bluuebunny/crossref_metadata_2025 huggingface dataset:
Speech analysis using a time-varying ARX model for separating the source-tract coupling of vowels
Authors: Funada Tetsuo
Published: October 2005

The purpose of this research is to extract formant frequencies precisely and to classify voiced/unvoiced intervals accurately based on a source-tract model. A sequential estimation of the source wave (i.e., the glottal volume flow) and the vocal tract (VT) characteristics is achieved by using a time-varying “ARX model,” where the term ARX model refers to an AR (autoregressive) model with an auxiliary nonwhite input (X input). This X input indicates the glottal volume flow in the present research. Applications to synthetic vowels generated by the two-mass model demonstrated the following results: (1) Much information on the glottal closure and opening was obtained from the X input; and (2) compared to the conventional (autocorrelation) LP method, formant frequencies (especially the first formant) during the open period of the glottis were estimated more accurately. It has also been observed from real vowels uttered by a male speaker that the phase of the X input agrees with the phase of the glottal movement which can be confirmed by electroglottography (EGG).
Excitation problem in speech synthesis
Authors: Atal Bishnu S.
Published: October 2005

Linear predictive coding methods require efficient representation of both the LPC filter and its input excitation to synthesize high-quality speech at low bit rates. Considerable progress has been made so far in encoding the filter parameters and it is possible to quantize these parameters with only 1600 bits/s without introducing distortion in the synthetic speech signal. However, it is still not possible to encode the LPC filter excitation at low bit rates and maintain high voice quality in the synthetic speech signal. In this paper, the problems associated with low bit representation of the excitation are discussed. To achieve low bit rates, a parametric representation is needed that can provide a compact yet accurate representation of the excitation. Such a compact representation is obtained by expressing the excitation waveform as a linear combination of the eigenvectors of the autocorrelation matrix of the LPC filter's impulse response. This representation allows the study of the effect of changes in the filter excitation on the speech output in a systematic manner. The signal-to-noise ratios necessary to represent various eigenvector components in the excitation without producing perceptible distortion in the output speech signal have been determined. Thus the minimum number of bits necessary to reproduce a speech signal is estimated. These results will be discussed in the paper.
Statistical modeling of dynamic spectral patterns for a speech synthesizer
Authors: Takahashi Sateshi , Satoh Yasuaki , Ohno Takeshi , Shirai Katsuhiko
Published: October 2005

A new method called the spectral locus control method (SLCM) is proposed, which can approximate the dynamic characteristics of the speech spectrum, such as in the transition from vowels to consonants or from consonants to vowels, effectively and accurately. The main procedures of the method are as follows. Continuous speech is segmented into VCV units, and these units are grouped according to the consonants. The spectrum patterns of the V1CV2 units in each group are analyzed to construct a statistical model which, given the spectra of V1 and V2, generates the spectrum loci for V1CV2 units. To synthesize continuous speech, a spectrum appropriate for a given consonantal context is first selected for each vowel V in every CVC sequence in the text. Then, the temporal sequence of the spectrum patterns for the entire V1CV2 is calculated based on the spectrum of the stationary parts in V1 and V2. Since VCV segment spectra are adapted to their consonantal environment, the synthesized speech is highly natural, especially in transitions.
A speech synthesis system by rule in Japanese
Authors: Teranishi Ryunen
Published: October 2005

In Japanese, text-to-speech systems have to deal with problems of complicated orthography and a writing custom without a clear word separation rule. The system shown here is a tentative one avoiding such troublesome problems, constructed to study the rules in Japanese that are useful for the speech synthesis. In order to obtain natural prosody, as in human text reading, the system should have some mechanism that divides the input sentence into several proper breath groups with pauses based on the analysis of the syntactic structure as a human does. The construction of the system has been accomplished, and it can respond to such a demand. This algorithm was realized as completed software on 2HD diskette, available for the PC-98 series of personal computer made by NEC. The features of the system are as follows. The input form is word units written in kana letter codes. These units are separated with the space code. No prosody code is necessary except for the punctuation marks. An original parser, based on traditional Japanese grammar, produces the prosody for the synthesized speech.
The Japanese speech synthesis system with text editing and automatic prosodic control facilities
Authors: Yamamoto Seiichi , Higuchi Norio , Shimizu Tohru
Published: October 2005

