bluuebunny/crossref_metadata_2025 huggingface dataset:
Perceiving fractal noises in pitch, loudness, and duration
Authors: Schmuckler Mark A. , Gilden David L.
Published: October 2005

Three experiments examined discrimination among a family of fractals possessing different power law spectra. In experiment 1, random numbers sequences were generated so that their spectra had slopes (in log-log plane) or 0 (white noise), − 1 (flicker noise), and − 2 (brown noise). The distributions were used to produce sequences that could vary in pitch, loudness, or duration. Listeners categorized each sequence as being of one of the three noise types. Thurstonian scaling revealed that noise segment classification could be accomplished on the basis of pitch and loudness variation, but not on the basis of duration changes. Experiment 2 explored simultaneous pitch and loudness variation, finding dimensional redundancy for white/flicker discrimination suggestive of a pitch-loudness interaction. Experiment 3 examined sensitivity for discrimination of noises as a function of the power law exponent. Listeners heard two sequences of tones and indicated whether they had the same or different power law exponent. The range of exponents spanned the interval [0, −3], with the stimuli under comparison differing by 0.2, 0.4, 0.6, or 0.8. ROC analyses revealed some variation in sensitivity to the fractal exponents as a function of position in the [0, −3] interval.
Noise generation by newly created bubbles
Authors: Longuet-Higgens Michael S.
Published: October 2005

Air bubbles trapped at a free surface by breaking waves, or emerging from an underwater nozzle, or splitting, or coalescing in various ways, all undergo strong “shape oscillations” before settling into a spherical shape. In a fully nonlinear theory (not the usual linearized approximation), the shape oscillations can be shown to emit a strong monopole radiation of sound. Second-order calculations suggest that this is a significant, perhaps dominant, source of bubble noise in the ocean. Resonance between the second harmonic of each shape oscillation and the radial “breathing” mode of a given bubble leads to enhanced emission of sound at or near the breathing-mode frequency. There can also be coupling between a bubble and the free surface from which it is detached. Oceanic data and laboratory experiments relating to these predictions will be discussed, particularly the well-known experiments by Fitzpatrick and Strasberg (1957); also, broadband spectra of oceanic noise from a variety of locations will be discussed.
Ambient sound beneath breaking surface waves
Authors: Farmer David M. , Vagle Svein
Published: October 2005

Ambient sound measurement obtained beneath breaking surface waves over the frequency range 44–20 000 Hz are described. The observations are obtained from an instrument located 24–32 m beneath the surface during experiments both in open ocean and continental shelf environments. Individual breaking events are positively identified using underwater video photography, and it is confirmed that they radiate sound throughout the observed frequency range. A distinctive feature of the acoustic signature is the presence of spectral peaks, the frequency of which may remain generally consistent from one breaking event to the next, but which can change significantly over the course of a storm or from one storm to another. A theory is proposed based on the concept of trapping of a portion of the sound in the surface waveguide formed by the ocean surface bubble layer. Simultaneous measurements of the bubble population and size distribution obtained with a multifrequency inverted echo sounder allows calculation of the resulting (dispersive) sound-speed anomaly profile. Comparison of the observed spectral peak frequencies with those predicted yields good agreement.
Bubble sources of the Knudsen spectra of ocean noise
Authors: Medwin Herman
Published: October 2005

The source levels, frequencies, and damping constants of individual bubbles, catastrophically generated by spilling breakers, have been measured in the NPS Ocean Acoustic Wave Facility, OAWF. OAWF consists of a 17-m-long×1-m-deep×1-m-wide water channel in which waves are driven by an oscillating plunger. The waves grow and then form spilling breakers on the surface of a large, nonreverberant volume at the end of the channel. Two calibrated hydrophones suffice to identify the type of bubble, its surface location, and its dipole axial source strength. The sources of the breaker noise have thereby been identified as transient bubbles of resonance frequency 350 Hz to 50 kHz, with lifetimes from 2 to 20 ms. These bubbles radiate as dipoles from positions within a few hundred μm to a few mm of the surface. The long-term average of the bubble radiations has the same slope as the Knudsen wind wave spectra at sea, 5 dB/oct. [Work supported by the Office of Naval Research.]
Some statistics of sea noise
Authors: Crowther P. A.
Published: October 2005

Surface-derived ambient noise has been measured on a long-term basis by the use of a bottom-mounted narrow beam sonar, in depths of 65–80 m, over windspeeds 3–20 m/s, in frequency bands over 6–50 kHz. Results are presented as time histories and derived statistics. It is found that, after detection and smoothing, the noise power in a narrow beam fluctuates widely, “spikes” in time being identifiable with individual breaking waves. The statistics of fluctuation is found to depend on beamwidth, the coefficient of variation being typically 10/beamwidth in meters. Time autocorrelation functions of noise power are presented, which also depend on beamwidth and windspeed. The mean horizontal speed of the noise sources can be deduced and is approximately half the windspeed in a well-developed sea. These results are interpreted in the light of known spectra of sea waves, and simple assumptions about the nature of the wave breaking process.
Perspectives on sound in Sanskrit literature on natural philosophy
Authors: Prasad M. G.
Published: October 2005

