bluuebunny/crossref_metadata_2025 huggingface dataset:
The analysis of an optical fiber microphone design
Authors: Hu Andong , Cuomo Frank W. , Zuckerwar Allan J.
Published: October 2005

The theoretical response of a condenser microphone [A. J. Zuckerwar, J. Acoust. Soc. Am. 64, 1278–1285 (1978)] and an electret microphone [R. Zahn, Acoustica 57y, 200–204 (1985)] has been developed based on the mean displacement of a stretched membrane. Recent developments have produced an optical fiber pressure sensor [F. W. Cuomo, R. S. Kidwell, and A. Hu, SPIE 661, 234–239 (1986)], which utilizes a stretched metallized Mylar film with an optical fiber probe acting at its center. This paper derives the theoretical expressions representing the peak displacement at that point. In addition, the effects of damping due to the backchamber are investigated to show the variations in amplitude and frequency response due to the different design approaches. It is shown that equivalent expressions for the peak displacement can be obtained exactly as well as by the lumped elements approach. Comparisons are made between the results previously obtained for a 1-in.-diam B&K microphone and the present analysis. [Work supported by NASA.]
Fiber optic sensor for measurement of pressure fluctuations in a turbulent boundary layer. Part I: Construction and calibration
Authors: Cuomo Frank W. , Kidwell Robert S. , Hu Andong
Published: October 2005

The measurement of mean velocities, forces, pressures, and shear stresses can provide useful information about transition, separation, and turbulent friction in the study of fluid flow. In most cases, previous sensors have lacked some of the features necessary to satisfactorily establish the flow properties. This paper discusses the development of an optical fiber pressure sensor whose active dimensions, sensitivity, and frequency response have shown to improve the measuring capabilities presently available. The device is designed for flush mounting in a suitable aluminum plate installed in the test section of a wind tunnel. Details of the construction will be discussed and data, related to the dc and ac calibration methods used, will be presented. [Work supported by NASA and NUSC.]
Reciprocal relation between the growth of an emitted cubic distortion product and the suppression of a spontaneous otoacoustic emission
Authors: Pasanen E. G. , Wier C. C. , McFadden D.
Published: October 2005

Emitted cubic distortion products were generated in ears having a spontaneous otoacoustic emission (SOAE). The frequencies of the primary tones were set to yield a distortion product 20 Hz below the frequency of the SOAE; the levels of the tones were equal and were varied over a wide range. With primaries below about 35 dB SPL, both emissions were present and could be measured separately. As the levels of the primaries were increased, and the amplitude of the distortion product rose, the SOAE was progressively suppressed from its level in the quiet until, at some level of the primaries, it was no longer measurable. Over the range where both emissions were detectable, the rate of decline of the SOAE as a function of the level of the primaries was approximately slope −1.0, while the rate of growth of the emitted distortion product was approximately slope +1.0—suggesting perfect energy conservation. For higher primary levels, where the SOAE was no longer measurable, the growth function showed saturation, being of lower slope, or even nonmonotonic in some cases. Moderate doses of aspirin abolished the SOAEs of all subjects, but only slightly diminished the distortion products. [Work supported by NINCDS Grant NS 15895.]
Cochlear model for acoustic emissions
Authors: Furst M. , Lapid M.
Published: October 2005

Acoustic emissions, both spontaneous and evoked, were measured in humans and in several animals. There were significant differences in the experimental results. In all animals a strong acoustic distortion product (ADP) was obtained, whereas in 60%–70% of the human subjects ADP was unmeasurable. Those human ears that emitted ADP usually possessed spontaneous emission (SE) and also Kemp's echo. The amplitude of the ADP was stronger when its frequency or the primary-tones frequency was in the vicinity of SE or near a peak of the Kemp's echo spectrum. In animals, the SE and Kemp's echo were rarely measurable. In order to understand the differences of the emission properties observed in various people and animals, a modification of the cochlear nonlinear transmission-line model is suggested. On each branch of the transmission line, a noise source was added to simulate cochlear internal noises. To simulate a local cochlear damage, the vertical resistance was modulated. The emitted spectrum predicted by the model contained a frequency whose CF is at the location of the damage. When introducing a two-tone stimulus (f1 and f2), the emitted spectrum predicted by the model contained components of the DP frequencies (i.e., 2f1 − f2 and 2f2 − f1). The amplitude of the DPs increased when there was a local damage at the CF of either the primary tones or the DP frequencies. Following this research, it is concluded that (a) normally ADP in humans is much smaller than in animals because of the different loads at the oval window, and (b) a local damage in the basilar membrane causes the rise of spontaneous emission and an increase of ADP amplitude.
Effect of high stimulation rates and paired stimuli on the electrically evoked brain-stem response
Authors: Brown D. D. , Miyamoto R. T.
Published: October 2005