A Japanese speech synthesis system using a Japanese speech synthesizer by rule and software for text editing and automatic prosodic control has been developed. In the case of Japanese, there is a possibility that difference in accent type will cause differences in meaning, and that a word included in a compound word, such as a compound noun or verb, will have an accent type different from its original one. Therefore, the correct specification of the prosodic symbols is not so easy, even for people without any difficulty in hearing. The software for the text editing and for the automatic prosodic control inserts symbols for the prosodic control into the input text during the process of the text creation and editing in which the input kana string is converted to a kanji-kana mixed sentence as in conventional Japanese word processors. On the other hand, the speech synthesizer uses phonemes as synthesis units and generates all acoustic parameters based on the 156 feature rules and 472 parameter rules. Since it also has the facility to send the synthetic speech through the telephone line, even a speech-impaired person can transmit messages to a distant listener.
A Japanese speech synthesizer based on production rules
Authors: Higuchi Norio , Yamamoto Seiichi , Shimizu Tohru
Published: October 2005

A Japanese speech synthesizer by rule, which uses phonemes as synthesis units and generates all acoustic parameters based on production rules, has been developed. The conversion from the input romaji string in Hepburn style to the synthetic speech waveform consists of (1) the generation of the phoneme/boundary string with the distinctive feature matrix based on 156 feature rules, (2) the conversion to sequences of the acoustic parameters based on 472 parameter rules, and (3) the generation of the speech waveform using a Klatt-type formant synthesizer. The first two processes are written in C language and implemented by a microprocessor (M 68000) and the last one is implemented by a digital signal processor (TI TMS32010). Both male and female voices can be synthesized with three different accent levels at seven different speech rates in real time. Nine kinds of subjective evaluation, which include tests for intelligibility, naturalness, and other nonlinguistic factors, were proposed and applied to the speech generated with the above-mentioned speech synthesizer. According to the results, 87.8% of the morae of the male voice and 81.6% of the morae of the female voice were identified correctly by three male subjects and two female subjects.
Rhythm control based on CV-syllable positioning for Japanese synthetic speech
Authors: Minowa Toshimitsu
Published: October 2005

A technique has been developed for Japanese speech synthesis-by-rule to control the rhythm of synthetic speech sounds to which little attention has been given so far. In Japanese speech sounds, syllables are generally believed to be the basic elements of the rhythm, with each syllable sound pronounced almost isochronously. It was found through listening tests that there is an important portion in a syllable for recognizing the syllable and the positioning of that portion determines the rhythm. The portion was termed auditory perceptual timing point (APTP) and was determined for each syllable in listening tests. Most APTPs were found near the voice onset, which closely agreed with the result obtained by Sato [H. Sato, Trans. Comm. Speech Res., ASJ, S77-31, 1–8 (1977)]. The rhythm pattern was, in principle, determined by the number of morae in individual words and the syntactic structure of an input text, though further investigation is necessary to construct detailed rules. It has been confirmed that the quality of synthetic speech sound can be improved by employing this rhythm-control technique.
On the unit selection measure of speech synthesis by rule using multiple synthesis units
Authors: Abe Katsuo , Sagisaka Yoshinori
Published: October 2005

In speech synthesis by rule, a synthesis scheme using nonuniform speech synthesis units to obtain the optimum synthesis unit sequence for a desired output speech has been proposed. In this paper, vowel spectrum variations are analyzed using the LPC-cepstrum distance to introduce a quantitative measure for unit selection. Using 5240 words, vowel spectral distortion resulting from contextual differences was studied and the following tendencies were found. (1) The neighboring phoneme, the position in the utterance, and the accentuation affect vowel spectral envelopes in this order. (2) For CVs whose following consonants have the same point or manner of articulation, the spectral distance among the vowels of the CVs is 12% smaller than the average. (3) The vowel spectrum of the word's final CV differs from that of the word's medial CV. Based on these results, a quantitative measure is introduced to represent the spectral similarities of each vowel. With this measure, the unit selection scheme was tested using open data. Through these experiments, it was not only confirmed that the previously proposed categorical measures are adequate for general unit selection, but it was also shown that some phoneme combinations should be specially scored for unit selection.
Quality assessment of synthetic speech using word intelligibility scores
Authors: Watanabe Toshiro
Published: October 2005

The quality of Japanese speech synthesized by rules cannot be sufficiently evaluated using only the 100-monosyllable list that has been used for speech articulation tests: Word intelligibility must also be assessed. Word intelligibility tests on both natural and synthetic speech were conducted to clarify the differences between synthetic and natural speech in learning effects, the effects of external noise, and word familiarity. Both trained and untrained listeners were used. Each word list of 155 words, selected by our previously developed method, reflected the most important word attributes of Japanese: word length, word familiarity, and initial phoneme occurrence. The experimental results show that the intelligibility scores for synthetic speech with untrained subjects were fairly low and independent of noise level. These were greatly improved by training, depending on the noise level. However, natural speech intelligibility depends little on training. The results also show that the familiarity of the test words and the first phoneme in the word are important factors in recognizing Japanese words in both synthetic and natural speech.
Morphophonological derivation of Japanese predicate phrases from a semantic base
Authors: Sato Shigeru
Published: October 2005