It is known that the literature in Sanskrit language that includes “Vedas and Upanishads” represents some of the earliest known references dealing with various topics of natural philosophy and science. The historical background of these references has been estimated to be of several thousand years BC. In spite of the lack of a precise historical account of the development of the literature in Sanskrit language, there have been many works of translation from Sanskrit to several European languages including English. The topic of “sound” seems to have received fundamental importance in Sanskrit literature in the areas of both natural philosophy, science, and art. This paper describes the perspectives on sound as found in Sanskrit literature particularly in “Vedas and Upanishads.” The paper also discusses some of the important applications of the knowledge and perspectives on sound that are reflected in acoustic phonetics of Sanskrit language, Vedic chants, and classical music and dance of India.
An experimental design for testing the validity of the collapsible tube model of the larynx
Authors: Berke Gerald S. , Arnstein David P. , Smith Marshall E. , Natividad Manuel , Conrad William A.
Published: October 2005

The in vivo canine model of the larynx was used to measure transglottic pressures and airflow during phonation. Direct intraglottal pressures were measured at various positions within the glottis. Conditions of supraglottal resistance were also simulated. Pressure drop-flow curves were compared with data on collapsible tubes. The in vivo canine model of the larynx demonstrates a number of the described features similar to oscillation in collapsible tubes. [Work supported by VA Technical Merit Review Grant.]
/s/ and /∫/ as a function of linguapalatal groove place and width
Authors: Fletcher S. G. , Newman D. G.
Published: October 2005

Palatometric feedback was used to guide three normal talkers in systematically varying the place and width of the sibilant groove. Linguapalatal contact was manipulated to produce 2- to 12-mm-wide grooves in each of seven-eight rows of contact sensors spanning the front third of the palate. Both the rows and the sensors in each row were 2 mm apart. The subject's task was to establish a stipulated groove, then channel air through it to produce a “hissing” noise. These responses were tape recorded then later low-pass filtered, digitized, and shaped to fit normal sibilant intensity, duration, and onset/offset times. The resultant sounds were played ten times each in random order to 14 listener judges in individual listening sessions. The listeners were instructed to label each sound heard as “definitely an ‘s,’” “probably an ‘s,’” “probably an ‘sh,’” or “definitely an ‘sh.’” Statistical analysis of these observations yielded main effects for both groove width and place. The findings will be discussed from the viewpoints of biomechanical and auditory shaping hypotheses. [Work supported by NIH Grant NS 24697.]
Electropalatographic evidence for long-range coarticulatory effects in VCVCV sequences
Authors: Recasens Daniel
Published: October 2005

This study reports some electropalatographic data on tongue dorsum contact showing coarticulation to extend up to three or four phonetic segments away from the target gesture. Coarticulatory effects in degree of palatal contact for V = [i] vs [a] and for C = [∫] vs [t] were analyzed along all possible VC [ə] CV combinations of those same vowels and consonants. It was predicted that the temporal domain of coarticulation would vary according to the degree of tongue dorsum contact involved during the production of the contextual phonemes. The findings show that, in general, the coarticulatory effects are inversely dependent on the degree of tongue dorsum contact for those phonemes that are immediately adjacent to the target (e.g., C2 for the V3-dependent anticipatory effects). Moreover, carryover effects reached the segments located at the other side of [ə] in specific circumstances (e.g., when C2 = [t]). The theoretical implications of the results of this experiment are discussed in light of recent findings in the literature.
On /r/ - /w/ substitution in child speech
Authors: Lindland Kevin G. , Slawinski Elzbieta B.
Published: October 2005

The role of phonetic context in /r/ - /w/ substitution errors was investigated. The subject population consisted of 20 children with normally developing articulation skills, 33 to 52 months in age. The subjects were given a minimal pairs discrimination task in order to assess the accuracy of their perception for “r” and “w” consonants. Subsequently, the subjects were administered a test of production in which they had to identify and produce the names of objects drawn from a common set presented to them. Acoustical analysis of the misarticulated /r/ reveals three patterns, which can be distinguished acoustically by different formant values and/or their changes in time. Results indicate that substitution errors are highly context dependent, with the percentage errors for the entire population ranging from 0% when /r/ was in the final position to 36% in the initial position in the word. These findings suggest the possible importance of a cognitive strategy, as well as the presence of shifted categorical boundaries. Results are discussed in terms of the developing child's limited mental resources and perceptual skills.
A new approach for speech dynamics studies
Authors: Mrayati Mohamed , Carté René , Castelli Eric
Published: October 2005

