplitude and the calculated lossless passive transverse velocity amplitude derived from the phase data.

Examination of the assumptions made during the calculation of the lossless passive transverse velocity amplitude showed that none could account for a 40-dB transverse velocity gain. Thus, it is concluded that this transverse velocity gain can only be accounted for by the contribution to the amplitude of the transverse BM velocity by mechanically active elements along the cochlear duct. From Mössbauer data showing much less sharp tuning, it was found that the actual transverse velocity amplitude was approximately equal to the calculated lossless passive transverse velocity amplitude, basal to the characteristic place. This result is attributed to the disabling of the mechanically active elements along the cochlear duct. Apical to the characteristic place, the actual transverse velocity amplitude is shown to behave in a manner that suggests that the effective damping increases markedly in this region. This increase in effective damping is not necessarily due to an increase in viscous damping but could be due to any mechanism removing energy from the BM traveling wave. As an example, this paper discusses how this effective damping could be accounted for the transfer of energy to another mode of vibration at the characteristic place.
Effects of varying signal and noise levels on pure-tone frequency discrimination in humans and monkeys
Authors: Sinnott Joan M. , Brown Charles H.
Published: October 2005

This study examined the effects of presenting various signal and noise levels on pure-tone frequency difference limens (FDLs) in humans and monkeys. Signal levels for two frequencies of 500 Hz and 4 kHz were varied both in quiet and in noise. For the lower frequency of 500 Hz, the two species exhibited qualitatively divergent FDLs as level increased in quiet: FDLs decreased at high levels for humans, but increased for monkeys. In noise, differences between human and monkey FDLs were less pronounced. At 500 Hz, monkeys exhibited slightly lower FDLs in noise than in quiet. At the higher frequency of 4 kHz, under all conditions, FDLs for both species were qualitatively similar. Results are discussed in relation to (1) different use of ‘‘temporal’’ coding mechanisms by humans and monkeys at 500 Hz, and (2) similar use of ‘‘rate-place’’ coding mechanisms at 4 kHz.
Effect of spectral envelope smearing on speech reception. II
Authors: ter Keurs Mariken , Festen Joost M. , Plomp Reinier
Published: October 2005

This paper describes two experiments on the effect of reduced spectral contrast on the speech-reception threshold (SRT) for sentences in a background of interfering sound. Signal processing is performed by smoothing the envelope of the squared short-time fast Fourier transform by a convolution with a Gaussian-shaped filter, and overlapping additions to reconstruct a continuous signal. In the first experiment the effect of reduced spectral contrast on the SRT for male speech is investigated and compared with previously obtained results for female speech [ter Keurs et al., J. Acoust. Soc. Am. 91, 2872–2880 (1992)]. Spectral energy is smeared over bandwidths of 1/8, 1/4, 1/3, 1/2, 1, 2, and 4 oct. The results show that, despite the differences in spectral pattern between male and female voices, the SRT in noise increases similarly for both voices for smearing bandwidths over 1/3 oct. In terms of the ripple density of the spectral envelope the results indicate that the range of lower spectral modulations, up to a limit of about 1.5 periods/oct, is sufficient for the intelligibility of speech in interfering sounds. In the second experiment the extent of the threshold difference between a speech masker and a noise masker is investigated for spectral smearing bandwidths of 1/2, 1, and 2 oct. The release from masking found for the speech masker relative to the (steady-state) noise masker decreases with spectral envelope smearing.
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s: Evidence for sensitivity to echo phase in bat sonar
Authors: Moss Cynthia F. , Simmons James A.
Published: October 2005