The definition of lexical base forms may greatly affect the structure of the phonological component of a system of speech synthesis from semantic representations. This paper presents a model of mora-generating/preserving phonology for Japanese constructed from an observation of the syntactic/morphophonological derivations of predicate phrases. The achievements of the model are the following. (1) The output of the mora-independent lexico-syntax of the predicate phrase is taken care of by the three-tier rule system of word formation, phonology, and phonetics. (2) The reduction of the number of rules and the simplification of their forms are attained by transferring to syntax processes previously regarded as phonological. (3) The mora preservation phenomena are exclusively handled by the seven cyclic phonological rules. (4) Along with the rule editor system, this phonological component is fully implemented in the computer as a subsystem attached to the speech synthesis system from the semantic base [S. Sato and H. Kasuya, European Conference of Speech Technology, Vol. 2, 414–417 (1987)].
Adverse effects of low-frequency ship motion: Problems of defining physiological criteria for crew protection
Authors: Thomas D. J. , Guignard J. C. , Willems G. C.
Published: October 2005

Large surface effect ship (SES) motion simulations and supporting studies using low-frequency (0.17–0.5 Hz) harmonic “heave” (vertical) motion indicated an urgent need to continue research to elucidate mechanisms of motion sickness and to develop predictive models of motion sickness incidence suitable for incorporation into human vibration exposure standards. Nineteen young male naval volunteers underwent severe SES motion simulations and a larger cohort of civilian volunteers underwent compound harmonic motion in a large-amplitude motion generator. Occurrence of and time to emesis were recorded, as well as head inertial and related biomedical responses, during the SES simulations. [The harmonic motion experiments have been reported previously by J. C. Guignard and M. E. McCauley, Aviat. Space Environ. Med. 53, 554–563 (1982)]. Severe kinetosis, usually leading to abandonment of tasks, was observed in 16 of 19 subjects exposed to the simulated operational environment; and marked postrun illness and postural instability were frequent. Habituation to severe motion was not observed. Some correlation emerged between inertial head motion and susceptibility to sickness. Time to emesis varied idiosyncratically and is of uncertain value as a criterion of exposure severity. Further systematic research is recommended to establish the significant parameters of the physical stimulus to seasickness. [Work performed while the authors were at the NAMRL Detachment, now the Naval Biodynamics Laboratory, New Orleans, LA.]
Human physiological response to low sound-pressure level infra- and low-frequency noise
Authors: Tokita Yasuo
Published: October 2005

During the last 20 years or so, there have been many complaints about infra- and low-frequency noise as an environmental problem in the neighborhood of factories, long road bridges, civil engineering blast sites, high-speed train tunnels, etc. The correlation between the physiological response in laboratory experiment results and field surveys on complaints is important in evaluating the infra- and low-frequency noise. This study has been with pure tone and band noise exposure (frequency: 2–500 Hz; level: 50–110 dB SPL) on the following items: (1) respiration and pulse, (2) blood pressure, (3) evoked potentials of brain waves, (4) stress hormones, and (5) sleep interference. According to these results, the quantitative relation between the human responses and sound-pressure level and frequency are not clear but, among these items, sleep interference has a rather better correlation with the sound-pressure level and frequency than the other items, and corresponds with the results of psychological experiments. [Work supported by Environment Agency.]
Experimental evaluation of dynamic fatigue in glass reinforced composite flextensional transducer shells
Authors: Benjamin Kim , Sturges James
Published: October 2005

Results from a controlled experiment aimed at determining the fatigue life, critical flaw size, and critical flaw location for glass reinforced composite flextensional shells are presented. Four shells, two E-Glass and two S-Glass were built into active projectors and cycled in air for 109 cycles. Intentional interlaminar and surface type flaws were symmetrically located on a shell of each composition. Static and dynamic stress levels simulated the anticipated conditions in water at full dynamic displacement and maximum operational depth. The fatigue failure criterion was defined as a 15% change in shell stiffness. The stiffness in the shells was computed from the measured transducer admittance and strain gage readings. Other sensory data included those from accelerometers, thermistors, and pre- and postcycling color photographs of the shell. The results indicate that all four shells survived one billion cycles without any significant performance degradation. In addition to these fatigue data, creep measurements obtained on the shells and ceramic stacks are also presented.
Evaluation of the complex coefficients ŝ33H, d̂33, and μ̂33T of length expander magnetostrictive rods
Authors: Ricketts Donald , Benjamin Kim
Published: October 2005