A new dynamic model for vocal tract configuration based on a theoretical quantification of dynamic area function A(x,t) into distinctive regions having areas S(r,t) is proposed. Articulation is transcoded into changes of these S(r,t) through three modes [Mrayati et al., Speech Comm. 7, 257–286 (1988)]. A dynamic control strategy of S(r,t) is extremely simple to apply and correlates well with human speech organ musculature movements. This strategy transprojects the classical longitudinal displacement of the constriction into a transversal area function variation of S(r,t). This dynamic strategy can incorporate easily context-dependent allophonic variations, such as: coarticulation, reduction, assimilation …. The distinctive regions, along the vocal tract, are associated with specific consonants and vowels. These new “phonetic regions” concept can be used to vectorially code target configurations for consonants and vowels. The new concept relates, in a simple fashion, changes in the acoustic domain to changes in regional cross-sectional areas. Some results on VV, CV, VC, and VCV command strategies are presented.
Closant curve characteristics of infant vocalizations
Authors: Robb Michael P. , Bauer Harold R.
Published: October 2005

The “closant curve” reflects the change in ratio of the frequency of consonants (closants) to vowels (vocants) in early vocal and speech development of young children (Bauer, 1988). Closant curves were plotted for seven children's monthly vocalizations across the first 2 years of life. The closant curves were then compared to monthly data plots of each child's vocal fundamental frequency and vocalization duration. Assessment of developmental trends for the phonetic and acoustic data was based on visual examination of the data plots and curve fitting procedures. The resulting developmental functions are discussed in relation to the reported closant curves underlying environmental and physiological characteristics of language acquisition. These analyses provide an acoustic and phonetic framework from which to study early speech development. [Work supported by U. H. Research Council.]
Determination of the elastic constants of anisotropic composite materials via laser photoacoustics
Authors: Castagnede Bernard , Sachse Wolfgang , Thompson Michael O.
Published: October 2005

This paper presents the solution of the inverse anisotropic medium problem in which the elastic constants of an anisotropic composite material are determined from ultrasonic wave speed measurements made in nonprincipal directions of a specimen. The ultrasonic waves were generated via the point-source/point-receiver technique using a 2-ns pulsed Nd:Yag laser as a source and a 2.5-mm-diam capacitive or a 1.3-mm piezoelectric transducer as a receiver. Data were acquired during an isoangular scan of the source relative to one of the principal acoustic axes of symmetry. In each waveform, the arrivals of the quasilongitudinal and the two quasishear bulk modes were measured. The elastic constants of the material were then recovered using an optimization algorithm. Experimental results are presented for a transversely isotropic composite material. It was found that the nonlinear fit between the experimental and the recovered slowness values is excellent. Some discrepancies are observed for the two shear modes. These are shown to be related to the complexity of the detected signals.
Biological cell characterization using inverse acoustic scattering
Authors: Chen X. , Apfel R. E.
Published: October 2005

The acoustic scattering properties of biological cells, especially red blood cells, have been studied both theoretically and experimentally. An apparatus consisting of two confocally positioned transmitter/receivers and a Coulter-type volume sensing zone was reported previously [J. Acoust. Soc. Am. Suppl. 1 84, S163 (1988)]. An inverse scattering theory incorporating the experimental data yields three bits of infomation for each cell: volume, density, and compressibility. Since both the density and compressibility of biological cells are almost linearly related to protein concentration, it might be possible to determine the protein concentration of these cells. The results obtained from several types of biological cells using this method will be presented. The changes in density and compressibility, and hence protein concentration, when cells undergo volume changes through swelling, shrinking, or heating will be assessed. [Work supported by the U. S. National Institutes of Health through Grant 5R01GM30419.]
Sound generated by fluid interfaces—Insights into sea surface noise
Authors: Cato Douglas H.
Published: October 2005

A theory of sound generation by motion of fluid interfaces has been developed and applied to the sea surface. The only assumptions are that the density discontinuity at the sea surface can be modeled as a Heaviside function, and that the relationship ∂p/∂ρ = c2 applies within a fluid, where p is pressure, ρ is density, and c is sound speed. The theory shows that the motion of an interface generates sound with source strengths depending on the difference between the product of the density and the square of the sound speed on either side of the interface. Distributed monopole and dipole sources are possible. The insight this provides into sound generated by surface motion, bubble cloud oscillation, bubbles, droplets, and spray will be discussed. Most sources would be expected to show dipole directionality with axes close to, but not necessarily, vertical. Wind-induced turbulent pressure fluctuations across the sea surface are shown to be not a significant source of noise in the ocean, even though they have been considered as such.
How raindrop impacts and air bubble resonances contribute to underwater sound spectra
Authors: Laville Frédéric , Abbott Grayson D. , Miller Matthew J.
Published: October 2005

Using underwater sound to measure the rate of rainfall over the oceans is a very promising approach. However, there are conflicting models on the spectral contributions of the two rainfall sound sources (raindrop impacts on the water surface and air bubble resonances). For example, each of these two sources has been alternatively proposed as a major contributor to the spectrum as well as the origin of a spectral peak around 15 kHz. High-speed data acquisition and processing of underwater sound recorded in a lake under real rain and artificial raindrop conditions were used to identify the two sound sources in the time domain and determine their respective contributions to the long-term spectrum. Bubble resonances were found responsible for the spectral peak around 15 kHz and very dependent upon surface condition, whereas raindrop impacts were found responsible for a broadband spectrum rich in frequencies below 15 kHz and less dependent on surface conditions. [Work supported by DOE.]
Modeling word-boundary phonological processes in continuous speech recognition
Authors: Williams Briony , Thompson Henry
Published: October 2005

