bluuebunny/crossref_metadata_2025 huggingface dataset:
Median adaptive filtering for spectral line enhancement
Authors: Haweel Tarek I. , Clarkson Peter M.
Published: October 2005

Adaptive line enhancement using the well-known LMS algorithm is a technique for the enhancement of low-level periodic components of unknown frequency in a background of broadband noise. The LMS filter operates by iteratively minimizing a quadratic index using an instantaneous estimate of the gradient of the quadratic performance surface to update the filter coefficients. Provided the adaptation rate is sufficiently small, the filter usually performs well. If the input data are corrupted by impulsive noise, however, the filter is badly degraded, and the adaptive process may actually fail. This paper addresses this problem by defining a median least-mean-squares (MLMS) algorithm where the instantaneous gradient estimate of the LMS is replaced by the sample median of that parameter. The MLMS filter is compared, through simulation, with the LMS and with an alternative gradient smoother that utilizes the sample mean (the ALMS algorithm). Results demonstrate that the MLMS is largely insensitive to the presence of impulsive interference and performs comparably with the LMS and ALMS when no interference is present.
Accurate spectral estimation of multiple sinusoids using an FFT
Authors: Burgess John C.
Published: October 2005

Accurate estimates of amplitude, frequency, and phase of multiple sinusoids in a signal can be obtained using a method described earlier [J. C. Burgess, J. Acoust. Soc. Am. Suppl. 1 83, S92 (1988)]. The FFT size required for specified spectral resolution is discussed, as well as some computational aspects.
Measurements of vehicle noise source height for noise barrier design
Authors: Glegg Stewart A. L. , Yoon J. R.
Published: October 2005

The height of a noise barrier required to give a certain insertion loss is strongly dependent on the elevation of the noise source above the ground. Currently, noise barrier design methods assume equivalent point source heights for different types of vehicles, but these have never been substantiated by measurements. This study used a microphone array, with parametric processing, to evaluate the equivalent noise source height of 100 different vehicles. The results give the noise source position as a function of frequency between 200 Hz and 2.5 kHz. The relative importance of engine, stack and tire noise on the source height is apparent. It was found that the equivalent source on heavy trucks is at 1.2 m, medium trucks 0.7 m, and small vehicles 0.6 m, above the ground. [Work sponsored by Florida Department of Transportation.]
Universal versus language-specific aspects of voicing-dependent vowel duration
Authors: Laeufer Christiane
Published: October 2005

All languages show vowels to be slightly longer before voiced obstruents. A phonological rule of vowel lengthening has been proposed for English because the variation seems to be much greater than in, say, French. This comparison, however, has been based on studies that do not control for word length, syllable structure, or for whether or not the languages have final devoicing. A comparison of voicing-dependent vowel duration in English and French monosyllables shows that phrase finally and medially under focus, obstruent voicing is a significant factor influencing the duration of preceding vowels in both languages. In unfocused medial position, on the other hand, the difference between the durations as a function of consonant voicing is much smaller. Moreover, if differences in syllabification are taken into account, the languages are even more similar. That is, in French, consonants tend to be resyllabified with the following syllable, unlike in English. These results suggest that languages universally exhibit fairly similar physiologically conditioned voicing-dependent variation in vowel durations. Under certain conditions, including word length, syllabic structure, stress, position in word, and utterance and speaking tempo, the difference is enhanced, while in others it is obscured. Thus a general phonetic account in terms of timing control is preferable to a language-specific phonological rule.
Very high levels of calmodulin (CAL) in outer hair cells (OHC) of organ of Corti (OC)
Authors: Takahashi Kuniaki , Thalmann Isolde , Thalmann Ruediger , Zenner Hans-Peter , Kraus Peter
Published: October 2005

Using immunohistochemical techniques, Slepecky et al. [Hear. Res. 321, 11–22 (1988)] demonstrated a selective accumulation of CAL, an ubiquitous calcium binding protein, in the inner hair cells (IHC) and OHC of the OC. By quantitative polyacrylamide gel electrophoresis, it was found that in chinchilla OC, CAL is by far the highest in OHC, where levels of 6%–8% of total protein are reached. For comparison, the highest known accumulation of CAL (in the heads of spermatozoa) amounts to 12%. CAL is considerably lower in IHC (about 1%), and is not detectable with usual protein loads in other cell types of the OC. CAL is also very low in stria vascularis and Reissner's membrane. The relative distribution of CAL in the OC is similar to that of glycogen, where the exceedingly high level of 600-mmol/kg dry weight is reached in OHC [Thalmann et al., Laryngoscope 80, 1619–1645 (1970)]. This circumstantial evidence suggests that activation of phosphorylase kinase may be an important function of CAL in OHC, whereby ultimately energy for contraction is mobilized from glycogen. Most likely CAL-induced activation of myosin light chain kinase also takes place as part of the contractile process. [Work supported by NIH and DRF.]
Modes of vibration and sound radiation from a snare drum
Authors: Huan Zhao , Rossing Thomas D.
Published: October 2005

