• Volume/Page
  • Keyword
  • DOI
  • Citation
  • Advanced
   
 
 
 

Journal of the Acoustical Society of America

BROWSE VOLUMES

Year Range: 
Search Issue | RSS Feeds RSS
Previous Issue Next Issue

Nov 1990

Volume 88, Issue 5, pp. 2059-2521

Page 1 of 6 Pages Next Page | Jump to Page

Thresholds for transient cavitation produced by pulsed ultrasound in a controlled nuclei environment

Christy K. Holland and Robert E. Apfel

J. Acoust. Soc. Am. Volume 88, Issue 5, pp. 2059-2069 (1990); (11 pages) | Cited 9 times

Full Text: | Download PDF

Show Abstract
Transient cavitation is a discrete phenomenon that relies on the existence of stabilized nuclei, or pockets of gas within a host fluid, for its genesis. A convenient descriptor for assessing the likelihood of transient cavitation is the threshold pressure, or the minimum acoustic pressure necessary to initiate bubble growth and subsequent collapse. An automated experimental apparatus has been developed to determine thresholds for cavitation produced in a fluid by short tone bursts of ultrasound at 0.76, 0.99, and 2.30 MHz. A fluid jet was used to convect potential cavitation nuclei through the focal region of the insonifying transducer. Potential nuclei tested include 1‐μm polystyrene spheres, microbubbles in the 1‐ to 10‐μm range that are stabilized with human serum albumin, and whole blood constituents. Cavitation was detected by a passive acoustical technique that is sensitive to sound scattered from cavitation bubbles. Measurements of the transient cavitation threshold in water, in a fluid of higher viscosity, and in diluted whole blood are presented. These experimental measurements of cavitation thresholds elucidate the importance of ultrasound, host fluid, and nuclei parameters in determining these thresholds. These results are interpreted in the context of an approximate analytical theory for the prediction of the onset of cavitation.
Show PACS
43.80.Gx Mechanisms of action of acoustic energy on biological systems: physical processes, sites of action
43.30.Nb Noise in water; generation mechanisms and characteristics of the field
43.80.Ev Acoustical measurement methods in biological systems and media
43.35.Ei Acoustic cavitation in liquids

Mode stretching and harmonic generation in the flute

John W. Coltman

J. Acoust. Soc. Am. Volume 88, Issue 5, pp. 2070-2073 (1990); (4 pages) | Cited 1 time

Full Text: | Download PDF

Show Abstract
Comparison of the resonance frequencies of modes in conical and cylindrical piccolos showed pronounced differences in the degree to which the octaves are stretched. Measurements for a note on a flute with only two modes showed that second harmonic generation and total radiated power were greatest when the modes were stretched 25 cents from an exact octave, supporting the idea that mode stretching may be beneficial in flutelike instruments.
Show PACS
43.75.Ef Woodwinds

Acoustic cues for consonant identification by patients who use the Ineraid cochlear implant

Michael F. Dorman, Sigfrid Soli, Korine Dankowski, Luke M. Smith, Geary McCandless, and James Parkin

J. Acoust. Soc. Am. Volume 88, Issue 5, pp. 2074-2079 (1990); (6 pages) | Cited 6 times

Full Text: | Download PDF


See Also: Erratum

Abstract Unavailable
Show PACS
43.71.Ky Speech perception by the hearing impaired
43.66.Ts Auditory prostheses, hearing aids
43.71.Es Vowel and consonant perception; perception of words, sentences, and fluent speech
43.64.Me Effects of electrical stimulation, cochlear implant

Stimulus order effects in vowel discrimination

Bruno H. Repp and Robert G. Crowder

J. Acoust. Soc. Am. Volume 88, Issue 5, pp. 2080-2090 (1990); (11 pages) | Cited 2 times

Full Text: | Download PDF

Show Abstract
In same–different discrimination tasks employing isolated vowel sounds, subjects often give significantly more ‘‘different’’ responses to one order of two stimuli than to the other order. Cowan and Morse [J. Acoust. Soc. Am. 79, 500–507 (1986)] proposed a neutralization hypothesis to account for such effects: The first vowel in a pair is assumed to change its quality in memory in the direction of the neutral vowel, schwa. Three experiments were conducted using a variety of vowels and some initial support for the hypothesis was obtained, using a large stimulus set, but conflicting evidence with smaller stimulus sets. Rather than becoming more similar to schwa, the first vowel in a pair seems to drift toward the interior of the stimulus range employed in a given test. Several possible explanations are discussed for this tendency and its relation to presentation order effects obtained in other psychophysical paradigms is noted.
Show PACS
43.71.Qr Neurophysiology of speech perception
43.71.Es Vowel and consonant perception; perception of words, sentences, and fluent speech
43.71.An Models and theories of speech perception
43.66.Jh Timbre, timbre in musical acoustics

