bluuebunny/crossref_metadata_2025 huggingface dataset:
Perceived rate of randomly modulated sounds
Authors: Kado Hisashi
Published: January 2006

The common measures used in evaluating the magnitude of a noise (Lx, Leq, and etc.) correspond to first- and second-order statistics of the distribution of sound levels. There are, however, perceived differences between sounds that have the same values of these statistics. It is therefore necessary to consider the spectral properties of the waveform envelope which correspond to higher-order statistics. In the present experiments the subject compares a test stimulus and a reference stimulus and selects the one which appears to be fluctuating more rapidly. The reference stimulus is a noise whose waveform envelope is a noise with a low-pass cutoff which varies from 0.125 to 8 Hz. The point of subjective equality is determined by adjusting the modulation frequency of the test stimulus, a sinusoidally modulated noise, in an adaptive procedure. The results indicate that when the median frequency of the spectrum of the envelope is above 1 Hz, it is a reasonably good predictor of the subjective rate, when the median is below 1 Hz the mode is a better predictor. [Work supported by NINCDS.]
Predicting masking by combinations of simultaneous maskers
Authors: Lutfi Robert A.
Published: January 2006

In a previous paper [Lutfi, J. Acoust. Soc. Am. 73, 262–267 (1983)], I proposed the following rule for predicting masking by combinations of simultaneous maskers; Xc = [X1p + X2p + ⋯Xnp]1/p, where X1, X2,…, Xn are the individual masking effects of the maskers, Xc is the combined effect, and 0.20 < p < 0.33. In this paper, the rule is used to predict the results of several experiments in which the individual effects of maskers in the combination are known [Bilger, J. Acoust. Soc. Am. 31, 1107–1109 (1959); Green, J. Acoust. Soc. Am. 41, 1517–1525 (1967); Canahl, J. Acoust. Soc. Am. 50, 471–474 (1971); Patterson and Nimmo-Smith, J. Acoust. Soc. Am. 67, 229–245 (1980)]. With the possible exception of the study by Bilger, a single value of p (= 0.3) yields predictions in good agreement with the data. The rule also predicts some general characteristics of the results of studies in which the individual effects of the maskers are not known but can be estimated [Johnson-Davies and Patterson, J. Acoust. Soc. Am. 65, 765–770 (1979); O'Loughlin and Moore, J. Acoust. Soc. Am. 69, 1119–1125 (1981)]. [Work supported by NSF and NIH.]
The effect of signal SPL on interaural time discrimination
Authors: Cohen Marion F. , McClave Anne E. , Pollanck Patricia Gregorio
Published: January 2006

Interaural time jnds for a partially masked sinusoidal signal were measured as a function of signal SPL, with the signal presented at a constant signal-to-noise ratio of approximately 10 dB above its masked detection threshold. The 250-, 500-, or 1000-Hz signal ranged in level from 13 to 58 dB SPL. The masker was a continuous white noise. A block up-down two-interval forced-choice procedure was used. Results show that interaural time discrimination improves with increasing signal SPL despite the constant S/N ratio, indicating that signal SPL is a primary determinant of precision of interaural time discrimination. When these data are compared with jnds measured at the same signal SPLs but with no masker, they are nearly identical at 500 Hz. At 250 and 1000 Hz they are nearly identical at the higher signal levels for two of the three subjects. At the lower signal levels, the jnds measured in the presence of the masker are somewhat larger, but demonstrate a similar decrease with increasing signal level. The differences noted across frequency may be due to the effect of the internal noise. [Work supported by NIH Grant No. 5 R01 NS16802.]
Depth dependence of the temporal characteristics of cw propagation in the Arctic Ocean
Authors: Giellis G. R. , Yang T. C. , Votaw C. W.
Published: January 2006

Previous work by P. Mikhalevsky [J. Acoust. Soc. Am. 70, 1717–1722 (1981)] investigated the temporal characteristics of cw signals propagating in the Arctic Ocean. The previous work was limited to receivers at a depth of 91 m. In this paper, the temporal characteristics will be extended to 28 hydrophones in a vertical array with depths from 30 to 960 m. The data were taken during the FRAM IV experiment in April 1982. The signals were generated at the Tristen ice camp and received at the FRAM IV ice camp located north of Spitzbergen. The range of approximately 275 km was over the Mid-Ocean Ridge; the average water depth along the path was 3000 m. Most of the data examined are at a high signal-to-noise ratio (> 10 dB) which makes it possible to study the temporal fluctuation and the statistics of the cw signals at different depths. By employing a mode filtering technique, the temporal dependence of the amplitude and phase of the lowest order mode has been investigated. The drift rates of the ice camps were negligible during the transmission periods and hence the results apply to transmission between essentially fixed stations.
Ultrasonic reflection measurements of floating ice
Authors: Brower Neal G. , Ng Kin W. , Mayer Walter G.
Published: January 2006