The orchestral snare drum is a two-headed instrument about 35 cm in diameter and 13–20 cm deep. Snares of metal or gut are stretched across the lower (snare) head, which is generally thinner than the upper (batter) head. The drum can be played with or without the snares. Coupling of the two heads through the air enclosed by the shell results in mode pairs in which the heads move parallel or antiparallel to each other. When struck near the center, the sound spectrum is dominated initially by a strong partial (typically around 180–200 Hz) due to parallel (01) motion of the two heads. This mode radiates energy rapidly, so after a short time, its amplitude falls below that of the modes with one or more nodal diameters. Partials due to the latter modes dominate the spectrum when the drum is struck off-center. Dipole, quadrupole, and other multipole symmetries are observed in the radiated sound field. Vibrations of the shell contribute very little to the sound spectrum. [Research supported by Remo, Incorporated.]
A wide dynamic range speech test for the evaluation of compression hearing aids
Authors: Neuman Arlene C. , Bakke Matthew H. , Levitt Harry
Published: October 2005

Wide dynamic range test materials have been developed for the evaluation of compression hearing aids. Anomalous sentences containing four monosyllabic test words per sentence were recorded by a female talker. The four test words in each sentence were then scaled (using a waveform editing system) to represent four different sound-pressure levels covering a 30-dB range (in 10-dB steps). Performance on this speech material was compared to a performance-intensity function obtained using monosyllabic word lists at the same four sound-pressure levels. Testing was done for a linear and two compression systems (compression ratio of 2.5, instantaneous attack time, release times of 20 and 140 ms). For all three conditions, frequency shaping based on a half-gain rule was used. Ten listeners with sensorineural hearing loss served as subjects. Results of the experiment indicated that the wide dynamic range test material was sensitive to the dynamics of the compression system being evaluated, while the monosyllabic word materials presented at discrete levels was not. [Work supported by NIH Grant No. PO1 NS17764.]
The P300 response to verbally presented digits
Authors: Molt Lawrence F. , Saul Richard S.
Published: October 2005

Latency and amplitude functions for the P300 component of the slow auditory evoked potential were measured for 20 normal hearing adults. Stimuli consisted of digitized numbers, 1–10 (excluding 7), of 480-ms duration. The method of presentation for stimuli pairs was in an oddball paradigm in which the frequently occurring digit was presented 80% of the time and the rarely occurring digit 20% of the time. Pairs consisted of acoustically similar or acoustically divergent waveforms (e.g., 4-5 vs 8-2, respectively). Evoked potentials were recorded from Fz, Cz, and Pz sites. While both mean latency and mean amplitude values differed among the three sites, with the greatest amplitude and shortest latency recorded from site Cz, differences were not statistically significant. Similarly, mean amplitude and latency value differences did not reach statistical significance for acoustically similar versus acoustically divergent digit pairs at any of the three sites.
Effects of intense sound exposures and aspirin on overshoot
Authors: Champlin Craig A. , McFadden Dennis
Published: October 2005

Brief tonal signals were presented 2 or 190 ms following the onset of a 200-ms broadband masker. These two conditions of signal delay were tested before and after a series of exposures to a tone intense enough to induce temporary threshold shifts (TTS). The difference in detectability between the short- and long-delay signals (i.e., overshoot) was reduced when a TTS of at least 10 dB was induced, but not when smaller amounts of TTS were induced. This reduction in overshoot was due to an improvement in the postexposure detectability of the signal delayed by 2 ms; the detectability of the signal delayed by 190 ms was unchanged pre- and postexposure. Recovery from both the TTS and the reduction of the overshoot followed slightly different time courses. Data are currently being collected for the two conditions of signal delay before and after the production of an aspirin-induced threshold shift. [Work supported by NIH.]
Binaural difference in rat
Authors: Psaltikidou Maria , Gaumond Roger P.
Published: October 2005

The binaural difference (BD) is the sum of monaural acoustic brainstem responses (ABRs) minus the binaural ABR. Its construction implies operation of a nonlinear element with binaural input. The BD in five rats was studied using insert earphones that provided 60 dB of interaural isolation measured in the occluded external meatus. Scalp to chin BD consisted of a double-peaked complex with a following trough occurring during peaks IV and V of the sum of monaural ABRs. BD magnitude was defined as the maximum peak-to-trough waveform amplitude. As stimulus intensity increased, BD magnitude increased linearly with the magnitude of ABR peak I. A bilaterally symmetric stimulus generally yielded smaller BD magnitude than did stimuli with small bilateral asymmetry (8-dB interaural intensity difference, 64-μs interaural time difference). With larger asymmetry, BD magnitude decreased. The linear increase of BD magnitude with ABR peak I magnitude is consistent with operation of an element that applies a rectification nonlinearity to the difference of binaural signals during peaks IV and V of the ABR.
Modal analysis technology review
Authors: McDonald Paul , Wiley Garth
Published: October 2005