The automatic recognition of continuous English speech, as against isolated words, must contend with phonological processes at word boundaries that can distort the mapping between surface form and lexical entry. Past recognition systems have used generative phonological rules in a precompilation stage to expand the working dictionary. However, this greatly increases the bulk and complexity of the lexicon. It is also possible to use analytic rules to undo the putative effect of phonological processes at the time of recognition. However, this can lead to the postulation of nonwords and to slower processing times. A solution that blends the advantages of both approaches, with the disadvantages of neither, is to use finite state transducers (FSTs) as filters on permitted matchings between input strings and lexical entries. These were implemented in a recognition system, and found to increase the percentage of words recognized by three percentage points, at a cost of halving the signal-to-noise ratio, compared to the same system without phonological FSTs.
Spontaneous emission of acoustic waves from shock fronts
Authors: Xu F.
Published: October 2005

The criterion for spontaneous emission of acoustic waves from shock fronts, 1 − M2 − (v0/v1)M21 − M2 + (v0/v1)M2 <j2 dvdp < 1 + 2M, was obtained by D'yakov et al. [Zh. Eksp. Teor. Fiz. 27, 288 (1954); V. M. Kontorovich, ibid. 33, 1525 (1957); G. R. Fowles, Phys. Fluids 24, 220 (1981)]. Based on the previous work of the interaction of acoustic waves with a shock front [F. Xu, Acta Mech. Sinica (English edition) 3, 113 (1987)], a new criterion is obtained in this paper: C0 > ω/ky < C1. Then, the causes for the discrepancy between these two criteria are discussed and some comments on the previous papers are made. In addition, the amplification of the shock front is investigated and the results are compared with those of previous works [G. R. Fowles, Phys. Fluids 24, 220 (1981); P. Harris, Tech. Rep. ARLCD-TR-78041 (1978)].
Temporal versus spatial considerations in reference loudspeaker systems
Authors: Augspurger George L.
Published: October 2005

Until recently, the performance of loudspeaker systems used for music mixdown or cinema reproduction has been specified in terms of on-axis anechoir frequency response or 1/3-oct room response. Now, readily available test methods allow measurement of delay characteristics as easily as conventional frequency response. So-called “time-corrected” playback systems, in fact, exhibit a wide range of temporal and spatial behavior. Theoretical and measured data are presented that indicate strong interaction between spatial side effects and listening room acoustics. Subjective and objective trade-offs are discussed for three typical playback situations.
Broadband acoustic-field simulations from standard ray theory
Authors: Flatté Stanley M. , Colosi John , Duda Timothy F. , Rovner Galina , Martin Jan
Published: October 2005

The complete wave field over a small region around 1000 km from a pulsed source is reconstructed in two ways. First, all the rays from the source to a vertical array of receivers at 1000 km are found, along with their travel times, number of caustics, arrival angles, and intensities. The pattern of wave fronts in a space at a given time is then reconstructed on a closely spaced grid surrounding 1000 km by treating these rays in an appropriate way. Second, the parabolic equation method is used at multiple frequencies to synthesize a pulse. The two fields are compared. Finally, the effect of internal waves is simulated by use of the first method, introducing random fluctuations on the travel times and arrival angles of each ray. [Work supported by ONR, Code 1125OA.]
Sound-speed determination using matched field processing techniques
Authors: Karangelen C. , Diachok O.
Published: October 2005

Matched field processing is shown to be effective for estimating sound speed in a deep, range-independent ocean environment. The amplitude and phase of signals from a distant source measured on a large aperture vertical array are sensitive to changes in sound speed. This sensitivity is exploited to infer the environmental sound-speed profile by matching predicted and measured amplitude and phase. A sound-speed profile model is developed based on a modified version of Munk's canonical sound field equation. This model is used to determine sound-speed profile using a 15-Hz signal from a 240-m explosive source detected on a 675-m vertical array at a range of 50 km in a deep-water Pacific environment, characterized by a classical range-independent sound channel at 700 m. The search for the best estimated profile is conducted by varying the sound channel axis strength and depth in the modified Munk equation while maintaining a constant sound-speed profile at great depths. Differences between the estimated and measured profiles are less than ± 2 m/s; the sound channel axis depth is determined within 20 m of the measured axis depth. Extension of this approach to include mesoscale features and greater range is discussed.
Piezoelectric ceramic compositional development
Authors: Berlincourt Don
Published: October 2005