Vowel amplitude variation associated with the heart cycle

Robert F. Orlikoff

J. Acoust. Soc. Am. Volume 88, Issue 5, pp. 2091-2098 (1990); (8 pages)

Full Text: | Download PDF

Show Abstract
Modulation of the acoustic amplitude of a sustained vowel across the cardiac (ECG) cycle was examined by signal‐averaging techniques. Ten normal men prolonged [a] at a comfortable F0 maintained within three SPL ranges: 60–68, 70–78, and 80–88 dB. Peak‐to‐peak amplitude variation associated with the heart cycle averaged 8.5% (s.d.=5.4) re: mean, varying from about 14% at low SPLs to approximately 3% at high SPLs. The amplitude modulation was estimated to account for 11.8% of the measured short‐term amplitude perturbation (shimmer), ranging from about 5% to almost 22% for individual samples. The mean deterministic shimmer (Sd ) was 0.036 dB (s.d.=0.019), with a trend toward decreasing Sd with increasing SPL. Additionally, fundamental frequency variation across the heart cycle within these phonations was comparable to that observed by Orlikoff and Baken [J. Acoust. Soc. Am. 85, 888–893 (1989)], and was shown to be uninfluenced by vocal SPL, although deterministic jitter (Jd) did decrease with vocal intensity. The results are discussed in terms of how the phonovascular relationship may affect the reliability and interpretation of acoustic shimmer measures.
Show PACS
43.70.Aj Anatomy and physiology of the vocal tract, speech aerodynamics, auditory kinetics
43.70.-h Speech production

Effects of postlingual deafness on speech production: Implications for the role of auditory feedback

Robin S. Waldstein

J. Acoust. Soc. Am. Volume 88, Issue 5, pp. 2099-2114 (1990); (16 pages) | Cited 16 times

Full Text: | Download PDF

Show Abstract
This study investigated some effects of postlingual deafness on speech by exploring selected properties of consonants, vowels, and suprasegmentals in the speech of seven totally, postlingually deafened individuals. The observed speech properties included parameters that function as phonological contrasts in English, as well as parameters that constitute primarily phonetic distinctions. The results demonstrated that postlingual deafness affects the production of all classes of speech sounds, suggesting that auditory feedback is implicated in regulating the phonetic precision of consonants, vowels, and suprasegmentals over the long term. In addition, the results are discussed in relation to factors that may influence the degree of impairment, such as age at onset of deafness.
Show PACS
43.70.Dn Disordered speech
43.66.Sr Deafness, audiometry, aging effects

The effects of bandwidth on the detectability of narrow‐ and wideband signals

Richard S. Bernstein and David H. Raab

J. Acoust. Soc. Am. Volume 88, Issue 5, pp. 2115-2125 (1990); (11 pages) | Cited 3 times

Full Text: | Download PDF

Show Abstract
The effects of masker bandwidth on the detection of narrow‐ and wideband signals have been investigated. For both kinds of signals, plots of threshold as a function of masker bandwidth yielded by both narrow‐ and wideband signals are reasonably described with two intersecting lines. Thresholds initially increase with masker bandwidth and then become independent of further increases. The rate of increase depends on the signal spectrum. The bandwidth at which the lines intersect varies with signal bandwidth and also mode of masker presentation (i.e., whether the masker is gated with the signal or is present continuously). Internal filtering is most accurate when the masker is present continuously. A model is proposed in which a listener’s decisions about the presence of narrow‐band signals are based upon estimates of stimulus energy within a critical band. These estimates are degraded by bandwidth‐dependent processing errors. When the signal to be detected spans several critical bands (i.e., is wideband), the model forms a test statistic by summing the outputs of the relevant critical bands. The model permits the contribution of each band to the sum to vary with masker bandwidth because it incorporates a form of lateral suppression. Thresholds of narrow‐band signals in gated maskers and wideband signals in gated and continuous maskers are predicted by the model. However, the model fails to account for the detectability of narrow‐band signals in continuous maskers.
Show PACS
43.66.Ba Models and theories of auditory processes
43.66.Dc Masking
43.66.Fe Discrimination: intensity and frequency