An overview of current experimental modal analysis technology is provided with consideration to its use with finite element analysis. The implications of this available technology to acoustic applications are included. The advantages and disadvantages of various practical modal analysis methods will be covered. A look to the future is provided in assessing the influences of computational and graphics advances anticipated in workstation hardware. Various application examples will be reviewed to illustrate methodologies employed.
A new approach to system identification in conjunction with digitally Swept MultiExciter Sine Control
Authors: Underwood M. A. , Stroud R. C.
Published: October 2005

This paper discusses a new adaptive control technique that applies optimization ideas to the general problem of updating an imprecise system Impedance Matrix, I, as part of the feedback control process that is associated with MultiExciter Swept Sinewave Control. Extensions of the approach to other MultiExciter problems such as waveform and random vibration control will also be explored. The effects on the ability of the Adaptive Control Algorithm to converge by low matching of phase and amplitude between the input channels, low coherence between the exciter drive vector and the control point response vector, and low input/output dynamic range are also analyzed. The effects of the above instrumentation errors on the optimization problem associated with experimental modal analysis are also explored as well as the potential benefits that an enhanced frequency response matrix estimate, as a result of the above system identification process, can provide.
Detection of global modes in a foundation and hull structural model from drive point mobility measurements on the foundation
Authors: McCoy Kevin , Lyon Richard H.
Published: October 2005

Closely coupled modes of two connected structures may be considered to be “global modes,” involving the combined motion of both structures. Such modes will account for the greatest energy transfer between the structures, and it is therefore desirable to find ways of detecting them. This paper is interested in the case in which it is convenient to make measurements on only one of the two structures, in this case a “foundation model” that is connected to a “hull model.” The foundation (a beam) is excited by a shaker, and its drive point mobility is measured. At certain frequencies, an interaction with the hull (plate) modes can be detected. This interaction can be modified by using small masses to change the coupling between the structures and the foundation resonances. In this way, the global modes can be separated truly from structural resonances that simply happen to lie close to each other.
A generalized approach to random noise synthesis
Authors: Hsueh Keng D. , Hamernik Roger P.
Published: October 2005

A generalized approach to the synthesis of Gaussian and non-Gaussian random noises having a preselected amplitude spectrum has been developed. An application of the inverse discrete Fourier transform following certain manipulations in the frequency domain allows both Gaussian (skewness = 0 and kurtosis = 3) and non-Gaussian (variable skewness, kurtosis, and crest factor) random noises to be constructed. The theoretical background and the results of a numerical simulation will be presented, which demonstrate the functional relation between the various spectral manipulations and the descriptors of the synthesized random noise. Analog signals were synthesized using the above method in conjunction with PC-based instrumentation. The results show that the approach is viable and that the synthesized random waveforms can be easily tailored to simulate a variety of real-world noise/vibration signals, e.g., high kurtosis (impulsive) industrial noises, helicopter noises, and missile vibration signals. The application of such a methodology to acoustical testing, environmental stress screening, structural testing, etc., will be discussed. [Work supported by NIOSH.]
Observations of acoustic backscatter from oceanic wastewater outfalls
Authors: Proni John R. , Dammann W. Paul
Published: October 2005

Narrow beam echo sounders were used to characterize the spatial distribution of wastewater plume material from six ocean outfalls in the South Florida area. Volume scattering strengths were computed and plotted as a function of depth and horizontal distance for 200-kHz echoes. Reduction in peak scattering strength with increased range from the outfall locations correlated well with reduction in concentration of Rhodamine-WT dye introduced into the undiluted wastewater at a concentration of 1 ppm. Sound power reflection coefficients ranging from 10−5 to 10−9 were observed for these wastewater plumes. The data presented demonstrate the degree to which the sound power reflection coefficient for a distribution of scatterers is dependent upon the concentration of those scatterers, and show the utility of the acoustical method in water-mass characterization.
Public reaction to low levels of aircraft noise
Authors: Wesler John E.
Published: October 2005

Several recent instances have raised the issue of public annoyance from the noise of airplanes flying at relatively high altitudes or at relatively large distances from the nearest airport. Public complaints have arisen about airplane flights over northern New Jersey as the result of changes in flight patterns associated with the major New York airports, even though in many instances those airplanes are flying at 15 000 ft or higher. Concerns have arisen regarding the noise levels on the ground from the new, swept-blade, advanced turboprop airplanes when they are flying at cruise altitudes of 30 000 ft and higher. Complaints about aircraft noise over national parks have resulted in a Congressional requirement to measure those noises and determine their severity. These noise levels do not meet the usual criteria for annoyance or interference with individual activity, whether in terms of average level or single events. A better understanding of the intrusive effects of low levels of community noise is needed, especially where present in areas of relatively low ambient noise levels.
Effects of changing the A-weighting design goal
Authors: Wong George S. K.
Published: October 2005