An experimental investigation of reflection of ultrasonic, bounded beams from floating ice has been conducted. Ultrasonic frequencies and ice thicknesses corresponding to frequency thicknesses of between 2 and 5 mm⋅MHz are considered. Nonspecular reflection profiles are obtained using schlieren techniques. These nonspecular profiles are compared to theoretical profiles calculated for water/ice plate/air systems. [Work supported by ONR.]
Acoustic scattering from Arctic sea-ice ridges using high-angle PE
Authors: Greene Robert R.
Published: January 2006

Long-range propagation loss runs with the high-angle PE model indicate that scattering of sound into the ocean bottom by sea-ice roughness can account for the frequency dependence of transmission loss in the Arctic. But differences between the slopes of the data and predicted transmission loss curves suggest the presence of an additional loss mechanism in the ice.
Auditory evoked potential asymmetries in right- and left-handed listeners
Authors: Hood L. J. , Martin D. A. , Berlin C. I.
Published: January 2006

While Decker [ASHA 23, 803 (1981)] reported no relationship of handedness to auditory tract preference in the auditory brainstem response, we know of no data for the middle and late potentials. This study examines the relationship of binaural interaction and lateral asymmetry in the middle and late potentials to handedness. Binaural, right monaural, and left monaural 100-μs pulses were presented to six right-handed and six left-handed normal-hearing subjects in four conditions of earphone and electronics rotations. Middle and late potential data were analyzed statistically using a time series ANOVA. Lateral asymmetries and binaural interaction were derived by subtracting left from right responses and summed monaural from binaural responses, respectively. Data indicate individual differences in the amount and direction of asymmetry. The characteristics of binaural interaction and lateral asymmetry, the relationship to handedness, and the influence of stimulus and recording asymmetries will be described for the middle latency and late potentials. [Supported by NINCDS ♯NS11647, the Louisiana Lions Eye Foundation and the Eye and Ear Foundation.]
Complete reconstruction of the auditory system: Adult gerbil
Authors: Harris David M. , Timmins Michael
Published: January 2006

The object of our work is the reconstruction of the entire auditory system of the adult gerbil. The methods to be described include perfusion with a vital stain, fixation and decalcification, photography of the cut surface of the tissue block (entire head) during collection of serial sections, tracing and digitizing structures of interest from the photographic data and computer graphic reconstructions. The image derived from digitizing is confirmed by histological staining and examining each section. Such reconstructions allow visualization in three dimensions, or in horizontal, coronal, or sagittal sections. These data allow for exact measurement of structures and their topological interrelationships. The auditory system depicted will include nervous pathways as well as inner and middle ear structures. A quantitatively exact reconstruction permits a precise determination of multiple physiological recording sites, for us a most important application. [Work supported by NIH.]
Recovery from saturation in lateralization of high-frequency stimuli
Authors: Hafter Ervin R. , Buell Thomas N.
Published: January 2006

Lateralization of trains of high-frequency clicks improves with both the number of dicks in a train (n) and interclick interval (ICI). Functions relating log threshold to log n are linear; however, the absolute values of the slopes decrease from the optimal—0.5 toward 0.0 with shorter ICIs (generally 10 ms or less). We have argued that shallower slopes are the result of a form of rate-dependent saturation which reduces the amount of binaural information derived from successive clicks in the train. The decay of the proposed saturation was measured here by presenting trains with an ICI of 2.5 ms but with each train broken by either one or three pauses. Of interest was the duration of the break needed to fully recover from saturation. A pause of 7.5 ms proved to be sufficient, implying that the saturation responsible for shallow slopes in log-threshold, log-n plots is the result of an active process rather than of fatigue. [Supported by NIH.]
Adaptive, forced-choice, loudness-balance tests in normal listeners
Authors: Wier Craig C. , Huckins Elizabeth , Schlauch Robert S.
Published: January 2006

An adaptive, two-interval, forced-choice procedure (Jesteadt, 1980) was used to obtain ABLB and modified-MLB loudness-growth functions for a partially masked, 2000-Hz tone burst. Subjects were six normally hearing adults. Steinberg and Gardner (1937) and Stevens and Guirao (1967) showed that the comparison of the loudness of a tone in quiet with the loudness of the same tone partially masked produced “recruiting” loudness-growth functions from normally hearing listeners. For the ABLB condition in the present study, a pulsed, wideband masking noise was presented to only one ear. The level of the tone in noise was varied and compared with the level of the tone in quiet presented alternately to the opposite ear. In the modified-MLB procedure, the tone was presented monaurally, alternately in quiet and in the presence of a pulsed, wideband masking noise to the same ear. Two noise levels (3 and 33 dB N0) and eight standard-tone levels were used. The loudness-growth functions obtained binaurally and monaurally were virtually identical. The results of these experiments validate Jesteadt's adaptive procedure for the measurement of loudness growth. A comparison of the ABLB and MLB data suggests that a similar, partial-masking procedure could be used to develop a monaural, single-frequency, loudness-balance test for clinical use. [Work supported by DRF and NINCDS.]
A comparison of the relative merits of four psychophysical procedures
Authors: Shelton Brian R. , Scarrow Irene
Published: January 2006