Temporal cues are a primary source of information for the single-channel cochlear implant subject. Two temporal features of the electrical auditory brain-stem response (EABR) in patients implanted with single-channel devices are being investigated. In the first of these studies the stimulus rate is varied from 10 to 150/s and changes in wave V amplitude and latency are recorded. Increasing the stimulus repetition rate causes a prolongation of wave V latency and a decrease in amplitude. These changes are particularly prominent for stimulation rates exceeding 70/s. In the second experiment the EABR response to the second stimulus of a stimulus pair is computed as the time interval between the two stimuli is varied from 0.5 to 10 ms. Initial data indicate a significant interaction only for interstimulus intervals less than 4 ms. In these two studies, variations across patients have been found and an attempt is being made to correlate these variations with other measures of the subjects' implant performance. Findings and clinical implications will be reported.
Timing in auditory perception
Authors: Hirsh Ira J.
Published: October 2005

Recognition of sequential patterns of sound is basic to speech recognition, music perception, and identification of environmental sounds from birdsong to thunderclaps. While we define these sound patterns as sequences of brief sounds, the usual acoustical description has principally been the spectrum. For some 30 years, work has been reported on fusion and successivity, duration discrimination, perceived order, and the influence of temporal variables on more traditional psychoacoustic dimensions. At Central Institute, with colleagues Divenyi, Lauter, Grant, Singh, and Monahan, our emphasis has been on listeners' sensitivity to changes in order and timing, and influence of nontemporal features on such perception. Here, some experiments are described that address interval perception in a series. In an isochronous series of three, six, or ten tones with intertone intervals of 200 ms or more, listeners can detect a temporal offset or “jitter” of between 5% and 10%, irrespective of the position of offset. For shorter intervals, performance is worse and there is a strong effect of position in the series, with offsets made late in the pattern being more easily discriminated than those made earlier. When a single pitch change is made in one tone in a series of six, detection of an offset is impaired when it coincides with the different pitch. Phrased patterns (do-re-mi-do-re-mi) on the other hand, have not shown such a strong position effect. It is recognized that temporal patterns influence, and are influenced by stimulus dimensions as well as by dimensions learned from formal systems like the rules of language and music. Our search continues to focus on an in-between perceptual territory, which lies beyond sensory processes, but yet is less dependent on rule learning, [Work supported by NINCDS and AFOSR.]
Determinants of experienced tension in timbre sequences
Authors: Povel Dirk-Jan
Published: October 2005

In listening to rhythms, subjects may experience a sensation of tension. It is assumed that this important psychological attribute is directly related to the process of perceiving a rhythm. In this study, tension judgments ascribed to repetitively presented binary isochronic sequences (onset-interval 130 ms) are examined. One “period” of a sample sequence can be notated as X X O X X O X O in which X and O represent two timbres produced by different drums. Subjects were asked to rate the experienced tension of 16 sequences on a five-point scale. Responses show consistent differences in judged tension. Tension judgments are explained by supposing that the sequences are perceived as composed of two separate sequences formed by the X and O events, respectively [Garner, The Processing of Information and Structure (Erlbaum, Potomac, MD, 1974)], each contributing its own tension to the compound tension sensation. The tension evoked by the separate sequences is explained by the displaced beat hypothesis proposed by Povel [“Time, Rhythms and Tension: in Search of the Determinants of Rhythmicity,” in Time, Mind and Behavior, edited by J. Michon and J. Jackson (Springer, Berlin, 1985)].
The perception of metrical structure: Experimental evidence and a new model
Authors: Lee Christopher
Published: October 2005

This paper presents experimental results which are in conflict with existing theories of metrical interpretation [H. C. Longuet-Higgins and C. S. Lee, Perception 11, 115–128 (1989); H. C. Longuet-Higgins and C. S. Lee, Music Perception 1, 424–441 (1984); F. Lerdahl and R. Jackendoff, A Generative Theory of Tonal Music (MIT, Cambridge, MA, 1983); D. J. Povel and P. Essens, Music Perception 2, 411–440 (1985)] and a new model, based on the algorithm in Longuet-Higgins and Lee (1982), which accounts for the main features of these results. Musically sophisticated subjects listened to sequences of identical tones in various rhythmic arrangements and transcribed them using standard musical notation. The results disconfirmed key predictions of all four theories, in that subjects displayed a general reluctance to choose interpretations containing an upbeat. However, the results did show that subjects were more prepared to avoid syncopated interpretations than those in which long notes occurred on weak beats. These tendencies are accounted for in the model by the assumption that the listener initially takes the first note of a sequence as the downbeat and is reluctant to abandon this (or later) hypotheses. But the occurrence of syncopations constitutes strong counterevidence and the occurrence of long notes on weak beats weak counterevidence, which can overcome the listener's conservatism. [Work supported by Nuffield Foundation, Royal Society and SERC.]
Rhythm perception and the perception of periodicity
Authors: Balzano Gerald J.
Published: October 2005