An analytical model is presented for evaluating the effective complex material coefficients of length expander piezomagnetic rods with magnetic field in the direction of wave propagation. In particular, the model facilitates the evaluation of complex ŝ33H, k̂33, μ̂33, and d̂33 from the measured set of quantities [fRI,QMI, fRE,QME,Zex, fex], where the superscripts I and E denote the ac open-circuit and short-circuit conditions, respectively. Thus fRI and QMI are the measured ac open-circuit resonance frequency and mechanical storage factor; likewise for fRE and QME. Here, Zex is the measured value of the electrical impedance at the frequency fex. An impact testing technique [J. Acoust. Soc. Am. Suppl. 1 67, S32 (1980)] is used to measure the resonance frequencies and mechanical storage factors of the free-free magnetostrictive rod test specimen for the two electric boundary conditions. Experimental results are presented for ŝ33H, d̂33, and μ̂33T of grain-oriented Terfenol-D rods for several levels of magnetic bias.
Reciprocity calibration of a Tonpilz transducer by the Delta-Z method
Authors: Patton Mark D. , Baker Steven R. , Wilson Oscar B.
Published: October 2005

The Delta-Z reciprocity calibration method [R. Bedard and S. R Baker, J. Acoust. Soc. Am. Suppl. 1 83, S19 (1988)] has been applied to a Tonpilz transducer. The diffraction constant and the free-field radiation impedance in water were calculated using the U.S. Navy's CHIEF computer code [H. A. Schenck, J. Acoust. Soc. Am. 44, 41 (1968)]. Measurements of the input electrical impedance in water and in air, and a standard comparison calibration were performed at the Transducer Evaluation Center (TRANSDEC), Naval Ocean Systems Center, San Diego, CA. The results of the Delta-Z method and the standard comparison calibration method will be compared. [Work sponsored by NRL-USRD.]
Measurement of flow resistance of porous materials at high flow rates
Authors: Galaitsis Anthony G. , Kiss Mark
Published: October 2005

A flow resistance measurement facility was developed to quantify the flow characteristics of porous sheets under pressure differences of up to 100 psi. These materials have been utilized in a particular class of silencers developed to reduce the noise of impulsive, high-pressure air discharges. The silencers consist of a porous-wall enclosure of volume V, they yield an insertion loss IL, and develop a backpressure ΔP. The porous material properties near the operating pressure must be known accurately to minimize (through analytical design) the silencer volume while meeting insertion loss and backpressure requirements. Prior to developing this flow facility, the analytical design was performed by extrapolating the material properties measured by manufacturers at low pressures (about 0.08 psi), which lead to uncertainties and, often, required design iterations. The usage of the flow facility allows accurate determination of material properties near the operating point and improves the efficiency of the design process. The presentation will describe the flow facility and recent test results, and will review the silencer design procedure.
A study on a detection method for chemically solidified soil using impulsive sound
Authors: Watanabe Tsutomu , Takahashi Takuya , Motooka Seiichi
Published: October 2005

In engineering foundation works, the chemical grouting method is useful to increase the bearing capacity of the soil. If the shapes and the location of the underground solidification caused by chemical grouting can be detected without digging up the ground, it will result in greater safety in engineering work operations. The authors devised a method to detect the location and the shapes of solidification by applying a pulse-echo method using impulsive sound. In this method, a hole (diam, 100 mm; depth 3 m) is dug vertically about 1 m from the location of the chemical grouting. The hole is filled with water and an electromagnetic induction-type sound source is suspended in it. The echo signals from the chemical solidification are received by four hydrophones suspended symmetrically with respect to the sound source. These outputs are processed by a technique similar to polarity correlation. Then, two-dimensional image patterns of the solidification can be obtained. Experiments in a model sand bath (300×300×300 cm) show satisfactory results.
Design of an automated, computer-controlled acoustic tank facility and its application to measurement of nonlinear effects
Authors: Schwaiger Gary P. , Ginsberg Jerry H. , Rogers Peter H.
Published: October 2005