The efficiency of four psychophysical procedures to estimate the detection threshold of a brief 1000-Hz sinusoid presented in noise was assessed in naive observers. Six threshold estimates were obtained for ten subjects in 50-trial adaptive runs using each of four procedures: two-alternative adaptive staircase, two-alternative maximum-likelihood, three-alternative adaptive staircase and the three-alternative maximum-likelihood method. The four procedures produced equivalent thresholds, although the staircase procedures provided biased measurements in the initial runs. A trial-by-trial analysis of the variability of threshold estimates revealed a marked advantage to the use of three alternatives with the adaptive staircase, but not with the maximum-likelihood procedures. The data provide information regarding the relative efficiency of the four procedures, and recommend that particular procedures should be used in specific instances. In general, the commonly used two-alternative adaptive staircase is the least efficient procedure of the four, and the three-alternative adaptive staircase provides the most rapid and reliable threshold estimates. [Work supported by NSERC.]
Frequency spread of TTS in humans and squirrel monkeys exposed to industrial noises
Authors: Nielsen Donald W. , Brandt Diane , Elliott Donald N. , Boisvert Peter , Hunter-Duvar Ivan
Published: January 2006

We have been investigating the squirrel monkey (Saimiri sciureus) as a suitable animal model for testing the effects of noise on hearing. Our previous research has shown a similarity between human and squirrel monkey temporary threshold shift (TTS) growth with time in the noise. The goal of this study was to measure the frequency spread of TTS to five different industrial noises for both humans and squirrel monkeys and to compare the results. Fifteen human subjects and six squirrel monkeys were exposed to each of the five noises for 8 h, and TTS was measured at ten different frequencies ranging from 250 Hz to 8 kHz. In the experiments with humans, five frequencies were tested twice (counterbalanced) after each 8-h session. In the squirrel monkey experiments, a complete 8-h exposure was required for each frequency tested, since only one frequency could be measured each time the monkeys were removed from the noise. The results indicate that in predicting human TTS from the squirrel monkey TTS for these exposures, TTS would be overestimated at 750 Hz and 1 kHz and underestimated at 5.6 kHz, whereas the TTS at other frequencies would correspond closely. [Work supported by NIOSH.]
Vibrotactile sensation as a function of hand preference
Authors: Verrillo Ronald T.
Published: January 2006

A psychophysical matching procedure was used to assess the subjective magnitude of vibrotactile stimuli in right-, left-, and ambi-handed subjects. No threshold differences were noted. At suprathreshold levels of stimulation, enhancement of sensation by a preceding stimulus was the same in all groups when both stimuli were presented to either hand ipsilaterally. However, when stimulus pairs were presented contralaterally, only right- and left-handers showed enhancement and suppression effects. No contralateral effects were observed in ambi-handers. The results suggest that bilateral neural connections in right- and left-handers may be different from those of ambi-handers.
Sound field fluctuation in shallow water waveguide
Authors: Clay C. S. , Wang Y. Y. , Shang E. C.
Published: January 2006

The sound intensity fluctuation induced by the fluctuation of the surface in a shallow water waveguide was analyzed by using the adiabatic approximation of the normal mode theory. The sound field fluctuation can be categorized into two parts: (1) The phase fluctuation of the normal mode corresponds to the fluctuation of the horizontal component of the wavenumber (I. Tolstoy and C. S. Clay, Ocean Acoustics, pp. 222–236). (2) The fluctuation of the eigenfunction corresponds to the fluctuation of the vertical component of the wavenumber and the fluctuation of the local water depth and the local source and receiver positions. The randomness of the field intensity defined as η = SQR [(Ī2 − Ī2)/Ī2] was calculated. It was found that the fluctuation of the second category can be the dominant comparing with the first category. The calculated result also agrees well with the experimental result conducted in a water tank [Y. Y. Wang et al., Chin. Phys. 2, 515 (1982)].
An investigation of acoustic interaction with the ocean bottom from experimental time series generated by explosive sources
Authors: Knobles David
Published: January 2006

Received time series from explosive sources in a thin sediment environment are compared to simulated time series calculated by a ray theory model. The comparisons are made in various frequency bands to aid in the identification of sediment penetrating arrivals by taking advantage of the frequency dependence of the absorption of the sediment. For large ranges, most of the received energy refracts within the sediment. As the range decreases, the fraction of received energy due to reflections at the water-sediment interface increases. Discrepancies between the experimental and simulated time series will be presented and interpreted in terms of reflections from thin layers within the sediment and nonspecular basement reflections. [Work supported by Naval Ocean Research and Development Activity.]
Seafloor acoustic backscatter measurements using SEA BEAM
Authors: de Moustier C. P.
Published: January 2006

