bluuebunny/crossref_metadata_2025 huggingface dataset:
Coupling finite elements and Helmholtz integral equations for solving radiation and scattering problems
Authors: Hamonic Bernard F. , Wilson Oscar B. , Baker Steven R.
Published: October 2005

Radiation and scattering problems encountered in fluid-structure interaction can be solved through a total finite-element formulation, or through a mixed formulation. In the first case, both the interacting body and the acoustic medium are modeled using finite elements. This approach leads to a linear equation system of large size when, for example, the overall dimensions of the transducer being analyzed are large compared with the wavelength of the radiated or scattered waves. In the second case, the body is modeled with finite elements and the fluid domain is implicitly represented through an integral equation. This completely eliminates the need for modeling the fluid, drastically reducing the size of the system to be solved. In this paper attention is focused on one combination for the mixed formulation, namely the coupling of the finite-element code ATILA [J. N. Decarpigny et al., J. Acoust. Soc. Am. 78, 1499–1507 (1985)] and the Combined Helmholtz Integral Equation Formulation code CHIEF [H. A. Schenck, J. Acoust. Soc. Am. 44, 41 (1968)]. Results of test cases for both axisymmetrical and 3-D structures are presented. [Work partially supported by DRET.]
The impact of transport category aircraft departure profiles on community noise exposure in the vicinity of John Wayne Airport, Orange County
Authors: Murphy Deborah W.
Published: October 2005

The noise resulting from transport category aircraft operations at John Wayne Airport, Orange County has confronted airport neighbors and airport operators and users since the introduction of turbojet aircraft to the airport in 1968. The unique distribution of population around the airport, coupled with the restricted number of commercial flights per day and the noise abatement flight procedures have led to a compromise which attempts to balance all airport activity with the community's needs for transportation and a livable environment. An analysis of numerous takeoff profiles was completed to determine the noise level sensitivity as a function of takeoff gross weight, aircraft configuration, and power reductions at various altitudes. Potential noise reductions can be obtained from the use of improved thrust and flap management techniques, tailored to the specific noise monitor requirements at John Wayne Airport, Orange County.
Statistical approaches to formant tracking
Authors: Gayvert Robert T. , Hillenbrand James
Published: October 2005

Formant trackers that rely on peak picking tend to make occasional large errors. This paper investigates two general methods for determining formant locations without explicit use of peak information. Both approaches involve statistical models derived from hand-traced formant tracks. In the first method, individual formant frequency values are estimated using a maximum likelihood classifier. In the second, formant probability distributions are found for each element of a vector quantization codebook, and formant values are then determined by conditional mean estimates. Hidden Markov models or simple smoothing can then be applied to provide continuity constraints. Both of these techniques have been quantitatively analyzed using a database of 78 utterances produced by four males and four females. The performance of these trackers across different training and testing sets will be discussed. [Work supported by Rome Air Development Center under contract F3060285-C-0008.]
A pitch extraction method using higher-order joint moment
Authors: Nomura Norio , Yoshida Yuichi
Published: October 2005

A modified higher-order joint moment method for a correct pitch extraction is proposed. The mean products of three vectorized samples are used, while the usual autocorrelation method uses the mean products of two samples. Each of three samples separated by the same sample interval is modified to a vector whose modulus is equal to the absolute value of the sample, and the argument is ± 120 deg in accordance with the sample's sign. Only when three original samples have the same sign does the product of three vectors become a real positive value. By averaging the products in a certain window, a sharp peak on the sample-interval axis is obtained at the pitch period. The effectiveness of the method is examined with real voiced signals. The peak at the pitch period is much sharper than the peak of the autocorrelation method, and the spurious peaks are very small. These properties result in a correct pitch extraction. It is also shown that the influence of the pitch fluctuation has properties similar to those as in the autocorrelation method.
Synthesis of Chinese by rules based on a multipulse excitation model
Authors: Changli Li , Fuyuan Me
Published: October 2005

According to the model of speech production, the characteristic parameter of speech can be divided into two parts: excitation and vocal tract parameters. Atal proposed the multipulse excitation model that can produce high-quality sythesized speech. This research shows that the intensity, duration, and pitch mode of single syllable Chinese produced by multipulse excitation may be changed when the adaptive method is utilized to process its multipulse sequences and vocal tract parameter. There are about 10 000 Chinese words in common use, but the pronunciation of many words is the same, so that only about 1300 syllables are independent. The Chinese language is a tone language. Each Chinese word is of four pitch modes, and the vocal tract parameter for the four modes of one word is almost the same. Therefore, there are 400 independent vocal tract parameters and 1300 multipulse sequences in Chinese. Based on the above strategy, a new method of sythesizing Chinese by rules has been proposed. The intelligibility and naturalness of the synthesized speech are satisfactory.
Quantitative evaluation of the perceptual significance of control parameters in synthesis by rule
Authors: Yamashita Yoichi , Mizoguchi Riichiro , Kakusho Osamu
Published: October 2005