Harmonic and melodic octave templates

Laurent Demany and Catherine Semal

J. Acoust. Soc. Am. Volume 88, Issue 5, pp. 2126-2135 (1990); (10 pages) | Cited 3 times

Full Text: | Download PDF

Show Abstract
For normal‐hearing adult listeners, two simultaneous pure tones with a frequency ratio close to 2/1 may perceptually fuse into a single sound, which shows that such listeners are sensitive to ‘‘octave harmony.’’ Many adult listeners are also able to consistently adjust two successive pure tones ‘‘one octave apart,’’ which shows that they possess melodic octave templates. According to Terhardt [J. Acoust. Soc. Am. 55, 1061–1069 (1974)], melodic octave templates and the perception of octave harmony originate from a common learning process taking place in early life. In the two experiments reported here, subjects performed repeated octave adjustments for pairs of simultaneous and successive tone bursts. Both tones were presented monaurally, at 45 or 65 dB SPL. The frequency of the lower tone (fref) was an independent variable, while the frequency of the higher tone was adjustable within a 500‐cent range. In some conditions, when the two tones were presented simultaneously, they were sinusoidally frequency modulated in a coherent manner, at a rate of 2 or 4 Hz; the aim of this frequency modulation was to force the subjects to adopt a synthetic listening strategy, i.e., to base their adjustments on perceived harmony. For fref values ranging from 270–2000 Hz, subjects performed consistent adjustments when the tones were presented successively: fref had little effect on the adjustments’ variability. However, in the same frequency range, the variability of the harmonic adjustments markedly increased with fref ; for the highest fref values, it was much greater than the variability of the melodic adjustments. The results suggest that, in adult listeners, the perception of octave harmony disappears at frequencies for which melodic octaves are still accurately perceived.
Show PACS
43.66.Ba Models and theories of auditory processes
43.66.Hg Pitch
43.66.Lj Perceptual effects of sound
43.75.Cd Music perception and cognition

The loudness of sounds that increase and decrease continuously in level

Georges Canévet and Bertram Scharf

J. Acoust. Soc. Am. Volume 88, Issue 5, pp. 2136-2142 (1990); (7 pages) | Cited 2 times

Full Text: | Download PDF

Show Abstract
A sound at a low level is heard as much softer after having decreased continuously from higher levels than if presented after a period of silence at that same low level. Canévet [Acustica 61, 256–264 (1986)] demonstrated this phenomenon for a tone that (1) decreased from 65 to 20 dB in 180 s; he also presented a tone that (2) increased from 20 dB, or (3) was presented as pairs of bursts at various levels in random order. Below about 40 dB, loudness changed most rapidly in the decreasing condition so that, at 20 dB, the tone was judged ten times softer than in conditions (2) and (3). In the present experiments, magnitude estimation was used to examine the possible role of judgmental biases and adaptation in this rapid loudness decline, which we call decruitment. Results show that decruitment did not come about because subjects made many successive loudness judgments; loudness declined as much when a tone was judged only twice, at the beginning and end of its 180‐s decrease. In contrast, interrupting the decreasing tone so that it was heard only at 70 dB and 160 s later at 30 dB greatly diminshed the decruitment. Similarly, pairs of 500‐ms tone bursts presented at successively lower levels instead of continously decreasing did not show decruitment, suggesting that sequential biases are irrelevant. The likely cause of decruitment is sensory adaptation.
Show PACS
43.66.Cb Loudness, absolute threshold
43.66.Mk Temporal and sequential aspects of hearing; auditory grouping in relation to music

Effects of stimulus level on forward‐masked psychophysical tuning curves in quiet and in noise