Acoustic backscatter from the seafloor is measured using the 12-kHz SEA BEAM multibeam echo sounding system aboard the R/V THOMAS WASHINGTON. During a cruise in the eastern tropical Pacific, a very dense survey of a 15- ×20-mile manganese nodule area was carried out, and the acoustic returns from the 16 beams were recorded digitally on magnetic tape. Using the intensity of the specular return for each ping, a reflectivity map of the area was produced. The patchiness of the nodule coverage is evidenced by definite highs and lows in the reflectivity pattern. This map later served as the basis for a near bottom survey using the Deep Tow instrumentation package of the Marine Physical Laboratory and there is remarkable agreement between the two sets of measurements. Bottom photographs taken throughout the area also confirm the nodule distribution. This paper compares and discusses these results.
Airport related residential acoustical insulation demonstration project
Authors: Weissenburger J. T.
Published: January 2006

Faced with making a choice between acquiring residences and clearing the property versus acoustically insulating them, the St. Louis Airport Authority undertook a demonstration project to develop firm data on the cost and value of retrofit acoustical insulations for residences in the airport environs. Six wood frame homes, typical of most of the housing stock in the area, were modified using standard available construction techniques and materials in an effort to improve their insulation against aircraft noise. Noise exposure measurements were made before and after modifications. This paper reports the details of the retrofit modifications employed and their effectiveness in reducing exposure to aircraft noise.
Nonlinear transient response of electrostatic loudspeakers
Authors: Schroeter Juergen , Sondhi Man Mohan
Published: January 2006

When an electrostatic loudspeaker (ESL) is used as an acoustic pulse generator [see M. M. Sondhi and J. R. Resnick, J. Acoust. Soc. Am. 73, 985–1002 (1983) for one such recent application], it may well be desirable to operate it at levels at which its motion is highly nonlinear. Available analyses of ESLs are inadequate for such applications because they are concerned with computing sine-wave response and estimating low-level harmonic distortion. To fill this gap we have developed an algorithm to compute the general, nonlinear transient response of a circular ESL to a circularly symmetric excitation. Our formulation is general enough to allow multiple support of the membrane at an arbitrary number of concentric circular ridges on the backplate. Our analysis includes the effects of a back cavity, a perforated backplate, and the coupling of the membrane motion to the eigenmodes of a cylindrical impendance tube in front of the membrane. (The program can be easily modified for other configurations, e.g., when the membrane is in an infinite baffle.) We will present some computed results and their comparisons to measured data.
Shock resistant PVDF hydrophones
Authors: Henriquez T. A. , Miller Mary Lou
Published: January 2006

Previous research successfully demonstrated the use of tubular PVDF in a shock resistant hydrophone [T. A. Henriquez, “Application of Tubular PVDF to Shock Resistant Hydrophones,” in Proceedings of the International Symposium of Applied Ferroelectrics (1983)]. The present work continued the development and refinement of a shock hardened underwater transducer. Nonvoided as well as voided tubular material with an outer diameter of 14 or 12 in. was used as the active element in each design. To evaluate the effects of stress rod material and circumferential prestress, the free-field voltage sensitivity (FFVS) was recorded for each transducer. The response of the element was dependent upon the acoustic coupling between the compression bolt and rubber mandril within the tube and between the rubber mandril and the tube itself. The hydrophones were also exposed to scaled down explosive tests to determine their ruggedness. The results support previous findings that tubular PVDF can be utilized as a broadband standard hydrophone. [Work supported by NAVSEA.]
Underwater transducer wetting agents
Authors: Ivey L. E. , Thompson C. M.
Published: January 2006

The problem of air bubbles forming and clinging to the surface of oily or dirty underwater acoustic transducers has been around since the first calibration facilities were established in 1941. An air bubble driven at its resonance frequency has a measurable interference effect on the transducer's calibration response. The bubble problem can be eliminated by thoroughly treating the transducer with a wetting agent after the grease and oil have been cleaned off with a strong detergent. The wetting agent in some cases can also act as the degreaser. There is a controversy among present-day calibration facilities as to which wetting agent is best. A survey was made of the Navy-sponsored underwater transducer calibration facilities to determine which wetting agent each was using. Each was using a different type. Samples of each type were obtained and evaluated at NRL-USRD. The tests performed were surface tension measurements, compatability measurements with several popular types of rubber materials, and acoustic tests of each type on standard transducers in the USRD's Lake Calibration Facility. The results of these tests shown in the form of curves, equations, and tabulated data validate which wetting agents are best to use as surface acting agents.
Analysis of segmented cylinder ceramic transducers using multiport interconnection techniques
Authors: Johnson Michael P. , Thompson Stephen C.
Published: January 2006

For many years transducer analysis using multiport interconnection techniques has been successfully applied to tonpilz designs and other transducer types in which the vibratory motion is entirely one-dimensional. The analysis of radially vibrating cylindrical ceramic elements has generally used either empirically derived equivalent circuit methods or has started directly from the differential equations of motion. For segmented cylinders it is possible to perform an approximate analysis using linear network analysis methods. This allows the use of standard multiport analysis computer codes such as SEADUCER to perform much of the calculation. Furthermore these methods eliminate the need for empiricism. Calculated radiation impedance and finite mounting surface impedance effects can be directly included. A description of the method and comparison of the results with previous methods will be provided.
Amplitude calibration of an FFT analyzer
Authors: Pope Joseph
Published: January 2006