In synthesis by rule, control parameters generated by rules do not always match desired ideal values (natural sounds). To effectively improve synthetic sound quality, it is important to evaluate the perceptual significance of individual parameters and to search for the parameters most important to speech quality. This paper describes a method for measuring relative weights among some control parameters to quantitatively evaluate their perceptual significance. Perceptual weights are measured by equalizing two kinds of distances between stimuli employed in listening tests. One is defined on the physical space delimited by the synthesis parameters of stimuli. Another is defined on the psychological space delimited by parameters obtained from multidimensional scaling (MDS) techniques. MDS distributes stimuli into the space of an arbitrary dimension based on preference lists for stimuli that are obtained through listening tests. To verify the validity of the proposed method, perceptual weights for the first and second formants of the isolated vowel /a/ were measured. [Work partly supported by a Grant-in-Aid for Scientific Research on Priority Areas from the Ministry of Education.]
Acceptability of text-to-speech systems: Its quantification using magnitude estimtion and its relationship to intelligibility and naturalness in various degrees of distortion
Authors: Pavlovic Chaslav , Rossi Mario , Espesser Robert
Published: October 2005

Because there are no physical measurements that quantify perceptual attributes of synthesized speech, subjective tests must be used. In this project the possibility of directly scaling acceptability via magnitude estimation is assessed. One hundred and twenty subjects took part in the study. Seven different synthesizers and three types of background noise were employed. The results are discussed in light of the following questions. (1) What is the relationship between acceptability on one hand and naturalness and intelligibility on the other in various degrees of distortion? (2) Are the objective intelligibility scores and subjective magnitude estimations highly correlated in all conditions? (3) Do the relative ratings produced by different groups of subjects agree? (4) Are the ratings on an absolute scale? (5) Are the ratings invariant to the stimulus set and range size? (6) Are the ratings practice invariant? (7) Do the ratings depend on the subject's familiarity with the test material?
A comparative study on the response of brain monoamines to whole body vibration and local vibration in rats
Authors: Ariizumi Makoto
Published: October 2005

Rats were subjected to whole body vibration in the prone position, and the rats' hind legs were subjected to local vibration for 4 h. The changes in levels of the major brain monoamines, norepinephrine (NE), dopamine (DA), and serotonin (5-HT), were measured in the whole brain and in seven brain regions. Regarding the affected degree of the NE level in the whole brain, NE was decreased significantly to 57% of that of the control level by whole body vibration (20 Hz, 50 m/s2), and was decreased only to 79% of that of the control level by local vibration (120 Hz, 50 m/s2). DA was unaffected either by whole body vibration or by local vibration. 5-HT was remarkably elevated by whole body vibration and slightly elevated by local vibration. These changes in NE and 5-HT were observed commonly in the hypothalamus and the hippocampus. The degree of response of NE and 5-HT was much more severe in whole body vibration exposure than in local vibration exposure.
Very-low-frequency vibration in a motion-base helicopter simulator compared to the simulated platform
Authors: Allgood G. O. , Kennedy R. S. , Berbaum K. S.
Published: October 2005

The physiological symptoms of simulator sickness parallel those of true motion sickness, and are being reported with increasing regularity in Navy flight trainers, particularly helicopter simulators. It was hypothesized that nauseogenic energies in the very-low-frequency (VLF) domain (viz., <1 Hz) would be present in moving base helicopter simulators but not in the actual aircraft that should have energies present at higher frequencies (viz., >2 Hz). A Navy helicopter and its simulator were instrumented to record frequency of acceleration and acceleration profiles with pilots-in-the-loop. Motion sickness was reported in the simulator where recorded energies exceeded limits for VLF vibration directed in Military Standard 1472C (MIL-STD-1472C). No sickness was recorded for pilots flying the helicopter where vibration profiles were at higher frequencies. The presence of VLF vibration in flight simulators and not in aircraft is implied as a cause of simulator sickness. Development of an advanced biomechanical motion analyzer could be employed to signal when on-line devices exceed human exposure limits as well as for test and evaluation. Furthermore, such data could inform future revisions of military standards for dynamic vehicle operations.
Acoustical properties of drill strings: Theory
Authors: Drumheller Douglas S.
Published: October 2005

Drill strings are assembled from 30-ft sections of steel pipe and used to drill wells to recover petrochemical and geothermal resources. The ability to transmit data by acoustic carrier waves generated within the drill string would have significant economic impact on the drilling industry. This poses an unusual waveguide problem. The string, which may be up to 30 000 ft long, forms a periodic structure of slender pipe threaded together with large-diameter tool joints. Classical analysis methods along with numerical algorithms can be used to study the behavior of both periodic and transient acoustic waves in this structure. Stress waves that propagate along this waveguide exhibit all of the phenomena associated with Brillouin scattering theory. The results show a comb filter structure with passbands that produce significant wave dispersion. In addition, the acoustic impedance that determines the design of sending and receiving transducers is a strong function of position as well as frequency. [This work was supported by the U.S. Department of Energy at Sandia National Laboratories under Contract DE-AC04-76DP00789.]
Acoustical properties of drill strings: Application
Authors: Drumheller Douglas S.
Published: October 2005