To hear something repeating is to hear something repeating, and this something must have a beginning and an end. Just where beginnings and ends are perceived in periodic duration patterns is the problem in rhythm perception with which the present studies are concerned. More specifically, how does the metrical framework generated by a periodic duration pattern map into the elements of the pattern? Where is the “downbeat” perceived to occur, and how is the time period initiated by that downbeat organized into subperiods? These questions were addressed using two distinct methodologies. In one, listeners from an introductory musical literacy course transcribed simple patterns into musical staff notation. In the other, listeners from a wider population responded to similar patterns by tapping along in time with them. Both methods revealed pronounced tendencies to hear patterns in particular ways. Some candidates for rhythmical “laws of organization” that account for these tendencies are offered, along with a commentary on the relationship between the two methods of measuring rhythmic interpretations.
The influence of temporal contour and interval information on the discrimination of rhythmic patterns
Authors: Monahan Caroline B. , Carterette Edward C.
Published: October 2005

Montone rhythmic patterns were constructed, each totaling 16 beats apportioned among 6 notes; we describe the rhythms as ordered sequences of beat interval values of 6 notes, e.g., 313144. Musicians judged similarity of standard-comparison pattern pairs. On target trials, the comparison had the same rhythm as the standard; on related trials, the comparison had the standard's temporal contour, that is, the same series of longs, shorts, and sames without regard to interval size (as in the pair 313144–315133); on lure trials, the comparison had a different contour and intervals. Twenty-four musicians were randomly assigned to one of three listening conditions that differed in trial types presented: (A) targets versus lures; (B) targest versus relateds, or (C) relateds versus lures. Two other factors were tempo and metrical simplicity (i.e., the extent to which groups of intervals formed simple submultiples of the 16 beats). Discrimination was better (1) in condition A than in B or C, although performance in both B and C was still above chance, demonstrating the importance of interval and contour information, respectively; (2) when the comparison tempo was the same as the standard's; (3) with metrically simpler standards. Tempo interacted with condition: discrimination of comparison patterns that were faster or slower was poorer in B and C than in A.
Effects of interpretation on timing in piano performance
Authors: Palmer Caroline
Published: October 2005

Three timing methods in piano performance were described in a previous report [C. Palmer, J. Acoust. Soc. Am. Suppl. I 79, S75 (1986)]: onset asynchronies, rubato patterns, and legato and staccato patterns. The present study examined the relation of these timing methods to performers' intended interpretations of an excerpt. Norated interpretations included indication of primary melody, phrasing, dynamics, and tempo changes. Performances were recorded on a computer-monitored (MIDI interface), velocity-sensitive, weighted keyboard. The performances showed consistent temporal patterns, directly related to the specified interpretations. Onset asynchronies occurred within chords, such that the primary melody (as notated by each performer) preceded the other voices. Asynchronies were generally largest on the first beat of each measure, marking the excerpt's metrical structure. Rubato patterns (deviations in tempo from mechanical regularity) showed larger changes between, than within performers' notated phrases. Legato and staccato patterns within phrases were accurately predicted by a combination of the durations of successive notes in the musical score, but only when performers' intended phrasings were taken into account. Each of these timing patterns decreased in degree when the pianists were asked to play unmusically. [Work supported by NSF and NIMH.]
Evaluation of ASR front ends in speaker-dependent and speaker-independent recognition
Authors: Junqua Jean-Claude
Published: October 2005

This paper extends previous experiments of Tsuga and Hermansky [J. Acoust. Soc. Am. Suppl. 1 80, Sl8 (1986)]. Those experiments evaluated the effect of spectral model order, in automatic speech recognition (ASR), using a small alpha-numeric data base. PLP (perceptually based linear predictive) and LP (linear predictive) analyses were compared, using a cepstral and RPS (root power sums) metric. Those experiments dealing with a bigger data base were validated (104 words, ten speakers). PLP RPS front end is compared with about ten other ASR front ends (LP ceptrum, LP RPS, critical band,…). Experiments were run at various different analysis model orders. Results of speaker-dependent ASR show that the low-dimensional PLP analysis is a good alternative to high-dimensional LP or filter bank analysis. The index-weighted metric improves the recognition accuracy, making the recognition results more uniform across the speakers. The speaker-independent experiments (which use the templates of one male and one female speaker as references) confirm the superior performance of the PLP method for extracting speaker-independent information. Most of the errors are on the consonants; some improvements on the recognition process are being investigated.
Predicting stress and syllable boundaries from segmental timing
Authors: Anderson Sven , Port Robert , Maki Daniel
Published: October 2005

Recent studies [Reilly and Port, J. Acoust. Soc. Am. Suppl. 1 78, S21 (1985)] show that timing measurements can be used to discriminate among items in a small vocabulary. Current efforts question the extent to which timing measurements can be used to augment basic segmental knowledge in a continuous, real-time speech recognition situation. These experiments examine a single template of segments (vowel-strong fricative-stop-vowel) each of which is embedded in different words in 112 different sentences for four English speakers. Syllable and word boundaries occur between each of the four segments. Discriminant analysis is used to predict the stress patterns, possible syllabifications, and word boundaries from timing measurements in the templates of a fourth male speaker after training on three others. Results indicate that within the template a linear combination of simple timing measurements can be used as predictors of lexical stress (95% correct). Word and syllable boundaries are much less reliably predicted (60%-75%) and apparently bear no simple linear relationship to timing measurements. [Research supported by NSF.]
Automatic speech recognition in a noisy automotive environment
Authors: Wilpon J. O. , Rabiner L. R. , DeMarco D. , Shipley K. L.
Published: October 2005