When used for frequency analysis, an FFT analyzer produces a line spectrum on the amplitude-frequency plane. Each spectrum line represents the output of a filter centered at the frequency associated with the line, the shape of the filter being determined by the effective time window applied to the data being analyzed. While it is generally recognized that the use of an explicit window function is often desirable—e.g., to control spectral “leakage”—many FFT users seem unaware that use of a window implies a compensatory calibration adjustment that is dependent upon the character of the data and the objective of the analysis. Effective filter shapes for rectangular, Hanning, flat top, and Kaiser-Bessel windows are discussed along with the concept of filter amplitude response and effective noise bandwidth. Appropriate calibration for rms, power, power spectral density, and energy spectral density spectra is explained.
Use and misuse of an FFT analyzer for octave-band and one-third octave-band analysis
Authors: Sohaney Richard C.
Published: January 2006

An FFT analyzer is often used as a “real time analyzer” to perform octave-band and one-third octave-band analyses. The limitations and assumptions which the filter synthesizing technique imposes on the data signal are discussed. A method for synthesizing ANSI Class II octave-band and Class III one-third octave-band filters using an FFT analyzer is explained with emphasis on the shortcomings of synthesized filters. Example analyses of various signals are performed with conventional ANSI Class III one-third octave-band filters and an FFT analyzer to demonstrate correct and incorrect use of the synthesized filter method.
Recognition of famous voices forwards and backwards
Authors: Van Lancker Diana , Kreiman Jody , Emmorey Karen
Published: January 2006

To investigate familiar voice recognition, samples of famous male voices were tape-recorded and edited on a PDP-11/34 computer. Three listening tasks were prepared. Subjects indicated whether they recognized voices from (1) 2-s samples; (2) different and reordered 2-s samples presented along with six choices; and (3) 4-s samples presented backwards (rerandomized and refoiled). Ninety-six subjects were divided into four groups, three by age and one given the backwards presentation first. For task 1, the mean recognition rate was 17%. In task 2, subjects correctly identified 69.5% of the voices they knew (ascertained by questionnaires). Recognition for known voices presented backwards was 12.8% less than for voices presented forwards. In the group given the voices backwards first, a similar difference (12.5%) was observed. A two-way ANOVA comparing the four groups on tasks 2 and 3 revealed main effects of group and task, but no group-by-task interaction; thus differences in performance between forwards and backwards presentations were unchanged across the four subject groups. These results indicate that voice recognition can be achieved given only limited information which includes rate, pitch, and pitch range. [Work supported by NIH Grant No. 4-443944-31117.]
Intonation, pitch height, and voice quality as indicators of speaker attitude
Authors: Scherer Klaus R. , Silverman Kim , Ladd D. Robert
Published: January 2006

Utterances from recorded interviews were rated for attitude conveyed. Stimuli were presented in both the original audio version and various degraded versions. Judgments of backwards and random-spliced versions correlated highly with originals, while judgments of low-pass filtered pitch contours did not. This suggests attitude is conveyed directly by voice quality largely independent of pitch contour. However, judgments of originals show (1) correlations between absolute pitch height and ratings of speaker arousal, and (2) interactions between linguistic categories of intonation and syntax in judgments of more “cognitive” attitudes like “friendly” and “critical” (e.g., final rises were rated friendly on yes/no questions but critical in WH questions). This shows that it is important to consider the linguistic structure of intonation and the accompanying text in studying intonational cues to attitude. Masking or merely averaging acoustic characteristics of F0 will obscure these interactions. [Work supported by German Science Foundation.]
Suppression of noise in speech using PARCOR analysis
Authors: Mano Atsushi , Ozawa Shinji
Published: January 2006

In this paper, a new noise suppression algorithm is presented for reducing any kind of background noise in speech. This noise suppressor uses three kinds of filters. The first filter, which is the inverse lattice filter implemented by PARCOR analysis of noise part, makes any colored background noises into white noises. The second filter, which is the white noise suppressor using spectral subtraction, reduces the white noise components in speech. The third filter, which is the noninverse type of the first one, restores the speech distorted by the first filter. The algorithm which detects nonspeech periods is combined with this noise suppressor, and the characteristics of these filters adaptively follow the nonstationary noise environment. The traffic noise and jet engine noise, etc., were used to estimate this system. As a result of the tests, we could improve the speech to noise ratio more than 20 dB. And at the same time, we could improve the articulation of the speech.
Acoustic scattering correction in a superleak partially filled with superfluid helium
Authors: Baker S. , Marcus J. , Williams G. , Rudnick I
Published: January 2006

The index of refraction n has been measured for the sound mode which propagates in a superleak partially filled with HeII so that the liquid forms a film on the powder grains. The mode is a thickness wave in this film. Motivated by a model which treats the scattering from the powder grains independently from the scattering from the vapor spaces, we determine the values of beta in the equation, n2 = f−β2 p−β1, where f is the fraction of the pore volume filled with liquid HeII and P is the fraction of the volume occupied by both liquid and vapor He4. The values of beta are interpreted in terms of the tortuosity of the microscopic geometry. The scattering correction has been measured for 0.3 < f < 1.0. In this range the agreement between the data and the model is excellent with values of β1≃0.71 and β2≃1.3. When f = 1, the pores are filled with liquid and the system is a fourth sound resonator. The value of n2 is in excellent agreement with directly measured fourth sound values. [Work supported by ONR.]
A comparison between Biot model predictions and shear wave attenuation measurements in unconsolidated laboratory sediments
Authors: Brunson Burlie A.
Published: January 2006