Analysis has demonstrated a viable data transmission system for communicating by acoustical carrier waves in drill strings used to drill wells. Because of the periodic structure of this waveguide, complex transmission characteristics exist. Two experiments have been used to study these phenomena. In the first experiment, a scale laboratory model was constructed to examine the acoustical response of this system. A longitudinal impulse was produced with a hammer and the resulting signal was measured at several locations with semiconductor strain gauges. In the second experiment, the impulse response of a 1500-ft drill string suspended in a mudfilled well was tested. The data exhibit all of the phenomena predicted by the theory. Measured attenuation is about 2 dB per thousand feet. Both the comb filter phenomena produced by the periodic structure of the pipe and tool joints, and the standing-wave interference pattern caused by echos between opposite ends of the drill string are present. [This work was supported by the U. S. Department of Energy at Sandia National Laboratories under Contract DE-AC04-76DP00789.]
The inner universe: A multidisciplinary approach to the acoustics of concert halls
Authors: Ando Yoichi
Published: October 2005

After introducing dimensional factors of the acoustic system in a concert hall, the capability of calculating acoustical quality at any seat based upon the theory of preference is presented. The effort to describe important qualities of sound in terms of the processes of the auditory pathways and brain has been brought to bear on the problem. If enough were known about how the auditory and the central nervous systems modify the nerve impulses from the cochlea, the design of concert halls could proceed according to guidelines derived from knowledge of these processes. An attempt to analyze this is made through study of the auditory-evoked potentials over the left and right human cerebral hemispheres. In addition, results of recording the slow vertex responses (SVR), which were obtained by adjusting the temporal and spatial physical factors, indicated that the information related to subjective preference appeared in the latency components. The longest-latency responses were observed at the most-preferred condition or at the most subjectively diffuse condition. The correlation between brain activity and subjective preference seems to indicate that subjective preference can be traced back to a primitive response in the “inner universe.” Thus, such a theory with temporal and spatial factors may be generalized in designing physical environments for every human activity.
Performance of a continuous transmission frequency modulation (CTFM) in-air ultrasonic system applied to dynamic production processes
Authors: Collier Robert D. , MacNeil Roderick C. , Kay Leslie
Published: October 2005

The measurement of the thickness of a continuous, moving sheet of deformable material on rollers and conveyor belts is required to control sheet thickness. Laboratory measurements demonstrate accuracies of ±0.005 in. with articulated separate transmitter and receiver transducers at distances of 2–7 in. with a CTFM signal 200-100 kHz. System performance as a function of temperature, alignment, distance, and target reflection characteristics are described. Laboratory dynamic simulation of production processes demonstrate capabilities to measure both step and continuously variable material profiles for conveyor speeds in the range of 30 fpm. Results are compared with in situ production measurements.
Design of an ultrasonic wattmeter
Authors: Kaneko Seiji , Gakumazawa Masahide , Shinoda Kiyoshi
Published: October 2005

A new type of ultrasonic wattmeter, which employs a multiplier IC, has been designed based on the fact that the power is represented by a product of the voltage, the current, and the phase in a high-frequency electric circuit. The output of the IC is averaged in time by a low-pass filter to give a dc signal proportional to the power. The present wattmeter, unlike the conventional ones, has the advantage that the measurements are made over a frequency range from 1 kHz to 1 MHz and also at a high voltage load. This paper describes the outline of the wattmeter and its characteristics.
Phase technique for calibration of large, low-frequency underwater sound projectors
Authors: Hugus G. D. , Carson D. L.
Published: October 2005

The acoustic source level of high-power, low-frequency (10–250 Hz) underwater sound projectors is difficult to measure accurately. This is primarily because of the inaccuracies in measurement of the distances between the effective acoustic centers of the projector and measurement hydrophones. These distances (up to 25 m) cannot be easily determined by direct measurement because acoustic measurements on these inherently large projectors are typically made in open water calibration facilities or at sea. A technique has been developed to measure these large distances more accurately by comparing the phase of two different hydrophone outputs in the farfield. This technique depends on the assumption that the projector acts as a fixed point source and is therefore omnidirectional at these low frequencies, and that free-field conditions exist. [Work supported by Space & Naval Warfare Systems Command.]
Evaluation of measurement accuracy of a sound intensity probe
Authors: Suzuki Hideo
Published: October 2005