A computer-actuated acoustic research tank facility designed for facilitating precise acoustic measurement of sonar transducers, small arrays, and structural models has been installed at the Georgia Institute of Technology. The apparatus consists of a wooden tank, having dimensions of 7-m diameter and 5-m depth, integrated to a computer actuated positioning and data retrieval system. A resident minicomputer controls positioning and signal generation, as well as providing temporary data storage and signal processing capabilities. Permanent data storage and post-processing are provided by a remote minicomputer. The multiaxial positioning and scanning system, which extends over the entire active tank area, provides accurate triaxial and rotational positioning of the transducer. Component hardware, positioning system design features, computer software and facility capabilities are presented. One current use for the facility is to perform a series of experimental measurements exploring nonlinear effects encountered in sound beams generated by a high-power transducer operating at a moderately high ka value. Waveforms are measured at a variety of locations in the near and farfields, both on and off axis. Signal processing then provides amplitude and phase information for the fundamental and higher harmonics.
Development and test of piezoelectric vibration calibrators to be used for frequency range above 10 kHz
Authors: Ge Li-Feng
Published: October 2005

Piezoelectric calibrators intended as vibration standards at frequencies higher than 10 kHz, based on the reciprocity method, have been developed and tested. Each calibrator is composed of a table, a two-disk active element, and a base. The first resonance frequency of the shakers is higher than 30 kHz; and the capacitance of the electrical port is about 10–15 nF. The design possesses many advantages, such as: simple structure, stable electrical and mechanical characteristics, low distortion, and good amplitude linearity. In particular, the electrical impedance changes resulting from mass load changes are sufficiently large and repeatable in the higher frequency range, 10–40 kHz, to permit accurate calibration by the piezoelectric reciprocity method.
A study on the standard for sound intensity measurement
Authors: Fujimori Takeshi , Sato Sojun , Miura Hajime
Published: October 2005

A study on a traceability system for sound intensity standards required for accurate sound intensity measurements is described in detail. Required characteristics for the system, such as a calibration method for sound intensity meters involving a field calibrator and a working standard, are discussed. A free-field calibration method using a progressive wave or a standing wave in an anechoic room is proposed. This is valid for the estimation of accuracies of intensity measurements in complicated actual sound fields. Some ideas on the generation of a calibrating sound field and a standard sound intensity probe are given. The standard probe is composed of several pressure microphones mounted on a rigid sphere and is precisely calibrated using a laboratory standard condenser microphone. To obtain enough accuracy in intensity measurements using the system, five sound intensity meters on the market were tested in various sound fields. The results show that the proposed traceability system is valid for sound intensity measurements.
The use of intensity techniques for noise source identification in small machinery
Authors: Dietschi U. D. , Bolton J. S.
Published: October 2005

Intensity techniques are now widely used for two purposes: sound power determination and source identification. The work described in this presentation illustrates an application of the latter type. As part of a noise control program for a small domestic appliance, a narrow-band intensity measurement system was developed based on a face-to-face microphone probe, a two-channel FFT analyzer, and a personal computer. The probe was used to measure the nearfield normal intensity on planes parallel to the device's major surfaces. Graphical display of these data allowed the radiated sound field to be visualized at any desired frequency. Further, it was possible to distinguish between various types of component sources by examining the field in detail. For example, airborne sound that is generated within the device and radiates to the exterior through apertures or “leaks” was plainly visible. Radiation from the exterior surface of the device resulting from direct vibrational excitation of the surface and from structure-borne vibration could also be identified. Knowledge of this type was used to advantage to guide the noise control treatment program. Nearfield surveys conducted after these modifications yielded a particularly vivid indication of the effects and benefits of the individual modifications.
Uses of sound intensity in architectural acoustics practice
Authors: Orfield Steven J.
Published: October 2005

Sound intensity has received significant attention over the past five years in the fields of noise control and product noise analysis. Yet, at this date, it is an infrequently used technique in consulting acoustics in general. Standards have been slow in developing and thus have not increased its use, and the most fundamental reason for its use is still the utility of its results rather than any existing standard. The benefit of intensity measurement in architecture is principally in its ability to indicate source direction, and this benefit is particularly important in a number of fields now being studied. The first of these is the open plan office, and the current work in intensity is based on an interest in the performance of acoustical dividers and absorbers. The second field, for which intensity seems to be the only measurement technique that is easily used, is the analysis of compound element building facades. This analysis is aimed at developing field transmission loss values for facade elements, such as windows, doors, and wall sections.
On the relationship between sound intensity, pressure, particle velocity, and measurement accuracy
Authors: Pope J.
Published: October 2005