The original piezoelectric ceramic material was unmodified barium titanate. It was little used except as compositionally modified. The earliest work was by W. P. Mason, and this led to improved characteristics for sonar transducers and then to phonograph cartridge applications. By the mid 1950s, lead titanate zirconate materials were shown by B. Jaffe to have higher piezoelectric coupling and application at a much higher temperature. Over the next 10 years, many modified compositions were developed. These led to much improved sonar systems and ultrasonic cleaners and to applications in ultrasonic bonders, stereo phonograph cartridges and even printers. The new compositions also made possible applications such as piezoelectric ceramic filters, gas ignition devices, and camera flashbulb actuators. More recently, specialized ceramics have been developed based on lead titanate and lead metaniobate, but major efforts have been directed to applications of lead titanate zirconate compositions, which now touch virtually every home and automobile. The history of compositional studies with ferroelectric ceramics is reviewed and the types of characteristics achieved are summarized. The compositional additives and some general principles to explain their behavior are discussed.
The Mason horn, extension and applications
Authors: Beshers D. N.
Published: October 2005

Warren P. Mason, in the mid 1950s, introduced the inverted exponential horn as a concentrator of ultrasonic waves. In 1968, he published an extended account of a stepped approximation to the exponential horn. Part of the horn was replaced by a half-wavelength piece that functioned as a mechanical transformer with a step reduction in diameter at the quarter-wave point. The specimen was shaped like a dumbbell to give a mass-spring-mass resonance with a further step down in diameter. The result was a substantial reduction in area and thus an increase in vibratory stress. The apparatus may also be described as a composite resonant oscillator: Each of the three elements, transducer, transformer, and specimen, is at resonance. Mason developed the theory of this stepped horn only for perfect resonance, and allowed for damping only in the reduced part of the specimen. Here, his theory is extended to allow for small deviations from resonance, such as must occur in practice, and for damping in the other elements. When the damping and the deviations from resonance are small, simple sum rules hold. Variations in specimen design are also considered. Applications, past and future, will be discussed.
Tutorial: Signal processing methods for machinery diagnostics
Authors: Lyon Richard H.
Published: October 2005

Conventional machinery monitoring systems rely mainly on signal energy processing methods such as power spectrum analyses in the frequency domain and envelope analysis in the time domain. Other, less well-known, techniques can be very useful in extracting diagnostic information from the vibration or sound produced from the machine as it operates. These methods include cepstral analysis as an order analysis or deconvolution procedure, and am and fm demodulation using the Hilbert transform to study variations in loading or speed of rotational devices. They also include various system modeling methods such as Kalman filtering and functional expansions of the transfer function. Illustrations of the use of these methods, and descriptions of situations in which each may be helpful will be presented in this tutorial.
Sound power level and directivity pattern determination of a space shuttle solid rocket booster
Authors: Adams Thomas S.
Published: October 2005

The sound power level and directivity pattern of sound were determined for a space shuttle solid rocket booster motor during a horizontal test firing at the Morton Thiokol Wasatch Test Facility in Utah. Six Bruel & Kjaer type 2230 precision sound level meters were placed at 60-deg intervals around the rocket motor at a distance of 1000 ft, except the location directly behind the nozzle (2600 ft). These analyzers were used to determine the overall sound-pressure level, between 20 and 20 000 Hz, at each of the six locations during the test. Two Cetec-Ivie model IE-30A audio spectrum analyzers were used to determine the shape of the spectrum at a one-third octave band resolution. The equipment was turned on approximately 2 h prior to the scheduled test and was unattended during the test. Various adjustments were made to the data to account for topographic and atmospheric effects. The sound produced was predominantly low frequency, and the directivity pattern exhibited major acoustic lobes 60 deg to both sides of the line of fire. A total overall sound power level of 196 dB was determined. [Work funded by NASA Stennis Space Center, Mississippi.]
On the difficulty of fricative perception by hearing-impaired subjects
Authors: Zeng Fan-Gang , Turner Christopher W.
Published: October 2005

By comparing conditions under which both normal and hearing-impaired subjects were presented with equivalent degrees of audibility for fricative perception, the current study isolated the factors of lack of audibility from that of loss of suprathreshold discriminability of fricative-perception cues. One moderate and two severe cochlear-impaired subjects identified synthetic fricatives, with either full cues (frication and transition) or frication cue alone, presented in closed-set response tasks across a wide range of presentation levels. The results showed that hearing-impaired subjects yielded equivalent recognition performance to the normals when given an equivalent degree of audibility for the frication cue, but they obtained poorer-than-normal performance if only given an equivalent degree of audibility for the transition cue. Considering the 20- to 25-dB intensity difference between the frication and transition cues, the difficulty that hearing-impaired subjects have in perceiving fricatives under normal circumstances may be due to two factors: the lack of audibility in terms of frication cue and the loss of discriminability in terms of transition cue. [Work supported by SU Senate Research Grant and Deafness Research Foundation.]
Modeling a vented push-pull hearing-aid fitting in situ as a feedback control system
Authors: Preves David A. , Newton James R.
Published: October 2005