The proposal by the International Organization for Standardization (ISO), document ISO/TC43/SCI N644, dated October 1988, to modify the A-weighted design goal by imposing additional attenuation to implement well-defined frequency cutoffs at 20 Hz and at 16 kHz to ensure consistent A-weighted measurements is examined. The merit of the above proposal is in doubt since the high-frequency cutoff at 16 kHz eliminates both audible and high-frequency components and can result in underestimation during A-weighted noise assessment, particularly when the noise is impulsive. The consequences of the above changes are serious: Future databases of A-weighted measurements will be incompatible with those from the past, and acoustical communities in every country will suffer financial loss due to the need to replace most of their measuring instruments to comply with the proposed A-weighting. A better approach to ensure consistent A-weighted measurement is to impose tighter tolerances in the high-frequency region of the A-weighting, such as those specified in ANSI S1.4A-1985 amendment to ANSI S1.4-1983.
Sound exposures and hearing thresholds of musicians in a major symphony orchestra
Authors: Royster Julia Doswell , Royster Larry H. , Killion Mead C.
Published: October 2005

Seventy noise dosimetry samples were obtained for musicians during rehearsals and performances of a major symphony orchestra. Audiograms were obtained for 59 musicians. The Leq during measurement periods ranged from 76–102 dBA (median = 90dBA), corresponding to on-the-job daily equivalent values of 72–98 dBA (median = 86 dBA). Using the ISO 1999.2 model, this exposure would be expected to produce 5–8 dB of NIPTS after 30 yr for typical ears (0.5 fractile) or 8–10 dB of NIPTS for very susceptible ears (0.05 fractile). The musicians' average thresholds were better than those for age-matched reference nonindustrial noise-exposed populations without occupational noise exposure, and only slightly worse than those for highly screened populations representing aging alone. However, audiogram patterns indicated a slight notch, suggesting a contribution from NIPTS. Bilaterally averaged thresholds for musicians in different instrument sections were essentially equivalent, but violinists and violists grouped together showed significantly poorer thresholds at 2–4 kHz in the left ear than in the right ear. Measured Leq correlated with HTLs at 3–4 kHz. Playing in a symphony orchestra appears to present a mild risk of hearing damage, but musicians in every age group displayed average hearing thresholds better than the general population.
Growth of threshold shift from intense impulses: Implications for basic loss mechanisms
Authors: Price G. Richard
Published: October 2005

For a given number of impulses, increases in intensity are normally associated with greater threshold shifts. The details of this growth should, in principle, reveal something of the basic mechanisms responsible for the loss. Experiments have been conducted in which the ears of 30 cats have been exposed to 50 impulses with their peak energies located at 4000 Hz and with peak pressures ranging from 135–145 dB SPL. Threshold shifts grew about 7.0 dB for every dB increase in SPL above 134 dB. Virtually the same result can be calculated from data for the chinchilla ear [R. P. Hamernik, J. H. Patterson, and R. J. Salvi, J. Acoust. Soc. Am. 81, 1118–1129 (1987)], which may indicate that this function is characteristic of mammalian ears. On the other hand, exposures to other impulses in both animals have shown much lower rates of growth; however, these differences can be explained by conductive nonlinearities in the middle ear and/or a limited range of growth of threshold shift.
The dependence of the experimental performance of a thermoacoustic refrigerator on geometry
Authors: Hofler T. , Suzalla M.
Published: October 2005

The results of recent measurements on an experimental thermoacoustic refrigerator will be presented. In simple terms, thermoacoustic heat transport is caused by the interaction of a high-amplitude standing wave with a stationary “stack” (a stack of parallel plates having uniform plate spacing). The focus of these measurements is on the effect changes in stack geometry has on refrigerator performance as determined by temperature span and thermodynamic efficiency. The experimental geometry variations consist of changing the position of the stack in the standing wave and changing the spacing between adjacent plates in the stack. The results show that refrigerator performance is very sensitive to the location of the stack in the standing wave, as expected from the theory of N. Rott, but performance is much more sensitive to plate spacing than predicted by the theory. [Work supported by the Office of Naval Research and the Office of Naval Technology.]
Nonlinear propagation of sound beams in various media
Authors: Zabolotskaya E. A.
Published: October 2005

Approximate nonlinear equations describing the propagation of finite amplitude sound beams in liquids, isotropic solids, and crystals are suggested. The equations have been derived under the assumptions of small perturbation and slow variation of the wave shape, which are due to diffraction and nonlinearity. Diffraction is taken into account within the quasioptical approximation. Nonlinearity is considered up to quadratic terms for the cases of wave propagation in liquids and crystals, as well as for longitudinal wave propagation in solids. As for nonlinear propagation of transverse waves in an isotropic solid, cubic terms in the magnitude of the perturbation are retained.
Frequency wavenumber analysis of seismoacoustic waves in an ice layer
Authors: Giellis G. , Yang T. C.
Published: October 2005

Seismoacoustic waves traveling in an ice layer over a deep Arctic Ocean are studied using frequency wavenumber analysis. Ice-ridge-generated noise can travel in the ice via ice-trapped waves and water-borne waves coupled to the ice. {Previous hydrophone data [B. Buck and J. H. Wilson, J. Acoust. Soc. Am. 80, 256–264 (1986)] indicated that the noise originated from the bottom of a ridge.} The data are simulated by a point source in the water (for the water-borne waves) and a point source in the ice (for the ice-trapped noise) using the SAFARI numerical code with a planar receiver array of vertical axis geophones. Frequency wavenumber analysis is applied to the simulated data to determine the wavenumber of the various waves traveling in the ice. The methodology and preliminary results of this analysis will be reported.
Duration effects on vowel perception by hearing-impaired listeners
Authors: Besing J. , Collins M. J. , Cullen J.
Published: October 2005