Sound intensity measurements are sensitive to instrumentation inaccuracies in a manner that varies with the sound field in which the meaurements are made. For this reason a number of “quality indicators” have been proposed to allow evaluation of the uncertainty of a measured intensity, the most widely used indicator being the “pressure-intensity index.” Intensity is usually calculated from measurements of sound pressure and either pressure gradient (i.e., two-microphone, p-p devices) or particle velocity (p-u devices). The interpretation and use of the pressure-intensity index is explored for such devices, with emphasis on practical use. It is shown that this indicator is a “phase index” for p-p devices, and an “impedance index” for p-u devices. It is shown also that proper interpretation of the indicator varies with the manner in which the measurement device is realized. For example, p-p devices that sum and difference before filtering require interpretation different from those that filter first. It is concluded that universal limits to ensure a given measurement accuracy cannot be established independent of the measuring device.
Dispersion and attenuation of wind-driven deep gravity waves
Authors: Larraza Andres , Putterman Seth
Published: October 2005

Field and laboratory measurements of the surface of the ocean driven by the wind yield phase velocities that differ from the linear theory dispersion law by as much as 60% [M. Ramamonjiarisoa and C. A. Coantic, Acad. Sci. Paris 288, 111 (1976) and V. Yefimov et al., Atmos. Ocean. Phys. 8, 435 (1972)]. Both short gravity waves and long low-frequency waves are generally observed. The modulations induced by the long wave may have substantial influence on the space-time structure of the high-frequency noise. The wave turbulence motion then becomes inhomogeneous and anisotropic. The short waves in turn modify the surrounding medium by virtue of their radiation stress. The nonlinear interaction between these two components of the sea wave spectrum leads to two propagating modes both of which have characteristics that differ substantially from the linear theory dispersion law. Comparison of the theory with experimental results will be made; further experiments will be suggested.
Observation of a topological kink soliton in a highly compliant waveguide
Authors: Denardo Bruce , Larraza Andrés , Putterman Seth
Published: October 2005

A localized nonpropagating 180° kink has been observed in the phase of the (0,1) cross mode of a parametrically driven trough of liquid. The depth must be less than approximately the width of the trough, so that the resonance hardens. Nonpropagating breather solitons have been observed in a similar geometry but in the deep liquid limit, where the resonance softens [Wu et al., Phys. Rev. Lett. 52, 1421 (1984)]. A kink is insensitive to the ends of the trough unless it is near an end, where it is repelled. If the amplitude is continually decreased, the node is forced to the center, and the wave becomes the linear (1,1) mode. A kink and antikink in close proximity repel each other. In a trough that has a slowly varying width or depth, a kink slowly migrates toward the region of smaller cutoff frequency. Theoretically, modulations of the cross mode are described to lowest order by the nonlinear Schrödinger equation with the opposite sign of the nonlinear term. This possesses a “dark” soliton solution [Hasagawa and Tappert, Appl. Phys. Lett. 23, 171 (1973)] of which the kink is a special case. [Work supported by DOE.]
Recent progress on nonpropagation hydrodynamic solitons
Authors: Wei Ronjue , Wang Benren , Mao Yi , Zheng Xiaoyu , Miao Guoqing
Published: October 2005

The present report will emphasize the recent development of the generation of nonpropagating hydrodynamic solitons of Faraday's oscillating water tank experiment first observed by Wu et al. [Phys. Rev. Lett. 52, 1421 (1984)] in a small rectangular trough (38 × 2.7 × 2 cm3) driven vertically and simple harmonically with a vibrator of fundamental frequency of 10 Hz. It will include: (1) the effect of the geometrical dimension, driving frequency, and amplitude on the stability region, etc. of the solitons, (2) further study on the multisoliton collisions, and (3) the condition and behavior for this solitary system that exhibits bifurcations and chaos, which therefore provide experimental evidence of the relation between these two dominant nonlinear phenomena. Some theoretical models are proposed as well as questions raised by this experimental simplicity. [Work supported by Chinese National Science Foundation.]
Wave amplification in chemically active bubble media
Authors: Zamarayev F. N. , Kedrinskii V. K. , Mader Ch. L.
Published: October 2005