David A. Nelson, Steven J. Chargo, Judy G. Kopun, and Richard L. Freyman

J. Acoust. Soc. Am. Volume 88, Issue 5, pp. 2143-2151 (1990); (9 pages) | Cited 6 times

Full Text: | Download PDF

Show Abstract
Forward‐masked psychophysical tuning curves were obtained from normal‐hearing listeners at different probe levels in quiet and in a broadband background noise. In quiet, tuning‐curve shape changed with probe level. For six listeners, tuning curves became broader with increasing probe level, primarily due to a decrease in the low‐frequency slopes. For one listener, tuning curves became narrower with increasing probe level. The addition of a background noise, which was presented continuously at a level 10 dB below the noise level required to mask the probe tone, reduced the masker levels required to mask the probe tone. The reduction was greater near the tip of the tuning curve than on the tail, so that tuning curves in background noise were narrower than those obtained in quiet. Tuning curves with comparable masker levels near the tip of the tuning curve (Lmtip) were similar in shape, regardless of probe level or whether tuning curves were obtained in quiet or noise. Comparisons of tuning‐curve characteristics derived by fitting tuning curves with least‐squares procedures, indicated that low‐frequency slopes decreased with Lmtip. As a consequence, Q10 dB values decreased with Lmtip. These results are consistent with the interpretation that tuning‐curve shapes are determined by the intensities of the maskers required to mask the probe tone. The addition of a background noise restricted (partially masked) the excitation pattern of the probe so that lower masker intensities were required to ‘‘forward mask’’ the probe tone, and narrower tuning curves resulted from less intense maskers. The results are well described by a two‐process model of auditory excitation patterns.
Show PACS
43.66.Dc Masking
43.66.Mk Temporal and sequential aspects of hearing; auditory grouping in relation to music

Frequency discrimination in the monkey

C. A. Prosen, D. B. Moody, M. S. Sommers, and W. C. Stebbins

J. Acoust. Soc. Am. Volume 88, Issue 5, pp. 2152-2158 (1990); (7 pages) | Cited 3 times

Full Text: | Download PDF

Show Abstract
This study evaluated frequency discrimination ability in 11 monkeys over an extended period of time using a repeating‐standard procedure and the method of constant stimuli. The intersubject variability of the difference limens for frequency (ΔF) was large, as reported by other investigators, but similar in magnitude to the variability of the difference limens for intensity (ΔI) from three of the same subjects in an intensity discrimination experiment. Continued training generally resulted in a rapid decrease in ΔF’s, followed by a longer‐term, slower decrease. For one subject ΔF’s slowly decreased throughout a 190‐week time period. This long‐term training effect was specific to frequency discrimination; a similar effect was not observed for the same subject tested in an intensity discrimination experiment. Finally, ΔF’s from the well‐trained monkeys of this study were larger than monkey ΔF’s from this laboratory reported in an earlier study, and than human ΔF’s. An anatomical explanation for the human/monkey ΔF magnitude difference is explored.
Show PACS
43.66.Gf Detection and discrimination of sound by animals
43.80.Lb Sound reception by animals: anatomy, physiology, auditory capacities, processing

The combination of interaural information across frequencies: Lateralization on the basis of interaural delay

Raymond H. Dye, Jr.

J. Acoust. Soc. Am. Volume 88, Issue 5, pp. 2159-2170 (1990); (12 pages) | Cited 16 times

Full Text: | Download PDF

Show Abstract
Three experiments were carried out that employed low‐frequency tone complexes with interaural delays that varied across the frequency domain. In the first experiment, threshold interaural delays were measured for three‐tone complexes for which one, two, or all three components were delayed. The center frequency was 750 Hz and the frequency spacing (Δf ) between components was 20, 50, 100, 250, or 450 Hz. For all Δf’s, the presence of two diotic components elevated the threshold interaural delays obtained for the third component relative to that obtained for a pure tone of the same frequency. In the second experiment, observers made left–right judgments regarding the direction of movement of signals for which two components were delayed by 25 μs to the left ear during one interval and to the right ear during the other interval, while a third component of a variable time difference was delayed to the opposite side as the tone pair. Subjects reported single intracranial images during each interval, and the data showed that interaural delays of one component to one ear could be offset by interaural delays of the other two components to the other ear. In the final experiment, threshold interaural delays were measured for five‐tone complexes in which one, two, three, four, or five components were delayed. The center frequency was 750 Hz and Δf was fixed at 100 Hz. Thresholds decreased in a linear fashion as the number of delayed components increased, falling by about a factor of 5 as the number of delayed components went from one to five. These results are consistent with spectrally synthetic binaural processing, with the lateral position of intracranial images determined by a combination of interaural information across the spectrum. These effects could be brought about by a linear combination of the outputs of frequency‐specific cross‐correlation networks or by a wideband cross correlation of the signals at the two ears.
Show PACS
43.66.Pn Binaural hearing
43.66.Nm Phase effects