Recently, a great deal of interest has been shown in performing reliable speech recognition in an automotive environment. In order to determine the level of recognition accuracy to be expected in such a hostile environment, a series of experiments was performed. The data base for these experiments was created by having four talkers (two male, two female) speak ten replications of an 11-digit vocabulary, zero-nine and oh (in an isolated format), while driving in an automobile. Five different environmental conditions were studied: (1) car off, fan off; (2) car idling, fan on high; (3) car traveling at 30 mph, fan on high; (4) car traveling at 60 mph, fan on high; and (5) car traveling at 60 mph, fan off. Two different recognition algorithms were tested, namely; (1) standard LPS-based template recognition, using dynamic time warping; and (2) continuous density hidden Markov model recognition, using Viterbi scoring and cepstral analysis. Several different distance measures were also evaluated for each of the recognition sytems. Results from the template-based algorithms showed speaker-dependent recognition accuracies from 96%–99% depending on which condition was used for training. A five-state HMM system yielded recognition accuracies from 93%–98% depending on the training condition.
Utilizing phase spectra for emulative speech recognition (ESR)
Authors: Cohen Nathan
Published: October 2005

A new approach is described to machine speech recognition that incorporates nonlinear spectral interferometry to model the binaural advantage in human speech recognition. This ESR scheme uses interharmonic and interear visibility observables from the spectra of phonemes to provide phoneme identification signatures. In particular, the phase spectra of phonemes are found to be coupled, but not redundant to, the amplitude spectra. Use of amplitude and phase spectra allows a critical resolution of phoneme ambiguities often present when derived from amplitude spectra alone. Interharmonic phase modulation appears to be a secondary means of encoding human speech and is primarily used to discern the speaker's identification and mood.
Dynamic frequency warping, the dual of dynamic time warping
Authors: Neuburg Edward P.
Published: October 2005

Comparison of two tokens of the same utterance is central to many automatic speech recognition systems. Matching is usually done in the frequency-time domain; token matching is effectively spectrogram matching. Dynamic time warping (DTW) overcomes, to some extent, the temporal variability of speech tokens; spectrograms are time-aligned by calculating similarity scores between segments of speech, now represented as “columns” of their spectrograms, and applying the mathematical technique called dynamic programming. DTW distorts the time scales of the spectrograms so that identical speech events in the two spectrograms now occur at identical times. Variability in frequency of these events is normally dealt with only by using robust distance measures. There is a better way. After time alignment, frequency variability can be dealt with specifically by doing a dynamic frequency warp (DFW), a process strictly analogous to the DTW. The “rows” of the speech spectrogram, which show the time behavior of the spectral components, play the same role in the DFW as the “columns” do in the DTW. Distances between the rows are calculated and passed to a dynamic program. The resulting DFW produces a distortion of the frequency scales such that identical speech events in the two tokens now occur both at the same time and at the same frequency. Experience with distance measures between rows is limited, but early results are promising: (1) Tokens that have been warped in both directions match better than tokens warped only in time. (2) Redoing the DTW after a DFW results in improved time alignment. (3) Tokens sound more like each other after DFW than before. (4) Pairs of speakers produce consistent frequency warps. Results will be demonstrated, and other applications suggested.
The effect of probe inertia on the accuracy of structural-intensity measurements in lightweight, built-up structures
Authors: Bernhard Robert J. , Mickol John D.
Published: October 2005

Two accelerometer methods have been proposed for measurement of flexural wave-energy propagation in beams and plates. In this investigation, the influence of the probe inertia was investigated both analytically and experimentally. It was found that for lightweight structures and narrow-band analysis the probe inertia changes the source impedance and thus significantly affects the intensity field throughout the structure. Power flow accountancy, done using contour integrals of intensity, was found to be highly suspect due to these inertial effects. When probe inertia is compensated by putting dummy probes at all measurement locations, the power flow appears to be consistent and stationary. The results indicate the need for either extremely lightweight or noncontacting structural-intensity transducers. In addition, the investigation points out the care which must be taken to make valid structural-intensity measurements. [Work supported by NASA Langley Research Center.]
Propagation of sound above a curved surface
Authors: Berry Alain , Daigle G. A.
Published: October 2005