The two-microphone sound intensity measurement technique is most commonly used. There are several causes of measurement error with this technique. Some of them are inherent in the technique itself, and others are due to the imperfect performance of the microphone probes. Various types of measurement errors are reviewed first. Then, the measurement accuracy of a one-dimensional, four-microphone probe is discussed. The accuracy of the intensity measurement can be evaluated either in an acoustic tube or in an anechoic chamber. The effective distances between pairs of microphones, active and reactive intensities, and particle velocities measured by use of the probe in an anechoic chamber are presented. These results, including the directivity patterns, are compared with those of a conventional intensity probe. The residual intensity index of the present probe is also presented.
Sound intensity measurement for impulsive sounds
Authors: Tachibana Hideki , Yano Hiroo
Published: October 2005

For the measurement of sound radiation properties caused by structure-borne sound from large structures such as buildings and bridges, impact excitation is much more advantageous and convenient than stationary random excitation. The application of the sound intensity method to such sound radiation measurements was investigated. First, it was theoretically clarified that the result obtained by time integrating the sound intensity impulse response (instantaneous sound intensity) measured by impact excitation corresponds to the sound intensity obtained by stationary random excitation under the conditions of linear vibration propagation and sound radiation. This theoretical relation was examined and confirmed by a basic model experiment in which sound power and sound energy radiation patterns from a steel I beam were measured by both of the exciting methods. As a practical application, the noise radiation properties of an iron railroad bridge were investigated. In this experiment, the rail was impacted by an iron hammer at a constant strength, and the sound intensity around the structure was measured and integrated. From this measurement, the effect of vibration isolation of the rails was examined.
Identification of multiple point sources using acoustic intensity
Authors: Usagawa Tsuyoshi , Miyazono Hiromitsu , Ebata Masanao
Published: October 2005

The identification of noise sources, especially in a small instrument, is important for effective noise control. When the noise radiated from a small instrument is analyzed, acoustic intensity data provide more information than conventional pressure measurement because the acoustic intensity is a vector quantity of the acoustic energy flow. This paper presents a method for the identification of multiple point sources using acoustic intensity, and discusses the performance of the proposed method using a model experiment. The proposed method is based on the assumption that an observed intensity vector is given as a summation of individual intensity vectors radiated from each noise source. This method identifies multiple source positions and estimates the intensity of each source simultaneously. In the model experiment, the proposed method separates three point sources, which are 60 mm apart from each other and radiate 500-Hz sinusoidal waves.
Thermoacoustic generation of narrow-band signals with high repetition rate pulsed lasers
Authors: Berthelot Yves H.
Published: October 2005

The generation of sound by absorption of laser light in water is analyzed for situations where the laser is pulsating at a high repetition rate. It is shown that pulsating the laser at any arbitrary repetition rate is, in general, not a very effective way to produce a narrow-band signal. For narrow-band sound generation, one should instead pulsate the laser at an optimum repetition rate that is determined by the optical frequency of the laser and, to a lesser extent, by the laser beam diameter. Expressions for the optimum repetition rate are derived from both a frequency domain analysis and a time domain analysis. It is found that, with the present laser technology, it is now possible to generate continuous thermoacoustic directional sound waves with high repetition pulsed lasers in such a way that these signals are detectable several kilometers away from the source. Spatially periodic laser deposition configurations on the water surface are also discussed, and further improvement in signal-to-noise ratio can be achieved, in principle, for a spatial periodicity tuned to the optimum temporal periodicity of the repetition rate of the pulsed laser.
Acoustically generated temperature gradients in short plates
Authors: Atchley Anthony A. , Hofler T. J. , Kite M. D.
Published: October 2005

The results of measurements of acoustically generated temperature gradients in short, thin plates located in a resonant tube were reported previously [M. Muzzerall et al., J. Acoust. Soc. Am. Suppl. 1 82, S21 (1987)]. The previous measurements agree well with theory [J. Wheatley et al., J. Acoust. Sue. Am. 74, 153–170 (1983)] as long as the acoustic pressure amplitude is less than or equal to 150 dB re: 20 μPa. Measurements made at a pressure amplitude of 162 dB show serious disagreement with theory in regions of high acoustic velocity. In order to investigate this disagreement further, a more extensive set of measurements was performed, employing various mean pressures, acoustic pressure amplitudes, plate lengths, and plate configurations. The results of these measurements are reported. The data are compared to theory with particular emphasis on velocity-dependent effects. [Work conducted for the Office of Naval Research and funded by the Naval Postgraduate School.]
Hand-arm vibration: New perspectives
Authors: Brubaker Robert L.
Published: October 2005