Echolocating bats, Eptesicus fuscus, were trained in two distinct behavioral tasks to investigate the images they perceive of a sonar point target. In the first task, bats were trained in a two-alternative forced-choice procedure to detect electronically simulated target echoes at a range of approximately 57 cm. Half of the trials in the detection task contained echoes from a stationary target (simulated by a fixed echo delay) and half contained echoes from a jittering target (simulated by an echo delay alternating between two time values over successive sonar emissions). In the second task, bats were trained in a two-alternative forced-choice procedure to discriminate between electronically simulated stationary and jittering targets, centered about a range of 57 cm. Both target detection and target jitter discrimination performance were assessed as a function of jitter magnitude, with jitter values ranging from 0–60 μs (corresponding to a change in distance of 0 to 10.3 mm). In both detection and discrimination tasks, the bat’s performance changed cyclically with the magnitude of echo jitter. Specifically, when the phase of the playback echoes was unchanged, performance levels were poorest at 0 and 30 μs, and when the phase of the echoes alternated by 180 deg from one to the next, performance levels were poorest at 15 and 40–50 μs. The results suggest that Eptesicus is sensitive to the phase reversal of echoes and thus have implications for assessing receiver models of echolocation.
Detecting temporal asynchrony with asynchronous standards
Authors: Zera Jan , Green David M.
Published: October 2005

The ability of listeners to detect asynchrony in either the temporal onset or offset of individual components in a multicomponent complex was investigated. A standard stimulus was created having a simple kind of asynchrony by linearly delaying successive components of a complex stimulus. Additional asynchrony (the signal) was created by altering the temporal position of a single component of the complex relative to its temporal position in the standard. Asynchrony discrimination was measured for harmonic signals (experiment I) and for complexes in which the components were logarithmically spaced in frequency (experiment II). For the onset condition, asynchrony in either the harmonic or logarithmic complex greatly increased the difficulty in detecting the temporal asynchrony of a single component. For the offset condition, asynchrony discrimination was not greatly affected by asynchrony in the standard.
A note on the prediction of sound intensity
Authors: Pan J.
Published: October 2005

Modal analysis has long been used to describe the acoustic characteristics of enclosures with absorptive boundaries. However, it is demonstrated in this paper that this analysis fails in predicting the sound intensity distributions in the enclosures. This is due to the boundary damping approximation involved in this analysis, which has a misleading effect upon the sound absorption material locations. To illustrate this point, the sound intensity in a one-dimensional pipe is calculated by modal analysis and the result is compared with the exact solution.
Comment on ‘‘Temporal modulation transfer functions in patients with cochlear implants’’ [J. Acoust. Soc. Am. 9
1, 2156–2164 (1992)]
Authors: Kohlrausch Armin
Published: October 2005

Shannon [J. Acoust. Soc. Am. 91, 2156–2164 (1992)] reports a strong decrease of modulation detection thresholds with cochlear implant patients for an increase of the stimulus level. His conclusion that this result differs from the findings for acoustical stimulation in normal-hearing subjects needs some clarification. His statement about acoustic stimulation is based on modulation detection measurements for noise carriers, where, indeed, thresholds in normal-hearing subjects are nearly independent of level, at least for carrier levels of more than 20 dB SL. For sinusoidal carriers, however, the modulation detection thresholds decrease with increasing carrier levels over the entire hearing range, similar to his findings with implant listeners.
Error estimates for free-decay damping measurements
Authors: Orduña-Bustamante Felipe , Boullosa Ricardo R.
Published: October 2005

An estimate is presented of the errors introduced by digital sampling of an analog time signal when the logarithmic decrement method is used to measure damping. Some computer simulations were made in order to verify that the error estimate is adequate. Other factors that influence the error in the measured damping ratio, such as noise and the value of the damping ratio, are also studied by computer simulation. It is shown that, on average, the digital sampling error depends mainly on the total number of sampled points, but that it is affected also by the ratio between the sampling frequency and the frequency of the signal. Digital sampling errors combined with the errors introduced by different signal-to-noise ratios impose bounds to the values of the damping ratio that can be measured accurately. It is also shown that the optimun number of data points, when digital sampling error is combined with noise, varies according to the signal-to-noise ratios and the value of the damping ratio. A brief discussion of other decay methods compared to the logarithmic decrement method is also given.
The sulcus connection. On a mode of participation of outer hair cells in cochlear mechanicsa)
Authors: de Boer Egbert
Published: October 2005