Classification of audiograms by sequential testing using a dynamic Bayesian procedure

Özcan Özdamar, Rebecca E. Eilers, Edward Miskiel, and Judith Widen

J. Acoust. Soc. Am. Volume 88, Issue 5, pp. 2171-2179 (1990); (9 pages)

Full Text: | Download PDF

Show Abstract
A new method for estimating audiograms using behavioral responses is presented. The method is based upon a modification of the Bayesian probability formula in which an outcome is predicted from a static set of events. In the new method, classification of audiograms by sequential testing (CAST), the probabilities of occurrence of audiogram patterns are dynamically updated according to the outcome of each test trial. Computer simulation using an infant response model suggests that the procedure is efficient, sensitive, and specific.
Show PACS
43.66.Sr Deafness, audiometry, aging effects
43.66.Yw Instruments and methods related to hearing and its measurement

The auditory periphery of the ferret. I: Directional response properties and the pattern of interaural level differences

Simon Carlile

J. Acoust. Soc. Am. Volume 88, Issue 5, pp. 2180-2195 (1990); (16 pages) | Cited 11 times

Full Text: | Download PDF

Show Abstract
The transformations of sound by the auditory periphery of the ferret have been investigated using an impulse response technique for a large number of sound locations surrounding the animal. Individual frequencies were extracted from the detailed spectral transformation functions (STFs) obtained for each stimulus location and, using sophisticated spatial interpolation routines, were used to calculate the directional response of the periphery at that frequency. The strength of the directional response was directly related to the analysis frequency. Furthermore, as the analysis frequency was increased to 20 kHz, the orientation of the directional response increased in elevation from the horizon (E0°) to about E30°, while the azimuthal location remained fairly constant at 30° to 40° from the midline. For analysis frequencies above 20 kHz, the response became increasingly directional toward the ipsilateral interaural axis. The interaural level differences (ILDs) were also calculated for all animals studied. ILDs increased from around 5 to 25 dB over the range of frequencies from 3–24 kHz. The two‐dimensional patterns of iso‐ILD contours were roughly concentric and centered on the interaural axis for frequencies below 16 kHz. For higher frequencies, there was a tendency for the ILD contours to be centered on more anterior and inferior locations. The increased directionality of the auditory periphery with increasing analysis frequency, together with the presence of sharp nulls in the response at high analysis frequencies, is consistent with a diffractive effect produced by the aperture of the pinna. However, this simple model does not predict the directional responses over the low to middle frequency range.
Show PACS
43.64.Ha Acoustical properties of the outer ear; middle-ear mechanics and reflex
43.66.Qp Localization of sound sources

The auditory periphery of the ferret. II: The spectral transformations of the external ear and their implications for sound localization

Simon Carlile

J. Acoust. Soc. Am. Volume 88, Issue 5, pp. 2196-2204 (1990); (9 pages) | Cited 2 times

Full Text: | Download PDF

Show Abstract
In the previous paper the directional response characteristics of the ferret auditory periphery were examined. In this study further measurements of the spectral transfer functions (STFs) of the auditory periphery were obtained at locations close to the tympanic membrane. There was considerable variation in the STFs recorded from different animals and between recordings made at each end of the auditory canal in the same animal. However, calculation of the so called ‘‘location dependency function’’ demonstrated that changes in the location of the stimulus produced the same pattern of changes in the STFs in all recordings. Changes in the spectral transformation for azimuth locations in the ipsilateral auditory field were examined by calculating the horizon STF. The gain transformations of frequencies below 20 kHz were found to be asymmetrical about the interaural axis so that maximum gain was obtained for anterior stimulus locations. In contrast, the maximum gain for frequencies above 20 kHz was obtained for stimulus locations about the interaural axis, and movement of the stimulus location into either the anterior or posterior fields produced symmetrical reductions in gain. These changes were related to the directional properties of the periphery examined in the previous paper (Carlile, 1990).The spatial resolution of the monaural information provided by the peripheral STFs is dependent on the rate of change of the transformations as a function of azimuthal displacement of the stimulus location. This was examined by calculating the unsigned first spatial derivative for each frequency in the horizon STF. The spatial derivative of frequency was found to be high for locations about the posterior and anterior median planes. This is discussed in terms of the results of behavioral experiments examining the resolution of sound localization in the ferret and other mammals.
Show PACS
43.64.Ha Acoustical properties of the outer ear; middle-ear mechanics and reflex
43.66.Qp Localization of sound sources
43.80.Lb Sound reception by animals: anatomy, physiology, auditory capacities, processing