A numerical simulation is performed for shock propagation in chemically active two-phase “liquid-explosive gas bubbles” media. It is established experimentally that this process is accompanied by formation of a stationary disturbance of the wavetrain form. It propagates with constant velocity exceeding essentially the equilibrium velocity characteristic of the main disturbance. Its maximum amplitude is practically constant and exceeds significantly the amplitude of an incident wave. Such a wave is called a bubble detonation wave or secondary detonation. The basis of this effect is the mechanism of separation of the shock wave in a bubble medium into the main disturbance and precursor [V. K. Kedrinskii, PMTF 4, 29–34 (1968)]. The latter decays quickly. However, in chemically active media these losses are compensated for by the bubble explosion energy, thereby providing the existence of a self-sustaining regime. The present study is concerned with two approaches. The first one considers the interaction of a strong plane shock wave with a single spherical bubble filled in with a one-to-one molar acetylene-oxygen mixture. It is shown that the refracted wave can initiate this mixture's detonation [V. K. Kedrinskii and Ch. Mader, Proc. XVI Int. Symp. on Shock Tubes and Waves, July 1987, Aachen, Federal Republic of Germany]. The mixture ignition due to adiabatic bubble compression by a weak shock wave is analyzed within the framework of relatively simple kinetics. The second approach is based on a two-phase model and considers the process of formation of the stationary disturbance in a semi-infinite active bubble medium. The results obtained are in good agreement with the experimental data [A. I. Sychev and A. V. Pinayev, FGV 3, 22 (1986)].
Transfer matrix theory of the propagation of leaky waves in periodic or inhomogeneous media
Authors: Sornette Didier , Macon Louis , Coste Jean
Published: October 2005

A general transfer matrix approach for the propagation of guided waves in the presence of inhomogeneities or “scatterers” is presented that particularly addresses the problem of coupling with radiation modes leading to a leakage of the guided wave in the bulk at each scattering. From symmetry and conservation laws, the general form of the transfer matrix is obtained in terms of four independent real parameters. For one-dimensional periodic lattices of identical scatterers, the leakage vanishes at the band edges: This coherent effect stems from the complete destructive interference between the radiations of the different scatterers at these particular points. The competition between Anderson localization and coherent leakage in the presence of disorder is discussed. Experimental results on the propagation of one-dimensional surface acoustic waves in quasiperiodic and random systems of parallel grooves are presented. Spectral and temporal properties of localization in such systems are well put in evidence in this model experimental system. [Work supported by DRET.]
Attempts to observe acoustic wave turbulence
Authors: Denardo Bruce , Gallet Francois , Putterman Seth , Migliori Albert
Published: October 2005

High-amplitude, low-frequency, broadband acoustic excitation of a medium should lead to a cascade of energy that is analogous to the Kolmogorov turbulence that characterizes vortex eddies. The acoustic cascade should also lead to the phenomenon of classical second sound [Larraza and Putterman, Phys. Rev. Lett. 57, 2810 (1986)]. Attempts to observe the turbulent acoustic power spectrum e(ω) ∝ ω −3/2 and higher-order correlations, in controlled laboratory experiments, are being carried out in the Leo P. Delsasso Acoustics Labs at UCLA. The Allen and Rudnick high-frequency siren [J. Acoust. Soc. Am. 19, 857 (1947)] is being used as a source. In the reverberation room amplitudes of about 160 dB have been achieved. While effects of higher harmonics and scattering of sound by sound can be seen, the intensity level is still apparently about 10 dB below the threshold for acoustic turbulence. [Work supported by DOE.]
Experimental study of streaming flows around a levitated sample in an ultrasonic standing wave
Authors: Trinh E. H.
Published: October 2005

Qualitative and quantitative investigations of the streaming flows caused by high intensity ultrasonic fields have been carried out in order to characterize the environment around levitated solid and liquid samples. The interaction of these second-order convective flows with the sample is of primary interest in the case where the properties of the levitated materials are under investigation. The detailed flow characteristics have been found to depend on the boundaries defining the ultrasonic standing wave and the enclosure around the sample region. Results of flow visualization experiments and velocity measurements will be presented for different frequencies and boundary geometries. [Work supported by NASA.]
Numerical computations of rectified diffusion in nonlinear bubble oscillations
Authors: Fowlkes J. B. , Prosperetti Andrea
Published: October 2005

A previous approach to the problem of rectified diffusion in bubble oscillations [J. Acoust. Soc. Am. 37, 493–503 (1965)] has been applied to the recent nonlinear bubble dynamics theory of Prosperetti et al. [J. Acoust. Soc. Am. 83, 502–514 (1988)]. A numerical solution is given that calculates growth rates and pressure amplitude thresholds for rectified diffusion and allows modification of the radial oscillations of bubbles due to gas diffusion. The results of the new approach are compared to other rectified diffusion theories, and differences in radial bubble oscillations with and without diffusion are examined. Some experimental measurements of rectified diffusion are also compared to the theoretical results. [Work supported by NIH.]
Acoustically driven instability of a liquid drop under microgravity
Authors: Manley Martin , Mehta Vineet
Published: October 2005