Shear wave attenuation measurements were made using three different water-saturated sediments. The sediment assemblages used in the attenuation measurements included well-sorted spherical glass beads, a well-sorted angular quartz sand, and a moderately sorted angular quartz sand. The median grain size of the sediments was held constant while the grain shape and size distributions were varied. Sediment physical properties were measured under the same conditions prevailing during the acoustic measurements. The properties were used as inputs to a theoretical attenuation model based on the Biot theory of propagation of waves in porous media. The model allowed attenuation versus frequency predictions to be made for each of the three sediment assemblages over the measurement frequency range of 1–20 kHz. The resultant comparisons between the measured and predicted attenuations demonstrated the importance of using measured model inputs obtained under controlled laboratory conditions when theoretical model capabilities are being evaluated. The model comparisons shed significant light on the ability of this particular model to predict shear wave attenuation in nonideal sediments. [Work supported by NORDA.]
The human engineering of a new teleconferencing service
Authors: Eigen Daryl J.
Published: January 2006

The Bell System is developing a new network teleconferencing service. This service enables the customer to set up a conference with or without an operator, and it incorporates state-of-the-art signal processing technology to enhance audio quality. The human engineering of this new service concentrated on characterizing the customer, optimizing the perceived audio quality, and making the conference setup procedures easy to use and error free. A coordinated series of interviews, surveys, laboratory studies, and field studies was implemented to increase customer satisfaction, performance, and usage. This sequence of testing is culminated in a Controlled Product Test, which is a dress rehearsal of a new service, using Bell System employees as the first customers. The CPT started on 7 February 1983 in selected sites in San Diego, Los Angeles, and San Francisco with two bridges located in Los Angeles. The CPT has since been expanded to additional sites in 12 states with four more bridges, two in White Plains, New York and two in Chicago, Illinois. This geographic diversity provides the opportunity for large calling volumes and a wide range of conference configurations. Over 2000 calls have been placed thus far, allowing the conference setup procedures to be simplified and perceived audio quality to be improved.
A portable, low-cost, digital system for room acoustics measurements
Authors: Essert Robert D.
Published: January 2006

A portable digital acquisition and analysis system has been developed around a low-cost microcomputer. The system consists of a programmable transient signal generator and an acquisition and analysis section. Designed principally for impulse response measurements in both full scale and model rooms, the acquisition system has been configured for portability and ease of use in the field. Data is “captured” directly into the computer main memory and is stored in digital form on floppy disks. A resident graphics module can be used to display the impulse response in various forms and to display the sound energy decay curve computed from the impulse response by backward integration. General purpose signal processing software has been written for computation of Fourier transforms, power spectra, cepstra, and digital filtering. The use of Wiener filtering for the deconvolution of room impulse response and source signal is being implemented, and is the subject of a companion paper. Some results of measurements made in existing performance spaces will be presented.
Human auditory steady state potentials analyzed with Fourier analysis: The zoom technique
Authors: Linden R. Dean , Hamel Gilles , Stapells David R. , Picton Terence W.
Published: January 2006

The Fourier analysis of evoked potentials has several advantages over averaging. Regan introduced a “zoom” technique that manipulates stimulus parameters during the analysis [Invest. Opthal. 12 669–679 (1973)]. Six subjects were tested using this technique to assess the effect of stimulus rate on the auditory evoked potentials. A 95-dB peSPL tone with 4-ms rise and fall times and a 2-ms plateau was presented to the right ear at a rate that increased from 10 to 60 tones/s over 50 s. This analysis was carried out for four different frequencies of the tone: 500, 1000, 2000, and 4000 Hz. The amplitude and phase outputs from the Fourier analyzer were repetitively recorded during the sweep in stimulus rate and averaged over 32 sweeps. The maximum amplitudes of the steady-state responses for the 500, 1000, 2000, and 4000 Hz tones occurred at rates of 40/s, 43/s, 44/s, and 46/s with amplitudes, of 1.42, 1.34, 1.30, and 1.11 μV, respectively. This technique may be used in “zooming” other parameters of interest such as intensity.
Phase consistency of evoked responses to auditory stimuli delivered at high rates
Authors: Makeig Scott , Galambos Robert
Published: January 2006