Signs and symptoms of the hand-arm vibration syndrome (HAVS) will be described followed by recent research developments. Much attention has been focused on elucidating the pathophysiological basis of vibration-induced injury to vascular, neural, muscular, and skeletal tissues in the hand and arm. New information suggests that chronic neural damage is less reversible after long-term cessation of vibration exposure than vascular damage. If confirmed, current hand-arm vibration standards (based mainly on prevention of onset of finger blanching symptoms) should be re-evaluated to also preclude onset of chronic neurologic deficits. Progress has been made in designing and validating new objective measures of peripheral somatosensory dysfunction, as well as improving and testing existing methods for assessing neural and vascular pathology. The Stockholm Workshop symptom scale (revised Taylor-Pelmear scale) for assessing the vascular component of HAVS will be described as well as a suggested new scale for evaluating neural damage.
Noise reduction strategies for the hearing aid user—A critical review
Authors: Buerkli-Halevy Ora
Published: October 2005

As recently confirmed by surveys conducted by the Federal Trade Commission and the Hearing Industry Association, communication in noise is still the number one problem of hearing aid users. This is confirmed by many studies which concluded that SNR enhancement is the single most important determinant of perceived benefit from amplification. In the last few years, development of special noise reduction circuitry has been a priority for many hearing instrument manufacturers. The resulting design can be divided into three major categories: automatic low-frequency suppression, user operated noise reduction systems, and adaptive compression systems. In order to make appropriate fitting choices, it is important to understand how these different systems function and in which environment each is most effective. A critical analysis of each signal processing scheme and its effect on speech discrimination in noise of each system is therefore essential.
Comparison of speech recognition results for a one- and two-formant speech coding strategy for a multichannel cochlear implant
Authors: Brimacombe Judith A. , Beiter Anne L. , Barker Mary J. , Mikami Karen A. , Staller Steven J.
Published: October 2005

This study compared clinical results obtained from 96 postlinguistically deafened adults implanted with the Nucleus 22 channel cochlear implant system. Seventy-one patients were fitted with a device that tracked the second formant of the voice, while 25 used one that tracked the first and second formant frequencies. Experience with the device averaged 3.7 months. Four subtests (four-choice spondee, vowel, NU♯6, and CID sentences) of the Minimal Auditory Capabilities Battery (Auditec of St. Louis recording) were used. In addition, 209 patients were administered live-voice measures (continuous discourse tracking and vowel/consonant identification) in three conditions: lipreading, lipreading plus device, and device only. Results revealed that the group using the two-formant tracker performed significantly better than those using the one-formant tracker on the following measures: (1) NU♯6 monosyllabic word test; (2) CID sentences; (3) continuous discourse tracking difference score (lipreading plus device minus lipreading); and (4) vowel and consonant identification in the device only condition. These findings suggest greater efficacy with the two-formant tracker; this is the strategy commonly in clinical use at the present time.
A speech coding method for an auditory prosthesis by extracochlear stimulation
Authors: Ifukube Tohru , Hirata Yoshihiro , Matsushima Jun'ichi
Published: October 2005

A new model of auditory prosthesis using a newly developed electrode is proposed. This prosthesis is less invasive than conventional ones because it can be placed superficially over the round window membrane of the cochlea rather than being inserted into the scala tympani. The electrode is coated with polyvinyl alcohol gel and is suitable as an extracochlear prosthesis because the electrode can firmly adhere to the round window membrane without damaging it. As a speech coding method, a new idea has been proposed to simultaneously transmit the pitch signal and the second formant frequency through the electrode. In this method, an additional stimulation pulse is inserted between successive pitch pulses, and the time delay between the additional pulse and the preceding pitch pulse is set in proportion to the second formant frequency. Two digital signal processors are used to extract the pitch and the second formant frequencies in real time. These time sequential stimuli are used for deaf patients. This new method of speech coding has proven effective for discrimination of the five Japanese vowels, and also for discrimination among pitch patterns.
An analysis of the characteristics of speech perception ability by the combined use of cochlear implant and lipreading
Authors: Fukuda Yumiko
Published: October 2005

In this study, test materials for evaluating speech perception ability by a combined use of cochlear implants and lipreading were first selected, taking into account various acoustical parameters such as fundamental frequency, intensity, and formant frequencies that can be transmitted through a speech signal processor in a cochlear implant, as well as supplemental speech information that can be obtained through mouth shape in lipreading. The test materials included the short versus long vowels, unvoiced versus voiced consonants. Japanese word accent and sentence intonation, and the 5 Japanese vowels, 50 monosyllables, 50 words, and 29 sentences that consisted of 100 words. Then, both the speech signal and the mouth shape for utterances of the test materials by a female speaker were recorded on videotapes and were presented to the subjects who had cochlear implants in Tokyo. The results showed that the test materials were useful for a detailed analysis of the characteristics of speech perception ability through the cochlear implants and lipreading.
Artificial speech production device for laryngectomees controlled by jaw and tongue movements
Authors: Matsumura Masafumi , Goi Yoshiyuki , Fujii Katsuhiko
Published: October 2005