Motile outer hair cells (OHCs) can only participate well in cochlear mechanics when one end of the hair cells is more restrained in its movements than the other. On this thought a model of the organ of Corti (OoC) is developed in which (in every cross section) the tectorial membrane (TM) is considered to consist of two stiff segments connected by a hinge. Movements of the TM then induce movements of the fluid contained in the inner spiral sulcus (ISS) and the fluid dynamics of the sulcus will play an important part in restraining the top ends of the hair cells. Since the model is meant to reflect low-level phenomena, it is conceived as linear. For simplicity, it operates in the long-wave mode. The sulcus is considered as a narrow channel that is closed at both ends. Since the sulcus fluid is an independent energy-storage element of the system, the entire cochlear model, consisting of two main fluid channels, the organ of Corti (OoC), the three-element TM and the sulcus, and equipped with motile OHCs, can be described by a fourth-order differential equation, with four boundary conditions. It is shown that the model can ideally produce response curves that reflect more than 40 dB of cochlear gain. When the actual effect of the sulcus is taken into account, however, the maximum gain is reduced to approximately 30 dB. The main reason for the reduction is that the sulcus cannot produce a large enough pressure to restrain the upper ends of the OHCs.
Shallow water active sonar environmental acoustic issues
Authors: Carey William M. , Cable Peter G. , Cox Henry
Published: October 2005

The propagation and reverberation of sound in shallow water under downward refracting conditions is governed by the geoacoustic and scattering properties of the bottom. The active bistatic sonar equation for this case is developed and it is shown that, when the transmission to the target and the bottom are approximately the same for reverberation-limited conditions, a simpler parametric form of this equation results. That is, the signal excess depends on the effective bottom backscattering strength, the coherent signal gain, and a reverberation area for a given target strength and threshold. A parametric analysis is presented that focuses on the frequency-dependent characteristics of the bottom backscattering strength and the effect of coherence on receiving array aperture. Recent reverberation and signal coherence measurements are reviewed and the needs for additional studies delineated. The coherence issue for the waveguide is stressed and the measurement problem of coherence is discussed and compared to the measurement of signal versus the receiver length in wavelengths.
Hybrid magnetostrictive/piezoelectric Tonpilz transducer
Authors: Butler John L. , Butler Alexander L. , Butler Stephen C.
Published: October 2005

A self-tuned magnetostrictive/piezoelectric hybrid Tonpilz transducer which produces enhanced motion at one end and canceled motion at the other is presented. The particular hybrid Tonpilz transducer described here consists of two 5.4 in. (13.65 cm) diameter circular pistons with a central mass separating a Terfenol-D magnetostrictive tube and a Navy type I piezoelectric ring stack driver. The in-water mechanical resonance and hybrid electrical self-tuning occurs in the vicinity of 4.25 kHz where a 15-dB front-to-back pressure ratio was obtained under array loading conditions. As a result of the shared electrical stored energy, an improved effective coupling coefficient is obtained.
Individual differences in vowel production
Authors: Johnson Keith , Ladefoged Peter , Lindau Mona
Published: October 2005

It is often assumed that a relatively small set of articulatory features are universally used in language sound systems. This paper presents a study which tests this assumption. The data are x-ray microbeam pellet trajectories during the production of the vowels of American English by five speakers. Speakers were consistent with themselves from one production of a word to the next, but the articulatory patterns manifested by this homogeneous group were speaker specific. Striking individual differences were found in speaking rate, the production of the tense/lax distinction of English, and in global patterns of articulation. In terms of a task-dynamic model of speech production, these differences suggested that the speakers used different gestural target and stiffness values, and employed different patterns of interarticulator coordination to produce the vowels of American English. The data thus suggest that, at some level of speech motor control, speech production tasks are specified in terms of acoustic output rather than spatiotemporal targets or gestures.
Lower pharyngeal wall coarticulation in VCV syllables
Authors: Parush A. , Ostry D. J.
Published: October 2005