There is a useful analogy between flat ground and curved ray paths in the atmosphere, and a curved ground surface above which there is no refraction. Many features of the sound field in the refractive shadow can be studied under controlled conditions indoors over a carefully constructed curved surface. Preliminary measurements, and comparison with simple theory restricted to the ideal cases of rigid and pressure-release boundaries, have been presented at an earlier meeting [J. Acoust. Soc. Am. Suppl. 1 79, S20 (1986)]. The measurements have been completed in the frequency range between 0.3–10 kHz, above a rigid boundary and a surface of finite impedance. Particular attention is given to the region across the shadow boundary. In the shadow, the theory has been extended by removing restrictive approximations and by calculating the creeping wave to higher-order terms. A numerical algorithm allows the extension to the case of a finite impedance. Above the shadow boundary the sound field is calculated using geometrical theory that accounts for reflection from a curved surface. At the shadow boundary both theories typically agree to within 0.5 dB. Good agreement between measured results and theory is obtained in general, except across the shadow boundary where discrepancies between 2 to 5 dB are observed. The theory may also predict a deeper shadow than measured at higher frequencies above the surface of finite impedance.
Acoustic insertion loss from compliant plate gratings
Authors: Ku C. George
Published: October 2005

The insertion loss from multiple gratings of compliant plate immersed in water is analyzed. The symmetry condition of the geometric configuration provides the convenience of using normal mode solutions of the equation of motion for plates. The structural motion of the compliant plates is coupled with the acoustic pressure of the surrounding water. Therefore, the magnitude of the insertion loss is controlled by the structural resonant modes of the plates. Numerical results are presented for compliant plate gratings with simply supported end conditions and clamped conditions. A brief discussion about the dynamic behavior of the grating is also presented.
Investigation of possibility of damage from acoustically coupled seismic waveform from blast and artillery
Authors: Sabatier James M. , Bass Henry E. , Rasper Richard
Published: October 2005

Noise from artillary fire causes complaints around military installations. Cracked house foundations are often included in these complaints. For the moderate size explosives involved and the minimum distances (> 2 km) to homes, it is improbable that directly induced seismic vibrations would have high enough amplitudes to cause this kind of damage. A possibility that has not been previously investigated is that the acoustically induced seismic wave might have a high enough amplitude to damage foundations. Measurements of the acoustic wave and the acoustically induced seismic wave were made for 8-in. artillery shells and 500-lb bombs with peak levels up to 145 dB, and it was found that the induced seismic wave was much smaller than the minimum criteria for damage. In addition, an acoustic-to-seismic steady-state plane-wave model was modified to predict the acoustic coupled seismic wave. Comparison between predicted and measured results will be presented. [Work supported by WES, CERL.]
Surface gravity wave effects on sound beams generated by a transient corrugated laser-deposited heat configuration moving at near sonic velocities
Authors: Pierce Allan D. , Hsieh Hsiao-an
Published: October 2005

Previous articles suggest the use of a multiply split laser beam moving over the water surface at sonic velocities to create an underwater sound pulse beamed at near horizontal directions. The moving corrugated wavetrain of laser-induced heating would systematically pump up the acoustic pressure and result in an improved optical-to-acoustic transduction efficiency, as well as yielding a controlled sound-radiation pattern. However, the extended nature of the effective source allows the possibility that gravity waves on the water surface may upset the phase coherence. Considered adverse mechanisms include the following: (1) the refraction of light at the tilted and curved rocking air-water interface causes the slabs of light penetrating to depths below the surface not to be perfectly parallel; (2) the rippled water surface causes the energy being added to a given moving (nearly planar) region of peak acoustic pressure to fluctuate; and (3) a nearly plane-acoustic wave propagating horizontally just below the surface is scattered by the surface ripples. Although the quantitative assessment of such mechanisms is still incomplete (and could be aided by experimental work), a rudimentary mathematical theory is given that suggests criteria for when gravity wave effects have minor influence. [Work supported by ONR.]
Parametric study of a laser-generated acoustic signal
Authors: Gonzalez Luis J. , Busch-Vishniac Ilene J.
Published: October 2005

The generation of underwater sound by a moving high-power laser source is numerically investigated. In particular, properties of the received acoustic signal, such as maximum and minimum periods and time-to-peak amplitude, are examined as a function of the source characteristics, such as velocity and source-receiver range and angle. Signal properties that are most and least stable to source geometry variations are quantitatively determined. [Research supported by ONR.]
Effect of acoustic tone perception on brain-generated electromagnetic waves: Preliminary evidence
Authors: Stocklin Philip L. , Cohen Glenn M. , Stocklin Norma L. , Blackburn Ron G.
Published: October 2005

Brain-generated microwaves were detected and measured for each of four subjects in the frequency range of the lowest electromagnetic standing wave (mode) of the adult human brain/skull cavity (200–400 MHz) in the presence and absence of a loud acoustic tone. In each case, the center frequency of the mode was found at, or within, a megahertz of that measured for each subject 1 yr earlier, and is different for each subject. In all cases, mode center frequency is within 10% of that theoretically predicted based on a subject's skull dimensions [P. L. Stocklin and B. F. Stocklin, T.I.T.J. Life Sci. 9, 29–51 (1979)]. In the absence of the acoustic tone, mode structure consists of a cluster of frequency spikes, each 2 to 3 kHz wide extending over 30 to 50 kHz. In the presence of a loud 1500-Hz continuous acoustic tone, the cluster of narrow spikes coalesces to one or two narrow spikes, 3 to 10 dB above the cluster average. [Work supported by Mentec Inc.]
Measurements of reflection and transmission coefficients of Rayleigh waves from cracks
Authors: Dong Ruiqi , Adler Laszlo
Published: October 2005