We will review our recent research in evoked response detection using auditory stimuli at rates around 40 Hz with emphasis on response phase consistency. Response phase often aggregates at stimulus intensities producing little or no rise in response amplitude beyond the level of the background EEG, and thus promises to be useful in electric response audio-metry. We note that others have previously proposed heavily weighting phase statistics in detecting responses to weak stimuli delivered at low rates (around 1 Hz). This sensitivity of evoked response phase to external stimulation suggests that phase of ongoing brain activity may be reorganized by weak stimuli before there is a detectable increase in net response amplitude.
Exploring the “McGurk effect”
Authors: Manuel Sharon Y. , Repp Bruno H. , Studdert-Kennedy Michael , Liberman Alvin M.
Published: January 2006

McGurk discovered that when a videotape of a talker is dubbed with similar but mismatched utterances, the optic information exerts a strong influence on the speech that is perceived, often without the observer's awareness. In several studies, we explored further the effects of conflicting acoustic and optic information on consonant perception. The following questions were asked: (1) Do subjects report a consonant even when no consonantal cues are contained in the acoustic signal (an isolated vowel)? (2) How is acoustic-optic place of articulation conflict resolved when acoustically specified manner (e.g., nasal, stop) restricts possible places of articulation? (3) How far does observers' awareness of acoustic-optic discrepancy reduce cross-modal integration? [Work supported by NICHD.]
Peak sound pressure level and spectral frequency distributions in echolocation pulses of Atlantic bottlenose dolphins, Tursiops truncatus
Authors: Ceruti Marion G. , Moore Patrick W. B. , Patterson Sue A.
Published: January 2006

We investigated the relationship between peak sound pressure level (PPL) and the shape of the spectral frequency distribution of echolocation pulses emitted by Atlantic bottlenose dolphins (Tursiops truncatus) in a target detection task. Two dolphins were trained to station on a bite plate and tail rest, echolocate, and report target condition (present or absent). A microprocessor data acquisition system [M. G. Ceruti and W. W. L. Au, J. Acoust. Soc. Am. 73, 1390–1392 (1983)], was used to collect amplitude and frequency data for each pulse and monitor subjects' responses. Ten sessions of 50 trials each (25 target present and 25 target absent, presented randomly) were chosen for each subject from sessions conducted during Nov. 1982. Pulses were classified as “bimodal” or “unimodal,” depending on the number of maxima in the frequency spectrum between 30 and 135 kHz. For bimodal pulses, the degree of bimodality BI, also was measured by summing the maximum intensities using the intervening minimum as a baseline. The results collapsed across subjects yielded 15 670 bimodal and 12 383 unimodel pulses. The mean PPL for bimodal pulses was 207 dB re:1 μPa and for unimodal pulses, 197 dB. The Pearson product moment correlation coefficient between BI and PPL was 0.69. Peaks in the averaged bimodal spectrum occurred at 60 and 135 kHz or beyond, while the averaged unimodal spectrum was peaked at 120 kHz. These findings suggest that bimodal frequency distributions during high-pressure echolocation represents a physiological process in the pulse emission system.
Spread signal processing
Authors: Callahan Andrew C.
Published: January 2006

The signal processing of broadband data often results in signals which are still spread over a region of parameters, such as, frequency, Doppler, delay, etc. It is of great interest to determine the possible gains achievable by the inclusion of all signal contributions into the formation of a region statistic. This is formulated into the analysis of distribution mixtures which is referred to as spread signal processing. To determine the potential gains in this approach, a particular example of signal plus noise and noise distributions were analyzed for detection performance as a function of signal level and the fraction of signal samples. Simulation results indicate an improvement over both the greatest sample and constant false alarm rate statistics for intermediate values of the signal sample fraction.
Determination of the broadband spectra for saturated parametric sonars
Authors: Wyber R. J.
Published: January 2006

The determination of the propagation of a finite amplitude wave by Burger's equation is well known. Application of this solution to parametric sonars generally involves modeling a plane wave in the nearfield and a spherical wave in the farfield. The transition between these regions is not well defined and generally involves an arbitrary choice of a transition region. To avoid this requirement and to allow the solution for a broadband parametric sonar to be extended to the finite amplitude case, a model has been developed based on the approximation that the nonlinear losses in the fundamental are due entirely to the work done on the nonlinear source function. The model developed is similar in principle to that of H. Merklinger [J. Acoust. Soc. Am. 54, 1760–1761 (1973)], except that the ad hoc estimate of the nonlinear source function is replaced by an integral of the primary intensity. This solution is extended to allow for broadband source signals up to 1-oct bandwidth. A solution is obtained for the spectrum of the total power in the primary beam, which is independent of the cross-sectional area of the beam and is thus valid in both the nearfield and farfield.
Performance of a broadband cross correlation detector
Authors: Tong Peter S.
Published: January 2006

Cross correlating the outputs of two spatially separated sensors is one way to implement a broadband passive detector and bearing estimator. In the absence of targets, the correlation function will be nearly zero for all delays, and in the presence of a target, this function will peak at the delay corresponding to the target bearing provided the target is motionless. If the target is moving, the signal received at the two sensors will experience different amounts of time compression, and as a result the correlation function will not reach its peak value. Exact expressions for this degradation are obtained in terms of target geometry, target velocity, signal frequency, bandwidth, and coherent integration time. These expressions, although adequate for performance computation in a given situation, are too complex for system trade-off studies. Greatly simplified expressions are derived in this paper which give valuable insight into the relationships of the many variables and are more suitable for system design and evaluation.
A comparison of broadband sonar discrimination between human listeners and a filter bank model
Authors: Martin Douglas W. , Au Whirlow W. L.
Published: January 2006