The effect of temporal differences on the identification of vowel tokens by hearing-impaired listeners was studied using six hearing-impaired listeners. Stimuli were synthetically generated and varied in duration of steady state, durations of the initial and final transitions, and F1, F2, and F3 location. Listeners were required to identify the given vowel token from a set often possible alternatives. In some cases, the hearing-impaired listeners' responses were similar to a group of normal-hearing listeners and, in some cases, they were very different and appeared somewhat idiosyncratic. The hearing-impaired subjects were different from each other; further, the labeling performance was different within the same subjects across different vowels. In addition, subjects with similar audiograms performed differently from one another on some vowels and similarly on others. These results and the particular effects of the alterations in the durations of the various components will be discussed. [Work supported by NIMH.]
Consonant types and tone in Siamese revisited
Authors: Drury John
Published: October 2005

This is a report of a production study of the effects of consonant types and lexical tones on F0 before and after the release of the consonantal gesture in the speech of one speaker of Central Thai. The study is designed to determine whether the duration or magnitude of the effects of consonants on F0 varies consistently with the lexical tone of the syllable in which the consonants occur. The main findings are that (1) segments that depress F0 have a smaller effect on the F0 of lower tones than of higher tones, while (2) segments that elevate F0 have approximately the same effect on F0 for all tones; and (3) consonants have only a brief effect on post-release F0, as in previous study [J. Gandour, J. Phon. 2, 337–350 (1974)]. The findings are discussed in terms of a model in which the measurable F0 of an utterance can be predicted by combining segmental F0/time functions with prosodic F0/time functions.
Perceptual units of the infant cry
Authors: Tsukamoto Taeko , Tohkura Yoh'ichi
Published: October 2005

Perceptual units for category identification of infant cries have been studied. Three cry categories discussed in this paper are the hunger cry, the call cry (i.e., cry calling for infant-mother interaction), and the anger cry. The original samples have been classified into these three categories. In order to generate stimuli to be used in the perceptual experiments, each of the cry samples is first segmented into single-segment units according to breath groups of the cry samples. Next, the single-segment units are combined with each other in temporal order to generate two-, three-, five-, and seven-segment unit stimuli. In addition to the multisegment unit stimuli thus obtained, the single-segment units and the three original samples are used as one-segment unit stimuli and full-segment unit stimuli, respectively, in the perceptual experiments. In identifying the cry stimuli, subjects are instructed to make a forced choice among the three cry categories. The experimental results show that category identification rates are greatly dependent upon the number of segments making up each stimulus. However, the identification rates are temporarily saturated at two-segment units in the call cry and at three- to five-segment units in the hunger and anger cries. This fact indicates that the units with two to five segments are the perceptual units. Temporal duration of the perceptual units across all three categories is similar (i.e., about 6–8 s).
Variations in measured voice quality related to linguistic and nonlinguistic factors
Authors: McRoberts Gerald W. , Faber Alice
Published: October 2005

Voice quality (VQ) in English has generally been treated as a suprasegmental, not varying across individual speech segments. Ladefoged [in Vocal Fold Physiology, edited by Bless and Abbs (1983)] has measured the spectral correlates of VQ differences in languages in which both VQ and tone (F0) are phonemically contrastive. Since VQ variation is not contrastive in English, it is in theory available to enhance other linguistic contrasts, or may vary as a function of nonlinguistic factors, such as fatigue or affect. Following Ladefoged, VQ was measured as the difference between the amplitude of F0 and the amplitude of the most intense harmonic in F1 using discrete Fourier transform spectra. Two separate corpora of utterances by subjects without known voice pathology were analyzed. One corpus compared qualitatively similar stressed and unstressed vowels (Vs) produced by a single speaker in “fresh” and “fatigued” circumstances. The other compared utterances produced by two trained actors in contrastively and normally stressed conditions, and simulated positive and negative effective conditions. These analyses suggest that this measure of VQ differentiates between Vs in (1) stressed and unstressed conditions, and (2) normally and contrastively stressed conditions. However, it did not differentiate between Vs in fresh and fatigued conditions or in the simulated affective conditions. The results are discussed in terms of the use of VQ and its relation to F0 for conveying linguistic and nonlinguistic distinctions. [Work supported by NIH Grants NS 07237, HD 1994, NS 24655.]
Speaker identity verification of spectrally ambiguous and distinguishable speech
Authors: Logan Robert J. , Feustel Timothy C. , Velius George A. , Pastore Richard E.
Published: October 2005