The angle and frequency dependence of reflection and transmission coefficients of Rayleigh waves from surface cracks in titanium has been measured. The measured results are compared to the recently developed theory of Angel and Achenbach [J. Acoust. Soc. Am. 75, 313–319 (1984)]. Very good agreement is obtained between experiment and theory.
Temporal gap resolution in masked normal ears as a function of masker level
Authors: Fitzgibbons Peter J.
Published: October 2005

Recent studies of temporal resolution in hearing-impaired listeners indicate that many ears with cochlear damage exhibit elevated temporal gap thresholds. The deviations from normal may be large, for equivalent-SPL comparisons, but are often small or absent for equivalent-SL comparisons. In the present experiment, we examine the premise that SL effects in gap resolution are independent of sound pressure level. Normal hearing subjects used a Békésy procedure to track the minimum level of an octave-band signal at 1 kHz needed to keep a periodic temporal gap at threshold. Performance functions were generated to show stimulus intensity as a function of gap duration, for gaps from 25 ms to the smallest detectable value. In separate conditions, the level of a white-noise background was varied to shift the subject’s audibility threshold. Results indicate that the limits of resolution are largely invariant with SPL, but criterion sensation levels for resolution are progressively reduced at higher SPL. It is concluded that gap thresholds from normal and impaired ears should be compared for equivalent SPL; comparisons made with SL equated are likely to underestimate the consequence of cochlear damage on resolving capacity.
Discrimination of dynamic interaural intensity differences
Authors: Grantham D. Wesley
Published: October 2005

An experiment was conducted to measure observers’ ability to detect time-varying interaural intensity differences (IIDs). In a two-interval forced-choice task, observers discriminated a binaural amplitude modulated (AM) noise in which the modulating sinusoid was interaurally in-phase from the same AM noise in which the modulator was interaurally phase-reversed. The latter stimulus produces a sinusoidally varying IID whose rate and peak IID depend on the frequency (fm) and depth (m) of modulation. The carrier was a narrow-band noise, interaurally uncorrelated, centered at 500, 1000, or 4000 Hz. Presentation level was 75 dB SPL; duration was 1.0 s. For a given fm, m was varied in an adaptive procedure to estimate the depth required for 71% discriminability (mthr). Three of the four observers displayed ‘‘low-pass’’ modulation functions: at 500 Hz, as fm increased from 0–50 Hz, mthr increased from 0.08 (IID=1.3 dB) to 0.50 (peak IID=9.5 dB). At 1000 and 4000 Hz observers were more sensitive to IID and the functions (mthr vs fm) were flatter than at 500 Hz. Comparison of these data to previously published data indicates that the binaural system can follow fluctuations in IID more efficiently than it can follow fluctuations in interaural time difference, although there are large individual differences in subjects’ capacity to process these two types of binaural cues.
Effects of age and mild hearing loss on speech recognition in noise
Authors: Dubno Judy R. , Dirks Donald D. , Morgan Donald E.
Published: October 2005

Using an adaptive strategy, the effects of mild sensorineural hearing loss and adult listeners’ chronological age on speech recognition in babble were evaluated. The signal-to-babble ratio required to achieve 50% recognition was measured for three speech materials presented at soft to loud conversational speech levels. Four groups of subjects were tested: (1) normal-hearing listeners <44 years of age, (2) subjects <44 years old with mild sensorineural hearing loss and excellent speech recognition in quiet, (3) normal-hearing listeners >65 with normal hearing, and (4) subjects >65 years old with mild hearing loss and excellent performance in quiet. Groups 1 and 3, and groups 2 and 4 were matched on the basis of pure-tone thresholds, and thresholds for each of the three speech materials presented in quiet. In addition, groups 1 and 2 were similar in terms of mean age and age range, as were groups 3 and 4. Differences in performance in noise as a function of age were observed for both normal-hearing and hearing-impaired listeners despite equivalent performance in quiet. Subjects with mild hearing loss performed significantly worse than their normal-hearing counterparts. These results and their implications are discussed.
An acoustic model of a multiple-channel cochlear implant
Authors: Blamey P. J. , Dowell R. C. , Tong Y. C. , Clark G. M.
Published: October 2005

A set of bandpass filtered, pulsed noise stimuli presented to three normally hearing subjects was found to have psychophysical properties similar to those of a set of pulsed electrical stimuli presented to two cochlear implant patients. Identical procedures were used to compare the performances of the two groups of subjects in the following tasks: (a) pulse rate difference limen measurements, (b) pitch scaling for stimuli differing in pulse rate, (c) pitch scaling and categorization of stimuli differing in filter frequency or electrode position, and (d) similarity judgments of stimuli differing in pulse rate and filter frequency or electrode position. By choosing the parameters of the acoustic stimuli appropriately, a high level of agreement between the two sets of results was achieved. Electrical stimuli on electrodes at different sites in the cochlea were matched with pulsed noise passed through bandpass filters with different center frequencies. Matching was achieved for equal electrical and acoustic pulse rates.
Half-order derivative of a sine-wave burst: Applications to two-dimensional radiation, photoacoustics, and focused scattering from spheres and a torus
Authors: Marston Philip L.
Published: October 2005