Electrically evoked whole‐nerve action potentials: Parametric data from the cat

Carolyn J. Brown and Paul J. Abbas

J. Acoust. Soc. Am. Volume 88, Issue 5, pp. 2205-2210 (1990); (6 pages) | Cited 1 time

Full Text: | Download PDF

Show Abstract
In a companion paper [Brown et al., J. Acoust. Soc. Am. 88, 1385–1391 (1990)], a method for recording the electrically evoked whole‐nerve action potential in human cochlear implant users was reported. The procedure for recording the response requires that two biphasic current pulses, a ‘‘masker’’ and a ‘‘probe,’’ be presented at a rate and level sufficient to drive the auditory nerve into a refractory state. The present study was designed to assess the sensitivity of that recording technique to variations in stimulation parameters. The experiments described in this paper demonstrate that: (1) the EAP as recorded in the cat is triphasic and is defined by two negative peaks occurring at latencies of approximately 0.26 and 0.82 ms; (2) EAP amplitude is independent of the level of the masker stimulus for current levels equal to or greater than the current level of the probe stimulus; and (3) the time course of recovery of the EAP from the refractory state is stable over a range of both probe and masker current levels.
Show PACS
43.64.Me Effects of electrical stimulation, cochlear implant
43.64.Pg Electrophysiology of the auditory nerve

Response of auditory‐nerve fibers to intensity increments in a multitone complex: Neural correlates of profile analysis

Bhagyalakshmi G. Shivapuja, Richard J. Salvi, and Samuel S. Saunders

J. Acoust. Soc. Am. Volume 88, Issue 5, pp. 2211-2221 (1990); (11 pages) | Cited 1 time

Full Text: | Download PDF

Show Abstract
Recent psychophysical studies have shown that the detection of an intensity increment superimposed on the center component (1 kHz) of a multitone complex (1,3,7, or 11 components) improves as more components are added outside of the critical band. It has been suggested that this form of intensity discrimination is based on a change in the neural profile. To test this hypothesis, neural profiles were constructed by plotting the degree of phase locking to the 1‐kHz tone as a function of each unit’s characteristic frequency (CF). Neural phase‐locking profile to the 1‐component signal at 1 kHz had a broad peak; however, the neural profile became narrower as the number of components in the signal increased. The just detectable increment for the 1‐component condition was −5 dB re: 1000‐Hz component level (3.86‐dB increment plus component level re: component level), whereas, for the 3‐, 7‐, and 11‐component conditions, it was −15 dB re: component level (1.42 dB). The neural and psychophysical IDL for the chinchilla were similar for the 1‐component condition. However, the overall trends in the psychophysical and neural data are different. In the psychophysical studies IDL is typically poorest in the 3‐component condition and improves when more components are added. By contrast, the neural IDL was poorest in the 1‐component condition and improved when more components were added. In the multicomponent conditions, units with CFs in 492–1380 Hz were found to be most sensitive in detecting the intensity increment to the 1000‐Hz component.
Show PACS
43.64.Pg Electrophysiology of the auditory nerve
43.64.Bt Models and theories of the auditory system
43.64.Tk Physiology of sound generation and detection by animals
43.80.Lb Sound reception by animals: anatomy, physiology, auditory capacities, processing

Stimulus dependencies of the gerbil brain‐stem auditory‐evoked response (BAER). III: Additivity of click level and rate with noise level