Feustel et al. [J. Acoust. Soc. Am. Suppl. 1 83, S55 (1988)] evaluated performance of humans in a speaker identity verification (SIV) task with the intent of improving the performance of a LPC-cepstral based verification algorithm. This study uses the general methodology and signal processing techniques developed by Feustel et al. to further investigate the nature of potential invariant acoustic information (within a speaker) used by humans to verify a speaker's identity. Subjects were presented natural speech or modified natural speech stimuli drawn from a corpus of spectrally distinguishable and spectrally ambiguous utterances. The results suggest that humans can use information from both the glottal (source) waveform and spectral envelope. Overall, the glottal waveform provides more reliable information. The results will be discussed in this context.
Speaker-independent word recognition with Lombard speech
Authors: Hanson Brian A. , Applebaum Ted H. , De Haan Gregory R.
Published: October 2005

To investigate the robustness of a small vocabulary, speaker-independent word recognizer trained on normal speech, testing was done using Lombard speech. The Lombard database was collected from talkers exposed to 85 dB SPL white Gaussian noise played through headphones. Recognition tests utilized a hidden Markov model recognizer with separate codebooks for static and dynamic cepstral features [Gupta et al., in Proc. IEEE ICASSP-87, 697–700 (1987)]. The front-end analysis used either standard linear prediction (LP) or a version of perceptually based linear prediction. (This PLP has filtering functions with 3-dB bandwidths of 1 Bark, a correction to Eq. (3) in Hermansky et al. [Proc. IEEE ICASSP-85, 509–512 (1985)].) The effects of the analysis technique, analysis order, distance measure (cepstral or weighted cepstral), and spectral features (static, dynamic, or integrated) were studied. The utility of static and dynamic features differed greatly according to the analysis method and distance measure. For example, for low (5th)-order PLP, the dynamic features alone performed much better than either static or integrated features. Alternatively, for high (14th)-order LPC with weighted cepstral distance, the dynamical features alone performed much worse than either static or integrated features, yet contributed to improved recognition rates with integrated features.
New architecture for sub-band coding
Authors: Yazu Takashi , Morito Makoto
Published: October 2005

Most sub-band coders (SBC) have been realized by quadrature-mirror filter (QMF) with frequency shift and low-pass filter (LPF). However, the QMF method has the following problems: (i) Band separation is limited to the power of 2; (ii) arithmetic errors are accumulated because of the cascade connection of QMF; and (iii) it is necessary to adjust the time delay if the number of QMF connections is different. This paper presents a new architecture for SBC with flexible band assignment, less arithmetic calculations, and small storage memory. In this SBC architecture, a finite impulse response filter (FIRF) is used instead of the QMF. The center frequency of each band is selected according to the equation Fs/Fk = D *n/m, where Fs is the sampling frequency of the input signal, Fk is the center frequency of the k th band, D is the decimation factor, and m, n are integers. According to this relationship, frequency shift and the LPF can be performed at the same time. In this case, data memory for the FIRF is drastically reduced because the input of the FIRF is the same for every band. With the new SBC architecture, the 16- to 12-kbps sub-band coder/decoder was implemented on a single DSP. Band selection and suitable bit allocation are discussed.
Speechreading sentences I: Development of a sequence comparator
Authors: Bernstein L. E. , Demorest Marilyn E. , Eberhardt Silvio P.
Published: October 2005

Previous research suggests that many lexical errors in the speechreading of sentences can be explained in terms of visual phonemic errors. However, description and quantification of perceptual errors at the phonemic level requires specification of stimulus-to-response alignments. Because speechreading produces numerous errors, including phoneme insertions, deletions, and/or substitutions, alignment is a nontrivial problem. This paper describes development of a sequence comparator that can be used to obtain alignments automatically for phonemically transcribed sentences. The comparator employs a weights matrix that reflects presumed visual distances between all possible segmental stimulus-response pairs to find the alignment that minimizes overall stimulus-response distance. Initially, the comparator used weights based on viseme groupings, but these weights resulted in multiple, equal-distance, alternative alignments. More effective weights were obtained empirically via multidimensional scaling of phonemic confusions. Vowel data were obtained from Montgomery and Jackson [J. Acoust. Soc. Am. 73, 2134–2144 (1983)] and consonant data from a nonsense syllable identification task, which employed 22 consonants spoken by the same talkers who produced the sentence stimuli for this study [Bernstein et al., J. Acoust. Soc. Am. 85, 397–405 (1989)]. [Work supported by NIH.]
General linearized solutions of the equations Of bubble pulsations
Authors: Cross Timothy M. , Finch Robert D.
Published: October 2005

Previous attempts to find linearized solutions of the equations for vapor bubble pulsations [e.g., see Finch and Neppiras, J. Acoust. Soc. Am. 53, 1402–1410 (1973)] assumed that oscillations would be the result of imposed sonic excitation. It was shown by Nicholas and Finch [12th Int. Cong. Acoust., Toronto, paper 14-2 (1986)] that nonlinear solutions of the equations in fact often showed exponential expansions or collapses. In this paper it is shown that if the governing equations are linearized using a Taylor series expansion, without assuming either oscillatory or exponential behavior, then there are regimes of bubble radius in which one or the other of these two is a necessary result. The simple case of a bubble with an insoluble gas content in a nonconducting liquid is considered. There can then be shown to exist a critical radius, with a value given by 2σ/3γP, where σ is the surface tension, γ is the ratio of specific heats, and P is the gas pressure. Below this size the behavior is exponential, above it, oscillatory. The results can be interpreted in terms of the stability of an open-loop control system.
Monolithic piezoelectric transducers
Authors: Harrison W. B.
Published: October 2005