The purpose of this study is to develop an artificial speech production device for laryngectomees. The device is composed of three parts, a measurement part for jaw and tongue movements, an estimation part for the formant frequencies, and a voice synthesizer. This paper argues that the formant frequencies of vowels can be estimated from tongue and jaw movements of the subject, and that based on this information, continuous vowels can be produced using a voice synthesizer. Curvature of midsagittal tongue surface is measured with piezoelectric film. The degree of the jaw opening is estimated by the electrical impedence between right and left buccalis. Experimental results suggest that the formant frequencies can be determined from the curvature of the tongue surface and the jaw opening. Therefore, a mapping function is introduced to obtain the formant frequencies for the synthetic speech. The amplitude and pitch of the voice are controlled manually by the person who wears the device. The articulation score of the voice produced by the device was 95% for isolated vowels and 90% for connected vowels. Results showed the usefulness of the proposed device.
Amplitude envelopes and speechreading
Authors: Grant Ken W. , Renn Rebecca J. , Yu Jenny S.
Published: October 2005

Amplitude envelopes derived from filtered bands of speech provide varying degrees of benefit to speechreading. Three parameters related to envelope extraction are examined using both easy and difficult sentence materials: (1) the bandwidth and center frequency of the filtered speech signal used to obtain the envelope, (2) the bandwidth of the envelope signal, and (3) the carrier signal used to convey the envelope. Thus far, results for normal hearing subjects presented with difficult speech materials suggest that: (1) the envelope of wideband speech did not always provide the greatest benefit to speechreading when compared to envelopes derived from octave bands of speech; (2) as the bandwidth of the envelope signal increased from 100–4000 Hz, auditory-visual (AV) performance improved slightly, whereas bandwidths below 100 Hz resulted in AV scores that were equal to or worse than speechreading alone; and (3) low-frequency carrier signals were better than high-frequency or wideband carrier signals, at least for envelopes derived from an octave band of speech centered at 500 Hz. Similar tests with easy speech materials are in progress. [Work supported by NIH.]
An application of the articulation index to hearing aid fitting
Authors: Rankovic Christine M. , Van Tasell Dianne J.
Published: October 2005

It is often assumed that the articulation index (AI) is a sensitive and appropriate metric for making comparisons among frequency-gain characteristics of hearing aids, and that the characteristic yielding the highest AI will result in the best speech intelligibility score. In evaluating these assumptions, 12 hearing-impaired subjects listened to nonsense syllables under conditions of amplification prescribed by the NAL [D. Byrne and H. Dillon, Ear Hear. 7, 257–265 (1986)] and POGO [G. A. McCandless and P. E. Lyregaard, Hear. Instrum. 34, 16–21 (1983)] hearing aid fitting formulas. These two prescriptive methods recommend similar frequency responses, although the POGO consistently provides more overall gain. In a third condition, the goal of amplification was to insure that the speech was fully audible (AI = 1.0). Results suggest that the articulation index is useful for predicting percent-correct intelligibility scores for some, but not all, of the hearing-impaired subjects. Interestingly, the AI = 1.0 condition did not necessarily yield the best percent-correct intelligibility score. [Work supported by NICHHD T32 HD-07151, NINCDS NS 12125, and the Bryng Bryngelson Communication Disorders Research Fund at the University of Minnesota.]
Time-varying/frequency-varying amplitude compression in compensating for recruitment of loudness
Authors: Rutledge Janet C. , Clements Mark A.
Published: October 2005

Research in the area of compensating for sensorineural hearing losses, including recruitment of loudness, has indicated that amplitude compression may be a solution. Previous systems developed have used filtering techniques that operate directly on the speech waveform and have parameters that remain constant with time. A system is proposed utilizing the sinusoidal speech model [J. R. Lies and M. A. Clements, J. Acoust. Soc. Am. Suppl. 1 82, S5 (1987)], which allows amplitude compression to vary with both time and frequency. The system uses an iterative procedure to determine the best compression coefficient for each sinusoidal component in each 10-ms frame of speech based on a model of the impaired person's masking profile. This allows the processing to adapt to the changing properties of the speech signal in addition to the frequency characteristics of the person's residual hearing. Since the operations are performed on the sinusoidal components rather than on the speech waveform, the distortions inherent in the multichannel filtering techniques have been avoided.
Videophonetics
Authors: Levitt Harry
Published: October 2005

Techniques developed for the acoustic analysis of speech have been applied in an analogous manner to video speech signals. This new field of investigation is referred to as videophonetics. New techniques that have been developed include video speech synthesis by means of cartoons or by concatenation of prerecorded diphones and triphones, analysis of phonetic contrasts in lipreading by interleaving frames of paired videorecordings, spatial high-pass and low-pass filtering of lipreading signals using bivariate Fourier transforms, and automating (in order to improve objectivity) the method of continuous discourse tracking for evaluating communication ability using lipreading and/or a sensory aid. [Research by NINCDS.]
Effects of vibrotactile aid usage on vocal development during the first six months of life
Authors: Proctor Adele , Andreassi Joanne , Wu Hsien-Lan
Published: October 2005