Robert Burkard and Herbert F. Voigt

J. Acoust. Soc. Am. Volume 88, Issue 5, pp. 2222-2234 (1990); (13 pages) | Cited 2 times

Full Text: | Download PDF

Show Abstract
Two experiments were performed that evaluated the effects of ipsilateral‐direct broadband noise maskers on the gerbil brain‐stem auditory‐evoked response (BAER) to click stimuli. In experiment 1, clicks were presented at 27 Hz at levels including 70, 80, 90, and 100 dB pSPL. Noise conditions included a no‐noise control, and included noise levels varying in 10‐dB increments from 20 dB SPL to a maximum noise level of 50, 60, 70, and 80 dB SPL for click levels of 70, 80, 90, and 100 dB pSPL, respectively. Gerbil BAER peaks were labeled with small roman numerals to distinguish them from human BAER peaks. The dependent variables included waves i and v latencies and amplitudes. Peak latencies increased and peak amplitudes decreased with decreasing click level and increasing noise level. To a first approximation, peak latencies and amplitudes showed changes with increasing noise level that were similar across click level. With increasing click level, there was little or no effect on the i–v interval. There was an increase in the i–v interval with increasing noise level. In experiment 2, click level was held constant at 90 dB pSPL, and click rates included 15, 40, 65, and 90 Hz. For each click rate, noise conditions included a no‐noise control, and noise levels included 20, 30, 40, 50, 60, and 70 dB SPL. With increasing click rate and noise level, there was an increase in peak latencies, an increase in the i–v interval, and a decrease in peak amplitudes. The magnitude of peak latency and amplitude shifts with increasing click rate was dependent on noise level. Specifically, the magnitude of rate‐dependent changes decreased with increasing level of broadband noise. These data are compared to human BAER experiments, and are found to be in fundamental agreement.
Show PACS
43.64.Ri Evoked responses to sounds
43.64.Qh Electrophysiology of the auditory central nervous system
43.66.Dc Masking

Transient wave estimation: A multichannel deconvolution application

J. V. Candy, R. W. Ziolkowski, and D. Kent Lewis

J. Acoust. Soc. Am. Volume 88, Issue 5, pp. 2235-2247 (1990); (13 pages) | Cited 3 times

Full Text: | Download PDF

Show Abstract
The evolution of new concepts in wave theory have led to proof‐in‐principle experiments aimed at validating the generation of a specified wave front. Not only have these concepts initiated research in transient wave theory, but they have also caused renewed effort in multichannel signal processing. In this paper, the development of a processor to deconvolve a transient acoustic wave from sensor array measurements is discussed. The design of the multichannel deconvolver coupled with model‐based signal processing techniques using acoustic pressure field measurements is discussed. Here, it is shown that an efficient solution to this problem can be obtained using a vector form of the Levinson–Wiggins–Robinson (LWR) algorithm.
Show PACS
43.60.Gk Space-time signal processing, other than matched field processing

Transient waves: Reconstruction and processing

J. V. Candy, R. W. Ziolkowski, and D. Kent Lewis

J. Acoust. Soc. Am. Volume 88, Issue 5, pp. 2248-2258 (1990); (11 pages) | Cited 4 times

Full Text: | Download PDF

Show Abstract
New solutions to the wave equation have been shown to exhibit enhanced localization and energy fluence characteristics. The transmission and reception of these localized waves create unique problems, since they are essentially transient wave fronts in both time and space. Nonetheless, the ability to transmit wave energy through space with these interesting properties has many potential applications in a variety of applications areas. To realize their potential, new methods must be developed to analyze and process these waves. In this paper, approaches to design receiving arrays to reconstruct these special transient waves from noisy measurement data are discussed.
Show PACS
43.60.Gk Space-time signal processing, other than matched field processing
43.30.Wi Passive sonar systems and algorithms, matched field processing in underwater acoustics

Spatial Fourier transform method of measuring reflection coefficients at oblique incidence. I: Theory and numerical examples

Masayuki Tamura

J. Acoust. Soc. Am. Volume 88, Issue 5, pp. 2259-2264 (1990); (6 pages) | Cited 22 times

Full Text: | Download PDF

Show Abstract
A new method using spatial Fourier transform has been developed to measure reflection coefficients at oblique incidence. The method involves the measurement of complex pressure distributions on two parallel planes lying close to the surface of a test material and decomposing each of the complex pressure distributions into plane‐wave components by using two‐dimensional spatial Fourier transform. The incident and reflected plane‐wave components on the surface of the test material can be mathematically separated by the use of plane‐wave propagation theory. This separation leads to the determination of reflection coefficients at arbitrary angles of incidence. Investigation has been made into the error due to the finite size of the measurement area to show that the magnitude of the error can be reduced by using a dipole source instead of a monopole source. Numerical examples are given to illustrate the validity of the method.
Show PACS
43.55.Ev Sound absorption properties of materials: theory and measurement of sound absorption coefficients; acoustic impedance and admittance
43.58.Bh Acoustic impedance measurement