Speech movements of the lower pharyngeal wall were recorded in two subjects using pulsed-echo ultrasound. The focus of the study was the pattern of coarticulation of pharyngeal wall movements. Using nonsense utterances as test material, both anticipatory and carryover coarticulatory effects were observed. The identity of the final vowel in VCV sequences affected the kinematic characteristics of the initial VC transition. Both the amplitude and the duration of the movement between the initial vowel and the consonant were greater when the final vowel was /u/ rather than /ɑ/. Similarly, the initial vowel affected the kinematic characteristics of the final CV transition. The amplitude of the movement from the consonant to the final vowel was greater with the initial vowel /u/ as opposed to /ɑ/. The coarticulatory patterns observed in this study are similar to those previously reported for the tongue dorsum and upper pharynx [Parush et al., J. Acoust. Soc. Am. 74, 1115–1125 (1983); Parush and Ostry, J. Acoust. Soc. Am. 80, 749–756 (1986)].
The effect of compressibility on acoustic streaming near a rigid boundary for a plane traveling wave
Authors: Qi Quan
Published: October 2005

The effect of compressibility on acoustic streaming near a rigid boundary is studied. The goal is to resolve an existing inconsistency: while the compressibility is a necessary condition for the propagation of acoustic waves, previous analyses of acoustic streaming are limited to incompressible fluids. Plane traveling waves are treated in the main text and the main results for a plane standing wave are presented in Appendix B. It is shown that the inclusion of compressibility leads to a larger streaming velocity outside the boundary layer. The effect is significant in gases, but not in liquids.
Increased accuracy in the application of the Sommerfeld–Watson transformation to scattering of acoustic waves from spherical shells
Authors: Rumerman M. L.
Published: October 2005

The Sommerfeld–Watson transformation (SWT) is applied to the normal mode series for scattering of an incident plane acoustic wave from a spherical elastic shell to express the scattered field in terms of its wave components. The ‘‘geometric’’ part of the scattered field is extracted and expressed as a line integral which is evaluated by shifting to a path of steepest descent (SDP). When deforming the path, one or more poles of the integrand may be crossed and their residues must be accounted for, resulting in discontinuities in the computed field unless the effects of these poles are also included in the evaluation of the SDP integral. Although the asymptotic form of the Legendre function used in the evaluation of the SDP integral is singular and invalid near the direction of backscatter, it is found that its use yields a finite and correct result for the pole correction, as well as for the saddle point approximation. This approach is applied to a thin steel shell in water using thin shell theory, and good agreement is obtained for scattering into the backward half-space between the results of the normal mode series, using shell theory, and the modified SWT calculation for ka≳3.
An iterative solution scheme for the exterior Neumann problem for Δ+κ2 using finite elements in a bounded domain
Authors: Shirron Joseph J.
Published: October 2005

A scheme is given for approximating the solution of the exterior Neumann problem for the Helmholtz equation that employs the finite element method in an artificially truncated domain. A sequence of approximate solutions is produced by iteration. The current iterate is used to compute a boundary condition on the truncating surface for the next iterate. While precise conditions under which the proposed method is convergent are not yet known, numerical experiments indicate its robustness.
Acoustic diffraction spectrum of a SiC fiber in a solid elastic medium
Authors: Sinclair A. N. , Addison R. C.
Published: October 2005

An analytical model has been developed to compute the diffraction spectrum and phase of a plane compression wave incident on a long fiber embedded in a solid elastic matrix. The spectrum is computed as an expansion of normal modes, thereby permitting insight into the resonant conditions set up in the fiber. The technique was used to model a reinforcing fiber, consisting of a graphite core surrounded by a vapor-deposited SiC layer, all consolidated within a titanium alloy matrix. Reasonable agreement was noted between experiment and theoretical prediction of the resonant frequencies, although bandwidth limitations of the experiments severely restricted the number of resonance lines that could be included in the comparison. A series of numerical studies was conducted to assess the characteristics of each observed resonance. The technique has potential application to characterization of the interfacial layer that links reinforcing fibers to their matrix.