This paper will review new piezoelectric transducers concepts where layers of lead zirconate titanite (PZT) have been electroded and then cofired into a monolithic structure that was ready for poling and use in various types of transducers. These techniques were initially developed for underwater military applications, but are now ready to be applied to commercial transducers. Several methods for constructing these transducers will be reviewed where either tape processed or more conventionally pressed PZT shapes were used. Typical applications and advantages for this technology in several acoustic transducers will also be described. Relationships between the thermal expansion characteristics of thickness and transverse mode transducers will also be presented along with data on a new, more temperature stable PZT composition. [Work supported, in part, by Office of Naval Research.]
Electrostrictive materials
Authors: Newnham Robert E. , Xu Qichang
Published: October 2005

Two types of electrostrictive materials will be described: (1) relaxor ferroelectrics with diffuse phase transitions, and (2) antiferroelectric ceramics with a field-induced ferroelectric phase. Transducers made from relaxor ferroelectrics such as lead magnesium niobate (PMN) and lead zinc niobate (PZN) are operated above Tc in a broad transition range characterized by the presence of nanometer-size microdomains that impart large electromechanical coupling coefficients to the ceramic. These relaxor materials show no piezoelectric effect under ac drive because of the absence of any net spontaneous polarization, but microdomain fluctuations give rise to large electrostrictive strains (∼10−3) with no hysteresis. Under dc bias, PMN ceramics show piezoelectric d33 coefficients as large as 1500 pC/N, three times those of PZT. The second type of electrostrictive transducer is based on a field-induced phase change from an antiferroelectric structure to a ferroelectric phase. Ceramics in the PbZrO3-PbTiO3-PbSnO3 ternary system are used as bistable transducers for digital motions where shape-memory hysteresis can be an advantage. Other uses for electrostrictive transducers include adaptive optic systems, micropositioners, and smart transducers in which the size of the piezoelectric coefficient can be adjusted with a dc bias field.
Rare earth magnetostrictive materials
Authors: Clark Arthur E.
Published: October 2005

The rare earth elements Tb and Dy possess the largest known magnetostrictions by far (nearly 1%). While these giant magnetostrictions are currently available only at cryogenic temperatures, magnetostrictive strains as high as 0.3% have been obtained at room temperature in compounds of the rare earths. In this presentation, focus will be on the recently developed Terfenol-D compounds prepared by free standing zone (FSZ) and modified Bridgman (MB) techniques. Because near balance of the magnetic anisotropy can be achieved in these compounds, huge magnetostriction “jumps” can occur. During these jumps, energy is transformed between an internal magnetic state to an external mechanical state causing large spontaneous magnetostrictions. Special emphasis in the presentation will be on: (1) magnetostrictive properties at low and high temperatures, (2) dependence of the magnetization and magnetostriction on external stress, (3) preparation techniques and growth morphology of the highly magnetostrictive compounds, (4) Delta E effect, and (5) theoretical models. Recent designs of low-loss transducer modules using high-strain rare earth alloys will be presented. [Work supported by the NSWC Internal Research Program and the Office of Naval Technology.]
The spherical wave absorption coefficient of a patch of material
Authors: Nobile Matthew A.
Published: October 2005

Recently, the concept of a spherical wave absorption coefficient, aspher, was introduced [M. A. Noblie, Proc. NOISE-CON 88, 567–572 (1988)] to account for the fact that when spherical waves are incident upon an acoustical material, the absorption coefficient is not the same as it would be under (less realistic) plane-wave incidence. In fact, aspher is not an intrinsic property of the material but depends on the distance of the sound source from the material. In this paper, an expression for the spherical wave absorption coefficient of a finite-size patch is derived (in the cylindrical coordinate system considered, the patch is either a circular or annular disk). As will be shown, this quantity can be approximated experimentally from measurements of the reflection coefficient at several oblique angles over the surface of the sample using the two-microphone transfer function method in a hemianechoic environment [an extension of M. A. Nobile, Proc. NOISE-CON 87, 611–616 (1987)]. This can be the basis for a simplified method of comparing the absorptive properties of two materials.
Ultrasonic investigation on different structure and composition copolymers
Authors: Hu Jiankai , Jing Zhou , Wenying Xu
Published: October 2005

Ultrasonic attenuation and velocities have been measured for three kinds of copolymers: MMA-BMA, MMA-MAA, and ST-MAA containing different kinds of cross-linkers. Experimental results indicate that ultrasonic attenuation increases with increasing BMA content in MMA-BMA, with adding cross-linkers EGD or BA in MMA and in ST-MAA; velocity decreases with increasing BMA content in MMA-BMA. The temperature dependences of ultrasonic attenuation in MMA-BMA show that the glass transition temperature Tg was decreased with increasing BMA content, and Tg could be easily determined with ultrasonic method. The results have been explained by thermoelastic mechanism and scattering mechanism.
Ultrasonic investigation of a BiSrCaCuO high Tc superconductor
Authors: Hu Jiankai , Zhang Qianlin , Tingzhang Deng , Cai Weili
Published: October 2005