For the investigation of acoustic feedback encountered with push-pull amplification for in-the-ear hearing aids, a simulated fitting was considered a feedback control system to include the acoustic leakage from the hearing-aid receiver traveling back to the microphone. A Gennum WS531 amplifier was connected to an earmold shell made for the right ear of KEMAR. This shell contained a Knowles EK 3024 microphone, a Knowles BK1613 receiver, and a 0.09-in.-diam by 0.8-in.-long vent hole. Transfer function magnitude and phase were obtained for the microphone, amplifier, receiver, for the open loop while breaking the signal path between microphone and amplifier, and for the closed loop, all with maximum gain before onset of oscillation. The acoustic feedback transfer function was determined by subtracting the forward loop transfer function from the closed loop transfer function. A circuit synthesis program was used to fit the transfer function measurement data into mathematical transfer function expressions. Comparisons were made between the synthesized transfer functions and the measured transfer functions.
Reducing the effects of target misalignment in an adaptive beamformer for hearing aids
Authors: Greenberg Julie E. , Zurek Patrick M. , Peterson Patrick M.
Published: October 2005

Previous work [Peterson et al., Acta Otolaryngol. (in press)] has shown that a multimicrophone adaptive beamforming system for hearing aids reduced noise effectively at input target-to-jammer ratios (TJRs) up to about 0 dB. However, that beamforming algorithm assumed that the target signal was identical at the microphones, and violation of this assumption caused degrading target cancellation when input TJR was greater than 0 dB. The present system uses a measure of intermicrophone correlation to estimate short-time TJR and inhibits adaptation dynamically when TJR exceeds a selected threshold. Computer simulations of a two-microphone system in anechoic and moderately reverberant environments show that this method reduces (speech) target cancellation when the identical target assumption is violated. The improved system reduces noise at both positive and negative input TJRs (±20 dB), with target misalignments up to 10 deg. Additionally, this processing scheme reduces the misadjustment noise caused by strong target signals in the adaptive feedback loop. [Work supported by NIH.]
Effect of number of channels on the performance of a multichannel compression hearing aid
Authors: Yund E. William , Buckles Krista M.
Published: October 2005

Previous research [E. W. Yund, H. J. Simon, and R. Efron, J. Rehab. Res. Dev. 24, 161–180 (1987)] has demonstrated the effectiveness of an eight-channel compression hearing aid for individuals with sensorineural hearing loss in the task of speech discrimination in a background of speech-band noise. In the present study, similar multichannel compression systems with 4, 6, 8, 12, and 16 channels were compared with each other and with linear (frequency equalized amplification) and unprocessed (flat amplification) conditions. The results on 15 hearing loss subjects showed increasing speech discrimination performance as the number of channels increased, although the difference between 4 and 8 channels was much greater than the difference between 8 and 16 channels. Confirming the previous results, the performance advantage of the multichannel compression conditions over the linear and unprocessed conditions was greater at low signal-to-noise ratios (S/N), e.g., + 5, 0, or − 5 dB as opposed to + 10 or + 15 dB. [Work supported by VA Rehabilitation Research and Development Service.]
P centers, C centers, or what else?
Authors: Pompino-Marschall Bernd , Kühnert Barbara , Tillmann Hans G.
Published: October 2005

Some questions concerning the underlying nature of the so-called P-center phenomenon (the acoustic anisochrony of rhythmically regularly produced/perceived sequences of monosyllables) are discussed in the light of recent EMG data from experiments with rhythmically produced /Cak/ syllables (with /C/ = /p/, /pf/, /pl/, /pt/, /pfl/). The differences between /p/ vs /pf/ syllables resemble the timing differences in the microbeam data of Browman and Goldstein [Haskins SR-93/94, 85–102 (1988)]. Whereas they found a stable timing relation between the “articulatory mean” of the initial consonants (termed C center) and the acoustical offset of the following vowel, in terms of their task dynamic description, however, the phase relations between consonant and vowel gestures should remain the same. This would result in a reduced stiffness of the vowel gesture for the syllables containing consonant clusters. The observations of the present study would also correspond to a reduction of stiffness for the vowel but the numerical data seem to contradict a stable phasing relation. In connection with the psychoacoustic interpretation of the P-center phenomenon, an argument is made against a strictly intrinsic timing interpretation (based on articulatory regularities) of the effects of rhythmical production/perception.
On precise in-duct measurement of fan noise
Authors: Swim William B.
Published: October 2005

Measurement of fan noise requires extremely careful control of a multiple of variables—including fan operating point and test system configuration—to obtain useful results. Frequently, fan noise measurements are contaminated by unrecognized fan or test system changes and spurious noise sources. A new fan test system was designed to eliminate extraneous noise sources and to provide good control of the fan and the test system. This system uses round sheet metal ducts to connect the test fan to inlet and discharge anechoic terminations. Rotating microphone probes, one in the inlet and one in the discharge, are used to measure induct SPL while the fan's air performance is being determined. Two test systems, one with a mean diameters of 8 in. and the other of 16 in., have been built. Measurements have been made on both axial and centrifugal fans of sizes from 6 to 20 in. Representative results of these measurements and data from recent studies of microphone probe design and surface microphone systems use for in-duct noise measurements will be discussed. Data will also be presented on the influence of tip clearance on axial fan noise.
Air-moving devices and active noise control
Authors: Olson D. A. , Wise S. S. , Eriksson L. J. , Allie M. C.
Published: October 2005