Broadband sonar discrimination capabilities of human listeners as a function of signal-to-noise ratio were compared with the constant-Q filter bank model of Chestnut, Landsman, and Floyd [J. Acoust. Soc. Am. 66, 140–147 (1979)]. A 48-s pulse (122-kHz peak frequency, 39-kHz 3-dB bandwidth) was used to collect echoes from aluminum, glass, and bronze cylinders. Discrimination experiments were performed in pairs, aluminum-glass, hollow-solid aluminum, and aluminum-bronze. A modified method of constants with the noise levels randomized in ten-trial blocks and the signals time-expanded into audio frequencies were used in the human experiments. Thirty contiguous filters covering a frequency band of 50 to 200 kHz were used in the constant-Q model. The human listeners performed significantly better than the computer model, requiring at least 5 dB less signal-to-noise ratio at the 75% correct threshold. Listeners reported that dominant cues occurred in the time domain.
Transient sound radiation from a clamped circular plate—Impulse response method
Authors: Akay Adnan , Latcha Michael
Published: January 2006

A theoretical analysis of transient sound radiation from a clamped circular plate is obtained using a pressure impulse response method. The vibration response of the plate to a transient point force is obtained. The modal pressure impulse response functions for the plate are derived from the Rayleigh surface integral and numerically convoluted with the modal acceleration response of the plate. The impulse response functions are closely related to the mode shapes and the geometry of the problem. They relate the spatial domain to the temporal domain of the pressure waves. The pressure impulse response waveforms are given for a number of plate modes and the changes in the waveforms with distance from the plate are shown. Sound radiation due to forced and free vibrations of the plate are discussed. The results are compared with those obtained by direct numerical integration of the Rayleigh surface integral and the experiments. [Work supported by WSU and NSF.]
Interpolation techniques for sound radiation prediction from a flat vibrating surface
Authors: Saha Pranab , Pierce Allan D.
Published: January 2006

In an attempt to develop a model of flat surface vibrations to make accurate farfield sound radiation predictions, two interpolation techniques (12-point formula and 24-point formula) have been developed. The purpose of these techniques is to generate additional surface vibration data at regions of sparse observations and in between gaps of existing data to obtain smooth surface vibration data. The development of the interpolation techniques is based on the equally spaced data points and the method of undetermined coefficients, given by the equation, an(0,0) = an(x,y) − (k1x + k2y + k3x2 + k4 xy + k5y2), where k1, k2,k3,k4, and k5 are the coefficients needed for the best fit of the curve in order to relate the unknown surface vibration an(0,0) with the known surface vibration an(x,y). An expression for the relative error has also been derived. This expression provides information about the vibration distribution over the flat surface and hence about the applicability of the interpolation techniques. The results of the application of these two interpolation techniques based on an available set of test data are reported here, and are compared with the measured sound pressure levels in the farfield.
Diffraction at a fixed point on a fluid-loaded plate
Authors: Waterhouse R. V. , Archibald F. S.
Published: January 2006

At the last meeting, some theoretical results were presented for the nearfield diffraction pattern formed by plane bending waves passing a fixed point on a thin elastic plate in vacuo. This case is a good approximation to a metal plate in air, but not to a metal plate with water on one or both sides. The latter case is of interest, for example, when the plate represents the hull of a ship, and in the present work the previous analysis is extended to this case. The fluid loading adds a third dimension to the problem, as now some of the acoustic energy is transferred from the plate to the water. The expression for the sound pressure contains a complicated integral which must be evaluated numerically. The integral contains a fifth degree polynomial in the denominator, the five roots of which contribute five poles to the integrand.
Eigenrays and eigenmodes for source-excited propagation in layered multiwave media
Authors: Lu I. T. , Felsen L. B. , Kamel A.
Published: January 2006

Coupling of wave species at interfaces and boundaries in a medium composed of plane multiwave layers creates a proliferation of ray fields even after relatively few multiple reflections. This inhibits a ray treatment of propagation from source to observer. The difficulty may be overcome by diagonalizing, in a plane wave spectral representation of the Green's function, the reverberation matrix F descriptive of the boundary coupling. The resulting eigenvectors of F represent combinations of the original Q wave species, to be referred to as eigenrays, which, except for multiplication by the eigenvalue λq, q = 1 ⋯ Q, remain unaltered after one complete reverberation. Thus eigenrays may be traced through successive reverberations like ordinary rays in a single-wave medium. This feature also permits the original multiwave Q × Q matrix problem to be decoupled into a sequence of scalar problems. Conventional eigenmodes are generated from eigenrays by imposing self-consistency (λq = 1) after one reverberation. Alternative representations are given for the multiwave Green's function by use of these new concepts. [Work supported by National Science Foundation and ONR Acoustics Branch.]