The half-order derivative (d/dt)1/2 s(t) is calculated for s(t) given by a burst of sine waves. The burst is N cycles in length, where N is an integer or a half-odd-integer. The result contains Fresnel integrals; it may be written in a compact form by using the auxiliary Fresnel-integral function usually denoted by g. Some novel properties of g were derived to facilitate a discussion of the result. The calculation is applicable to several dissipationless acoustical models for which the response to an exp(−iωt) input is proportional to ω1/2 exp(−iωt−iπ/4). Some examples include the pressure radiated by two-dimensional sources such as the optoacoustic radiation from a thin modulated laser beam. Another example is the pressure from virtual ringlike sources such as those present in focused backscattering from large spheres and a torus. The scattering from a large torus is modeled for the case of an incident plane wave which propagates parallel to the symmetry axis.
Application of the Schur algorithm to the inverse problem for a layered acoustic medium
Authors: Yagle Andrew E. , Levy Bernard C.
Published: October 2005

The Schur algorithm is a signal processing algorithm which works on a layer-stripping principle. Its time-domain version, the fast Cholesky recursion, is a fast and efficient algorithm well-suited for high-speed data processing. In this paper, these algorithms are applied to the inverse problem for a continuous layered acoustic medium. Three different excitations of the medium are considered: impulsive plane waves at normal incidence, impulsive plane waves at oblique incidence, and spherical waves emanating from an impulsive point source. The fast algorithms obtained for each of these problems seem to be computationally superior to past work done on these problems that employed Gelfand–Levitan theory to reconstruct the potential of a Schrödinger equation.
Discrimination of relative distance in the auditory modality: The success and failure of the loudness discrimination hypothesis
Authors: Strybel Thomas Z. , Perrott David R.
Published: October 2005

Difference limens for auditory distance discrimination were determined using the method of limits. Eight untrained subjects were tested across a wide range of distances (49 to 4876 cm) using broadband noise sources. For sources in excess of 304 cm, the Weber ratios observed approximated the function predicted by the inverse first power loss equation. At the longer distances, the subjects appear to be judging distance on the basis of the loudness difference cue. At the shorter distances (49 to 304 cm), the Weber ratios were considerably larger than would be predicted from the inverse first power loss equation. In this range, distance discrimination becomes increasingly degraded as the referent source approaches the subject.
A phase difference technique for acoustic imaging
Authors: Hayama Masaru , Toda Kohji
Published: October 2005

A method for obtaining C-scan images is reported by using a Lamb wave device with two arched interdigital transducers, in which the phase difference between an input signal as a reference and a delayed signal reflected at a sample surface or inside a test sample is detected as a function of the position. The transducers are used as a sound beam radiator or a detector at a liquid–solid interface. Acoustic imaging results are satisfactory in measuring the stress fields in ceramic samples.
A compaction model for compressional wave velocity in marine sediments
Authors: Bryan George M.
Published: October 2005

A compaction model is developed to describe the compressional wave velocity as a function of depth near the seafloor in marine sediments. The model assumes that the porosity decreases with increasing depth of overburden and approaches a limiting value in the first few hundred meters. The limiting value is estimated from sonobuoy data applicable to the deeper sediments where further compaction is negligible compared to age-dependent effects. Available consolidation data from the western North Atlantic are used in the model to predict the velocity profile, which is in good agreement with hydrophone-pinger data.
Beam-intensity calculations with uncertain sound speed profiles
Authors: Frankenthal Shimshon
Published: October 2005

A nonsecular procedure is derived for modifying the intensity distribution I(μ) computed in a channel with a sound speed profile μ, so as to account for small changes Δμ in the profile. This procedure is then employed to relate the statistics of a random ensemble of profiles to the statistics of the corresponding intensity distributions. The difference between I(〈μ〉), the intensity distribution for the mean (expected) profile, and 〈I(μ)〉, the expected intensity distribution for the ensemble, is studied: it is taken as a measure of the relevance of I(〈μ〉) as an approximation. At very short ranges, the two quantities are identical. At very long ranges, 〈I(μ)〉 obliterates entirely the range dependence of I(〈μ〉). The usefulness of I(〈μ〉) as an approximation is therefore restricted to ranges up to some transition range rT. The transition range is inversely proportional to the relative magnitude ε of the deviation from the mean profile 〈μ〉, and also to ratios constructed from the characteristic (focusing) length of the channel, the injection depth of the beam relative to the channel axis, and the correlation length of the ensemble of profiles. Expressions for rT are derived in several limiting situations, and potential applications are discussed.
The attenuation of ultrasound in biological fluids
Authors: Narayana P. A. , Ophir Jonathan , Maklad N. F.
Published: October 2005