Active noise control presents unique advantages for the reduction of noise from air-moving devices. These include excellent low-frequency performance, minimal flow restriction, and ease of installation. In this paper emphasis will be placed on the use of active noise control for silencing discharge noise from fans. A recently developed digital system using adaptive signal processing will be described. Results will be presented from a number of centrifugal and vaneaxial fans. Application guidelines have been developed to ensure that the system performance is maximized. These guidelines will be presented using results from actual field testing.
Localization and biophysical characterization of an adaptation mechanism in amphibian hair cells
Authors: Assad John A. , Hacohen Nir , Corey David P.
Published: October 2005

Hair cells of the bullfrog sacculus adapt to maintained displacement stimuli in a manner that indicates an active movement of the elastic element that exerts stress on transduction channels. Transduction and adaptation were present in dissociated hair cells studied with whole-cell, patch-clamp recording and video microscopy. The adaptation rate—as well as the position of the resting current-displacement [I(X)] curve—depended on extracellular calcium and on membrane potential, suggesting that calcium passes into the cytoplasm to reach its site of action. Following a step hyperpolarization, the adaptation increased within milliseconds, suggesting that the calcium site is within a few microns of the ion channels through which calcium enters. The voltage dependence of the resting I(X) curve, together with the “gating springs” hypothesis for transduction, predicts a 40-nm movement of the bundle away from the kinocilium when the cell is depolarizcd. This was observed.
Bidirectional transduction by haircells
Authors: Searle C. L. , Secker-Walker H. E. , Wilson T. A. , Freeman D. M.
Published: October 2005

Given the two-port description of a hair cell from Weiss [Hear. Res. 7, 353–360 (1982)], with the Davis model for the angle-dependent conductance representing the transduction channels [Leong and Weiss, Hear. Res. 20, 175–195 (1985)], and the data of Crawford and Fettiplace [J. Physiol. 364, 359–379 (1985)], Howard and Hudspeth [Neuron 1, 189–199 (1988)], and Holton and Hudspeth [J. Physiol. 375, 195–227 (1986)], analysis shows that (1) the feedback inherent in the two-port must be positive, and (2) degenerative (outward-flowing) channels must be present. The static nonlinear behavior of the system, and the small and large signal dynamics, have been studied. The system displays Q multiplication at low signal levels, bandwidth increase with signal level, a compressive nonlinearity at CF, and continuous oscillations if the parameters are slightly misadjusted—all well-documented physiological phenomena.
Mechanical purification and protein constituents of amphibian hair cell stereocilia
Authors: Shepherd Gordon M. G. , Barres Barbara A. , Corey David P.
Published: October 2005

In order to understand the molecular basis for transduction and adaptation in hair cells, it is necessary to know the identities and relationships of proteins in the mechanically sensitive stereocilia. A rapid and specific purification procedure has been developed for stereocilia of the bullfrog sacculus, that relies on mechanical adhesion of the stereocilia to nitrocellulose paper (“bundle blot purification”). The high purity and yield of the preparation were demonstrated by light and scanning electron microscopy. Electrophoretic separation of proteins revealed 12–18 major bands. Actin stained with phalloidin and was a prominent band at 42 kD. Fimbrin (68 kD) was identified by immunoblot. Both bands were present in cores prepared by treating stereocilia with triton to remove membrane and soluble proteins. Calmodulin (17 kD) and calbindin (28 kD) were identified by immunocytochemistry or by comigration. The electrophoretic pattern differs substantially from that of brush border microvilli, which are structurally analogous; in particular, the calmodulin-regulated, myosinlike 110-kD protein of brush border was not detected in stereocilia.
Vibrational modes of drumheads in various types of drums
Authors: Rossing Thomas D.
Published: October 2005

As vibrating systems, drums can be divided into three categories: those consisting of a single membrane coupled to an enclosed air cavity (e.g., kettledrums); those consisting of a single membrane open to the air on both sides (e.g., tom toms, congas); and those consisting of two membranes coupled by an enclosed air cavity (e.g., bass drums, snare drums). In kettledrums, the enclosed air raises the frequencies of the axisymmetric modes while air loading lowers the frequencies of the musically important modes with one to four nodal diameters, and shifts them into a nearly harmonic series. In two-headed drums, coupling of the two heads through the air cavity results in mode pairs in which the heads move either parallel or antiparallel to each other. Dipole, quadrupole and other multipole symmetries are observed in the radiated sound fields. Some drums, such as the kettledrum and tabla, convey a strong sense of pitch; others such as the bass drum and snare drum do not. The sense of pitch may be enhanced by loading a portion of the drumhead, as in the Indian tabla and mridanga.
Calculations of timpani normal modes for arbitrary kettle shapes
Authors: Davis Robert E. , Tubis Arnold
Published: October 2005