The temperature dependences of ultrasonic attenuation and velocity in a high Tc superconductor BiSrCaCuO were measured by the ultrasonic pulse echo method, in the temperature range 77–300 K. Ultrasonic attenuation anomalies were found at 95, 135, and 250 K; the slope of ultrasonic velocity was changed at these temperatures respondingly. There are two superconductive phases in BiSrCaCuO samples at 87 and 120 K found from x-ray examining. The ultrasonic anomalies at 95 and 135 K are related to the superconductive transition at 87 and 120 K resulting from microstructure instability. The same phenomenon has also been found by internal friction experiments [H. Shen et al., Proc. of 2nd Internal Friction and Ultrasonic Attenuation Conference of China, C63 (1988)]. An ultrasonic anomaly at 250 K is a relaxation process, similar to the mechanism in YBCuO [M-F. Xu et al., Phy. Rev. B 37, 3675 (1988)].
Vibrotactile threshold for subjects with peripheral neuropathies
Authors: Novick L. H. , Verrillo R. T. , Bolanowski S. J. , Ecker A. , Gescheider G. A.
Published: October 2005

Thresholds for detecting vibrotactile stimuli applied to the thenar eminence through a 2.9-cm2 contactor were measured for patients having reflex sympathetic dystrophy (RSD, n = 7) and carpal tunnel syndrome (CTS, n = 2). Thresholds were measured with a two-interval, forced-choice tracking procedure. The vibratory stimuli consisted of 700-ms sinusoidal bursts with rise-fall times of 37 ms. The frequency of the test stimuli ranged from 3–300 Hz. Three of the RSD patients demonstrated a 20- to 30-dB loss of sensitivity at all frequencies for their affected hands. These patients had normal sensitivity in their unimpaired hands. The other four RSD patients demonstrated no loss in sensitivity in neither their affected nor normal hands. The two patients tested with CTS also had normal threshold functions for both their affected and unaffected hands. The results suggest that the clinical diagnosis of RSD subsumes several pathological conditions one of which may or may not be a decrease in threshold sensitivity. The fact that CTS patients may not experience impaired threshold suggests that this neuropathy may not affect nerve fibers mediating low-threshold tactile sensitivity.
Piano history—Science and art
Authors: Strong William Y.
Published: October 2005

The piano enjoys a unique position in both the concert hall and the home. It has been in existence for well over 200 years and was the subject of much development activity during the middle 1800s. Competition was fierce at the time and innovation was crucial to a company's survival. Artists placed musical demands on the instruments, designers countered with their own ideas and capabilities, and reputations were made on the resulting instrument. Some of the great minds of the day focused on the piano acoustics challenge, perhaps most notably Hermann Helmholtz, a name familiar to all acousticians. This talk will present a few interesting historical details of the liaison between science and art in the design of the piano.
Properties of piano hammers
Authors: Askenfelt Anders
Published: October 2005

Measurements of the compliance and loss factors for raw hammer felt and completed hammers have been recorded. The results of these measurements will be presented and their implications discussed. The potential for new materials for use in piano hammers is suggested and possible candidates for these new materials will be discussed.
Revision of a two-piston eardum model to account for real ear canal geometries
Authors: Stinson R. Michael
Published: October 2005

The acoustical behavior of the middle ear can be conveniently summarized using acoustical network representations, analogous to electrical circuits. Previous middle ear networks, though, have been derived assuming that the ear canal could be treated as uniform in cross section and terminated perpendicularly by the eardrum. At frequencies above about 5 kHz, it is necessary to account for the actual ear canal geometry (varying cross-sectional area along canal length). In this paper, the two-piston representation of Shaw [J. Acoust. Soc. Am. Suppl. 1 62, S12 (1977)] is incorporated into a horn equation description of the sound field in ear canals. Several of Shaw's network parameters must be revised so that eardum impedances and ear canal standing wave ratios will be correctly predicted when this network is used in conjunction with realistic canal geometries. The analysis makes use of recent measurements of human ear canal geometry.
An approach to finite-element modeling of the middle ear
Authors: Funnell Susan M. , Funnell W. Robert J.
Published: October 2005

It is becoming increasingly clear that the mechanical behavior of the middle ear is more complex than first thought. In particular, recent experimental evidence indicates that the ossicular axis of rotation is not fixed. Finite-element analysis is a method well suited to the detailed analysis of complicated mechanical systems. A finite-element model of the eardum has previously been developed. This paper presents work on a system developed to model the middle ear ossicles and important soft tissues. Anatomical information is input to computer by digitizing outlines of structures of interest from serial section slides. The resulting three-dimensional volume is discretized into tetrahedral elements using an automatic mesh generator developed to mesh irregularly shaped objects. Preliminary results of analysis on middle-ear structures will be presented and modeling of the middle ear as a whole will be discussed. [Work supported by MRC Canada.]