A profoundly deaf female infant was identified at age 5 weeks old and provided with a single-channel, vibrotactile aid at age 6 weeks. Video and audio tapes recorded in the home were analyzed for phonetic and acoustic changes in vocal development and compared to vocal changes in a normal-hearing, female infant recorded under similar conditions at the same ages. The purpose of this poster session is to demonstrate phonetic and acoustic similarities and differences between noncry vocal behavior of the deaf and normal-hearing infant during the first 6 months of life. Results of phonetic transcription and measurements of fundamental frequency [F0], duration, and amplitude will be discussed relative to tactile aid usage and Waterson's [1988] theory of prosodic development in the acquisition of speech production. [Work supported by DOE, Grant No. 133 GH70189 and Department of Health and Human Services, Grant No. RR07143.]
Active vibration control of flexible structures by means of a quasimodal control method
Authors: Nonami Kenzo
Published: October 2005

This paper proposes a new quasimodal control method that can actually be realized by combining the advantages of a modal control and of an output feedback. This method consists of displacement and velocity feedback. A quasimodal control method with only velocity feedback before is proposed. But some poles were still close to the imaginary axis. This method with displacement and velocity feedback is an improvement over that method only with velocity feedback. These feedback gains are also determined based on a modal analysis. The optimal gains of the new quasimodal control method coincide with the optimal gains of an output feedback method. The quasimodal control method becomes effective for cases where the number of controlled modes is more than the number of sensors and actuators.
Active vibration control for a precision pointing system on the Space Station
Authors: Sirlin Samuel W.
Published: October 2005

Disturbance attenuation is receiving increasing attention for spacecraft as it becomes evident that precise pointing is required in disturbance-rich environments. Future observational instruments have extremely tight pointing and stability requirements (for example, 50 nanoradian stability for a proposed planet seeking Circumstellar Imaging Telescope). At the same time, these instruments may be mounted on board the Space Station, together with various machinery and people that continuously generate noise which excites the flexible station structure and tends to disturb pointing payloads. The softmounted inertially reacting pointing system (SIRPNT) combines a precise inertial pointing system, as would be used on a free-flying spacecraft, with an actively controlled (piezoelectric polymer based) softmount capable of keeping the payload tied to the base-body but suppressing the vibration that would be ordinarily transmitted across the mount. Previously presented proof-of-concept analysis has been extended to predict the system performance in a realistic environment. Optimization of the softmount control system is done to minimize the transmitted vibration subject to sensor and actuator limitations.
Pulse modeling in ocean acoustics: Brute-force Fourier synthesis versus time-domain techniques
Authors: Jensen Finn B.
Published: October 2005

There are fundamentally two different approaches to the pulse modeling problem. The first is to generate pulse results by Fourier synthesis of time-harmonic solutions. This approach requires little programming effort and builds on the existing set of cw codes (normal modes, FFP, PE), which have been developed and continuously improved over the past decade or two. The second approach is to solve the problem directly in the time domain, which requires the development of an entirely new set of propagation codes. A similar effort would only pay off if there are significant computational advantages to solving the problem in the time domain. The issue of computational efficiency is addressed by solving characteristic short- and long-range propagation problems in shallow and deep water, employing both Fourier-synthesized cw codes (FFP and PE) and time-domain codes recently developed by M. B. Porter (TDFFP) and M.D. Collins (TDPE).
Materials performance evaluation for hand-arm vibration isolation
Authors: Hampel Gary A.
Published: October 2005

Materials performance affecting the level of surface vibration ordinarily in direct contact with the hand under specified conditions will be reported. The intent of this study was to develop a method by which elastic materials can be evaluated for isolation properties relating to test surface vibration using nonrigid loading. The hand and arm, as a complex system, are used in place of the more traditional and simpler rigid mass load. This research and the scope of these findings report performance as a more realistic means for determining isolation, i.e., vibration reduction caused by an intervening material. The findings reported use this method to identify materials that exhibit isolation and display this positive effect, if it occurs, by graphic means over the frequency range of 1 to 800 Hz. The Z-axis (basicentric) hand-arm apparent mass, evaluated while grasping a 3.81-cm-diam handle with 4.5-Newton total grip force accelerated at 4 m/s2, will be used as the reference load to determine material isolation effectiveness, reported as transmissibility.
Time-domain elastodynamic scattering problems
Authors: Achenbach J. D. , Hituse S. , Zhang Ch.
Published: October 2005