Active reduction of a one‐dimensional enclosed sound field: An experimental investigation of three control strategies

A. R. D. Curtis, P. A. Nelson, and S. J. Elliott

J. Acoust. Soc. Am. Volume 88, Issue 5, pp. 2265-2268 (1990); (4 pages) | Cited 2 times

Full Text: | Download PDF

Show Abstract
Three control strategies for the active reduction of an enclosed sound field are experimentally implemented and compared. A strategy of energy minimization successfully reduces the sound field at both resonant and antiresonant frequencies, whereas strategies developed for the control of propagating sound in ducts do not perform well.
Show PACS
43.50.Ki Active noise control
43.50.Gf Noise control at source: redesign, application of absorptive materials and reactive elements, mufflers, noise silencers, noise barriers, and attenuators, etc.

Modal sampling method for the vibration study of systems of high modal density

J. L. Guyader

J. Acoust. Soc. Am. Volume 88, Issue 5, pp. 2269-2276 (1990); (8 pages) | Cited 2 times

Full Text: | Download PDF

Show Abstract
This paper presents a method to predict the vibration response of a group of modes, using a sample of modes of the group. The accuracy of the prediction can be improved when increasing the size of the sample. Numerical calculations made on a beam, driven in longitudinal motion, show the ability of the method to reconstitute the energy response of a group of modes including resonant modes, with a sample of a number of modes close to that in the effective damping bandwidth.
Show PACS
43.40.Cw Vibrations of strings, rods, and beams
43.40.At Experimental and theoretical studies of vibrating systems

Application of integral equation technique to nonlinear stochastic response of rectangular plates

R. S. Srinivasan and P. A. Krishnan

J. Acoust. Soc. Am. Volume 88, Issue 5, pp. 2277-2283 (1990); (7 pages)

Full Text: | Download PDF

Show Abstract
The present paper is an illustration of the application of the integral equation technique to a nonlinear random response problem. This method relies heavily on the effective use of matrices ab initio and appropriate matrix manipulation to get the equivalent linear stiffness matrix which is the crux of the equivalent linearization technique used here. Rectangular isotropic plates clamped laterally and free of in‐plane stresses at the edges are subjected to random excitation. The analysis is done using Von Karman equations and numerical results have been presented.
Show PACS
43.40.Dx Vibrations of membranes and plates
43.25.Dc Nonlinear acoustics of solids
43.40.Hb Random vibration

Absorption of ultrasonic waves in aqueous solutions of N‐methylacetamide and zinc chloride

Piotr Miecznik

J. Acoust. Soc. Am. Volume 88, Issue 5, pp. 2284-2290 (1990); (7 pages)

Full Text: | Download PDF

Show Abstract
Measurements of the absorption coefficient of ultrasonic waves have been made within the frequency range of 10–100 MHz in aqueous solutions of N‐methylacetamide (NMAA) and zinc chloride (ZnCl2). The method used to prepare quasi‐two‐component solutions permitted a change in the ratio of the number of amide molecules to electrolyte molecules, given a constant number of water molecules. The results indicate the occurrence of a single relaxational process in the solutions under investigation and the frequency range adopted. On the basis of the theory of relaxational absorption of sounds, relaxation parameters as well as the enthalpy of activation of the solution in question, which was 16.4 kJ/mol, have been calculated. The character of the relaxational process observed was determined on the basis of the analysis of relaxational curves. The discovered relaxational process in H2O–NMAA–ZnCl2 was ascribed to the formation and disintegration of ‘‘solvatomers’’ composed of [NMAA Zn(H2O)3]2+.
Show PACS
43.35.Bf Ultrasonic velocity, dispersion, scattering, diffraction, and attenuation in liquids, liquid crystals, suspensions, and emulsions
Page 1 of 6 Pages Next Page | Jump to Page
Close

Home Publications Meetings Contact ASA Site Map Back to Top

Copyright © Acoustical Society of America

close