A practical boundary-integral-equation method is developed for extending a previous study of timpani normal modes [R. S. Christian et al., J. Acoust. Soc. Am. 76, 1336–1345 (1984)] to the case of rigid kettles that are azimuthally symmetric but have arbitrary contours. Room acoustics effects are ignored. Calculations and comparisons of results with experimental data are carried out for various kettle shapes including cylindrical, hemispherical, hemispherical with a cylindrical sleeve, hemiellipsoidal, and paraboloidal. It is found that the closeness of the musically important modal frequency ratios, f11:f21:f31:f41 to 2:3:4:5, can be quantitatively accounted for by calculations of the effect of air loading. These desirable ratios may be approximately achieved if V/πa3 ≳ 1.08 and ρ0a/σ≈1.51, where V is the kettle volume, a is the timpani membrane radius, ρ0 is the mass density of air, and σ is a real mass density of the membrane. The timpani spectrum depends weakly on the detailed kettle shape as long as these conditions are satisfied. An infinite rigid baffle in the membrane plane may be used to greatly simplify the calculations without having a significant effect on the results.
Sound spectra of ancient Chinese bells in the Shanghai Museum
Authors: Rossing Thomas D. , Hong-Fan Zhu
Published: October 2005

Sound spectra were obtained from ten bells from the 5th–12th centuries BC by striking them at the appropriate sui and gu locations. Although considerably different in size, shape, and ornamentation, all the bells had oval cross sections and thus exhibited two-tone behavior [T. D. Rossing et al., J. Acoust. Soc. Am. 83, 369 (1988)]. Frequency ratios of the two fundamental (2,0) modes range from 1.11–1.31 (181–473 ¢). In most spectra, it is possible to identify two partials most likely associated with two (3,0) modes that have frequencies 2.32–3.29 times the lowest (2,0) mode. The spectra of these bells are compared to those of other Chinese bells, including those of the renowned Zenghou Yi.
Relations between neural-count and intensity-jnd functions for single auditory-nerve fibers
Authors: Hellman William S. , Hellman Rhona P.
Published: October 2005

The equation d'σ(x) = ΔN(x), where x = I/I0, is applied to physiological measurements reported by Delgutte [The Psychophysics of Speech Perception (1987)] for single auditory-nerve fibers. Intensity-jnd functions are derived for individual fibers with the neural-count function as input. The experimentally obtained neural counts are represented by a hyperbolic tangent function, and the measured dependence of the count variance on the count is approximated by a combination of linear functions. Here, ΔN(x) can be obtained without approximation for arbitrary values of Δx for a hyperbolic tangent function; it can also be represented by a Taylor series in Δx. The exact equation for ΔN(x) gives a U-shaped jnd function in agreement with the numerical procedure used by Delgutte. By comparison, the Taylor series for ΔN(x) converges to the exact solution in fifth order. [Partially supported by the Rehabilitation Research and Development Service of the VA.]
Perilymph volume flow in the fistulized cochlea
Authors: Inamura Naoki , Salt Alec N. , Thalmann Ruediger , Vora Arti
Published: October 2005

It has been previously reported that in guinea pigs the rate of perilymph volume flow is extremely low (<2 nl/min) when the cochlea is in the physiological, sealed state. In the present study, flow rates were monitored when the otic capsule was perforated. Flow was measured by comparing dispersal of a tracer (trimethylphenylammonium) in both directions along scala tympani with ion selective microelectrodes. Perforation of the cochlear apex produced a high rate of flow averaging 0.4 μl/min. The rate was reduced considerably when cerebrospinal fluid (CSF) pressure was released by opening the dura at the foramen magnum. Under these conditions, the mean rate of 6.9 nl/min. Blockage of the cochlear aqueduct was even more effective, reducing flow to a mean value of 1.4 nl/min. In the latter condition, the flow rate was not significantly different from the normal, sealed cochlea condition. These results demonstrate that the high rates of flow in the fistulized cochlea arise from a nonphysiologic entry of CSF into scala tympani through the cochlear aqueduct. For experimental purposes, this nonphysiologic flow can be reduced by releasing CSF pressure, or better still, by occluding the cochlear aqueduct. [Work supported by NIH Program Project Grant No. PO1 NS24372.]
Comparison of long-term average speech spectrum at ear-level and standard recording positions
Authors: Cornelisse Leonard E. , Gagné Jean-Pierre , Seewald Richard C.
Published: October 2005

The long-term average speech spectrum (LTASS) was obtained from ten male and ten female talkers for two microphone positions. Dual channel recordings of speech were recorded and played back for spectrum analysis. One channel recorded the speech signal obtained from a microphone placed 30 cm directly in front of the talker, while the second channel recorded the speech signal obtained from a microphone placed at the talker's ear. A spectrum analyzer (BK type 4132) was used to compute the overall levels and 1/3-octave-band levels. The analyses indicated significant differences in overall and the 1/3-octave-band levels between the two recording positions. The mean overall level at the position of the ear was approximately 6 dB higher than the overall level at the standard recording position. Also, 1/3-octave-band analyses indicated that the ear-level LTASS consisted of more low-frequency energy and less high-frequency energy than the LTASS recorded at the standard position. The implications of these results for hearing aid selection will be addressed.