The interaction of an ultrasonic pulse with an inhomogeneity in an elastic solid is generally analyzed by the use of the FFT over time in combination with a numerical method for the scattering problem in the frequency domain. For pulses with a high-frequency content this approach tends to be computationally intensive. In this talk the possibilities of a direct analytical time-domain approach are explored, and then there is a discussion of numerical techniques. Results are presented for a single crack (slit and penny-shaped) and for macrocrack-microcrack configurations. Some of these results have been obtained by the finite difference method. The main emphasis in this paper is, however, on the development of a time-domain boundary integral equation method. By the use of appropriate representation integrals, a system of boundary integral equations has been obtained, which has subsequently been cast in a form that is amenable to a solution by the boundary element method in conjunction with a time-stepping technique. Particular attention has been devoted to dynamic overshoots of the stress intensity factors. Elastodynamic stress intensity factors have been computed as functions of time, and they have been compared with results of other authors.
Acoustic radiation from an impulsive point source in a continuously layered fluid—An analysis based on the Cagniard method
Authors: de Hoop Adrianus T.
Published: October 2005

Acoustic radiation from an impulsive point source in a continuously layered fluid with depth-varying parameters is investigated theoretically with the aid of the modified Cagniard method, that starts with a one-sided Laplace transformation with respect to time and a Fourier transformation with respect to the horizontal space coordinates. Using appropriate one-sided Green's functions, the system of transform-domain differential equations in the depth coordinate is rewritten as a system of integral equations that, for not too rapidly varying fluid properties, can be solved by iteration. The modified Cagniard method leads to space-time expressions for the relevant iterates. To show the generality of the method, the fluid is assumed to show anisotropy in its volume density of mass. The continuously refracted waves emitted by the source and the singly, continuously, reflected wave in an isotropic fluid are discussed in detail. With this method, no difficulties arise with “turning rays” as is the case in the frequency-domain analysis of the problem. [Work done as a Visiting Scientist at Schlumberger-Doll Research, Old Quarry Road, Ridgefield, CT 06877-4108.]
Reduction of bit-error rates by adaptive equalization for a 500-kbit/s underwater acoustic communication system
Authors: Yauchi Shinji , Kaya Akio
Published: October 2005

For the video transmission from untethered subsea robots, an acoustic communication system using the 16-ary quadrature amplitude modulation (16QAM) method is being developed to perform at a capacity of 500 kbit/s at a maximum transmission range of 60 m. To improve the quality of the transmitted information, an adaptive equalization technique will be introduced to this system. In this study, an adaptive equalizer based on a baseband decision-directed equalization method is realized as a computer program. The optimal parameters of the equalizer such as the number of taps and step sizes are determined by computer simulations. In order to estimate the performance of the acoustic communication system with an adaptive equalizer, random code data modulated by 16QAM are transmitted to a receiver located 10–60 m from the source through a surface channel, and stored in a digital memory bank for the off-line processing of an adaptive equalizer. Results show that the bit-error rates in the output of the receiver are about 10−4 at a range of 60 m because of the channel distortion caused by multipath interference, the attenuation characteristics of the channel, etc. However, the bit-error rates in the output of the adaptive equalizer are reduced to less than 10−7 in the 10- to-60-m range. [Conducted as part of the R&D Program of the Large-Scale Project “Advanced Robot Technology” by the Agency of Industrial Science and Technology, Ministry of International Trade and Industry.]
Wave field factorization for broadband acoustic signals
Authors: McCoy John J.
Published: October 2005

A framework has been demonstrated for factoring the Helmholtz equation, governing for narrow-band acoustic signals, into a pair of one-way equations [L. Fishman and J. J. McCoy, J. Math. Phys. 25, 285–296 (1984)]. This factoring requires a range-independent propagation environment. The factored Helmholtz equation has been demonstrated to provide a marching algorithm [L. Fishman and J. J. McCoy, Geophys. J. R. Astron. Soc. 80, 439–461 (1985)], which is a generalization of the split-step algorithm, the latter being frequently used for marching the solution of the ordinary parabolic wave equation. The extension of these results to broadband acoustic signals is considered. Two factorizations are considered and shown to apply to different initial/boundary value problems. The first factorization is based on a Fourier synthesis of that obtained for narrow-band signals. This applies for a forcing problem as a broadband time series acting across a source range plane. The second factorization obtains for the time coordinate of the wave equation. This applies for an infinite spatial domain problem for specified initial time data.
The finite difference method for time domain solutions to range-dependent bottom-interaction problems
Authors: Stephen Ralph A.
Published: October 2005

An explicit second-order finite difference scheme has been used to solve the elastic wave equation in the time domain. Solutions are presented for the perfect wedge, the lossless penetrable wedge, and the plane parallel waveguide, which have been proposed as benchmarks by the Acoustical Society of America. Good agreement with reference solutions is obtained if the media is discretized at 20 grid points per wavelength. The principle disadvantage of the technique is long computational times that are between 10 and 20 h on a minicomputer without an array processor. The method has the advantage of providing phase information and, when run for a pulse source, of providing insight into the evolution of the wave field and energy partitioning. Arbitrarily more complex models including velocity gradients, strong lateral heterogeneities, and random media can be solved with no additional computational effort. The method has also been formulated to include shear wave effects. [Work supported by ONR.]