We have measured the ultrasonic attenuation parameters α0 and n of several biological fluids in vitro in the frequency range 2.8–6.8 MHz, using frequency shift and/or a sinc (⋅) sidelobe ratio technique. The parameters α0 and n describe the frequency-dependent attenuation via a power-law model, i.e., α( f)=α0 f n, where f is frequency. The samples investigated were blood, pus, cyst fluid, bile, and infected hematoma. It was found that the values of α0 span approximately the range between 0.03–0.3 dB cm−1 MHz−n, and values of n range from about 1.1–1.3.
Calculation of nonlinearity parameter for seawater
Authors: Endo Harumi
Published: October 2005

The nonlinearity parameter B/A of seawater from 25%o to 40%o has been calculated in the temperature range from 0° to 35 °C and at pressure of up to 1000 bars by use of the acoustic and thermodynamic approaches. Their agreement with an accuracy of 2% is within ±0.1 over most of the range from 0° to 35 °C up to 800 bars. The ratio B/A at each concentration increases monotonically with pressure in the range covered. The increase rate at higher temperatures becomes slower than that at low temperatures. It is found that the B/A at low pressure increases with increasing temperature, and at higher pressures decreases. However, the quantity B/A at moderate pressure near 500 bars is relatively insensitive to temperature changes and then reveals a small temperature minimum. Although the B/A at a fixed P and T increases with increasing salinity, the increment of B/A decreases as pressure increases. To find a relationship between B/A and T or P, we obtained the thermodynamic expressions for [∂(B/A)/∂T]P and [∂(B/A)/∂P]T, respectively.
Monitoring for vowels in isolation and in a consonantal context
Authors: Rakerd Brad , Verbrugge Robert R. , Shankweiler Donald P.
Published: October 2005

The identifiability of isolated vowels (/V/) was compared to that of vowels in consonantal context (/pVp/) when subjects performed a monitoring task. On successive blocks of trials in a test series, the subjects listened for instances of one or another of nine monophthongal vowels (/i,I,q,1,v,a,c,U,u/) and identified each test item as being an instance or not. On average, resulting false alarm errors occurred significantly less often in the /pVp/ condition, consistent with the previous finding that vowel perception may be aided by consonantal context. This beneficial effect of context was found to be restricted to the class of open vowels, however, with perception of the close vowels being somewhat hindered by context. The error data for misses also showed an interaction between context and vowel height. Various accounts of the interaction are considered.
Frequency-dependent elastic properties of rubberlike materials with a random distribution of voids
Authors: Varadan V. K. , Varadan V. V. , Ma Y.
Published: October 2005

A self-consistent multiple scattering approach with suitable pair-correlation function has been employed to study the frequency-dependent properties of rubber materials containing a random distribution of voids. Numerical results indicate that for a certain range of concentration of voids, the resonance of the voids controls the frequency-dependent effective properties such that the imaginary part of the effective wavenumber becomes greater than the real part. This indicates that the propagating wave will be damped out over a distance smaller than a wavelength. The so-called ‘‘superviscous’’ propagation seems to occur over a bandwidth controlled by the concentration of voids (c). The bandwidth increases with increasing concentrations in the range 0<c<0.15. For c>0.15, this kind of behavior does not exist for the material system considered.
Intensity and sound pressure level
Authors: Behar A.
Published: October 2005

Acoustical terms are often misused—a typical example being the use of ‘‘intensity’’ instead of ‘‘sound pressure level.’’ In the last year and in 7 out of 11 issues of JASA, this fundamental mistake appeared in titles of papers. Authors and reviewers should pay more attention so the quality of papers published in our Journal do not contain such a fundamental error.
Discrimination of fundamental frequency of synthesized vowel sounds in a noise background
Authors: Scheffers Michael T. M.
Published: October 2005

An experiment was carried out, investigating the relationship between the just noticeable difference of fundamental frequency (jndf0) of three stationary synthesized vowel sounds in noise and the signal-to-noise ratio. To this end the S/N ratios were measured at which listeners could just discriminate a series of changes in  f0 in the range from 10% to 0.5%. Similar measurements were obtained for pulse trains and for pure tones as a reference for the results. A measure of S/N ratio based on an approximation of the critical bandwidth appeared to provide a fairly good predictor of the masked threshold of each signal, measured in a second experiment. Using this measure, it was found that a given change in the fundamental of a pulse train could be discriminated at a lower S/N ratio than in a pure tone with a frequency equal to that fundamental. The results for the vowel sounds were found to be in between those for a low-frequency pure tone and those for a pulse train. Owing to the signal-generation method (viz., changing  f0 by changing the sampling frequency), three cues could in principle be used to discriminate a change in the fundamental of a vowel: A change in the residue pitch, a change in the pitch of a single prominent harmonic, or a change in the spectral envelope of the signal. It can be inferred from the results that the subjects used that particular cue which yielded best performance. Which cue was optimal depended not only on the vowel but also on  f0 and on the presented change in  f0. It seems, however, that the pitch of a single harmonic was the cue most often used. Another interesting result is that changes in  f0 greater than about 5%, could for each signal be discriminated when the signal was just above masked threshold.