When the media surrounding a heated liquid drop is excited by an acoustic wave, acoustic streaming can modify the transfer of heat from the drop's surface. Spatial variations in the heat transfer can result in an adverse surface tension gradient and as well as internal flows in the drop. These flows can also result in unstable modes of vibration. The stability of surface waves on a liquid drop suspended by a plane acoustic wave in a microgravity environment will be discussed. In particular, how acoustic streaming affects the stability and heat transfer at the liquid-gas interface will be addressed.
The effects of vibration on the flow of a non-Newtonian fluid
Authors: Hughes W. Jack , Vavreck Andrew N.
Published: October 2005

The flow characteristics of non-Newtonian fluids are nonlinear and depend upon the generation of a minimum or threshold shear stress before flow is initiated. These fluids are composed of aggregate solids suspended in a fluid medium; examples are concrete and printer's ink. A non-Newtonian fluid was vibrated with ultrasonic and low audio frequencies while its flow characteristics relative to the intensity of vibration were noted. The frequency-amplitude product for threshold shear stress was investigated and has been replaced by a function that is more highly frequency dependent.
Otoneurologic manifestations following carbon monoxide poisoning
Authors: Makishima Kazumi
Published: October 2005

Otoneurologic examinations were conducted in the 15 patients who suffered from acute carbon monoxide (CO) poisoning in a coal mine accident. The audiologic and vestibular derangement found in the patients were not significant, but were suggestive of central origin. One victim in the same accident, who had severe psychopathological symptoms and had no otoneurologic complaint, showed moderate histopathological changes in the cerebrum in the presence of no specific change of the inner ears. The effect of CO poisoning on the threshold sensitivity of the auditory responses was studied in a CO poisoning animal model of guinea pigs. Loss of auditory threshold sensitivity was most prominent at the auditory cortex, and was next most severe at the inferior colliculus. There was no loss of sensitivity at the cochlea. Some of the patients mentioned above have been studied again about 25 years after the accident. It is felt that there must be increased and more considerable widespread pathology in the brain following CO poisoning to explain the present audiometric and vestibural pictures.
Signal processing for sensory aids
Authors: Levitt Harry
Published: October 2005

Sensory aids can be subdivided in two important ways: by modality (auditory, visual, tactile, or direct electrical stimulation) and by degree of signal processing (nonspeech, speech-specific, feature-extraction, and speech-recognition). Nonspeech processing aids are designed to make maximum use of the impaired sensory system regardless of whether communication is by speech or other means. Speech-specific processing is designed to match the average spectral and temporal characteristics of the speech signal to the characteristics of the impaired auditory system. Feature-extraction systems involve the automatic extraction of phonetic or articulatory features from the speech signal. Speech-recognition processing makes use of automatic speech recognition techniques to facilitate the communication process. The application of each of these different forms of signal processing to sensory aids of various kinds will be described. [Work supported by NINCDS.]
Enhancing the clarity of speech for listeners with hearing impairments
Authors: Braida Louis D.
Published: October 2005

Many listeners with hearing impairments exhibit reduced auditory reception of speech both in quiet and in the presence of interfering sounds. A variety of efforts intended to enhance the clarity of speech for such listeners are reviewed. The results obtained will be interpreted in terms of a model [P.M. Zurek and L. A. Delhorne, J. Acoust. Soc. Am. 82, 1548–1559 (1987)] that treats the hearing loss as equivalent to an additive noise with a power spectrum determined by the audiogram. [Work supported by NIH.]
Computer-based speech training aids
Authors: Watson Charles S. , Kewley-Port Diane
Published: October 2005

During the past 20–25 years, a number of research groups, and also individual investigators and clinicians, have attempted to apply computer technology to the problem of teaching speech to deaf or misarticulating children. Most early computer-based training aids used single- or multidimensional visual displays of speech as alternatives to auditory feedback. More recently, computer-based speech recognizers have provided evaluations of the “quality” of utterances, eliminating the requirement that the child interpret various complex transforms of the acoustic waveform. In other systems, feedback is based on the output of sensors that measure the position or motion of the articulators. Positive results obtained with several of these approaches suggest that this line of research will lead to common use of computer-based speech training within 5–15 years. [Research supported by NSF.]
Speech perception with cochlear implants and tactile aids
Authors: Blamey P. J. , Clark G. M. , Dowell R. C.
Published: October 2005

Three related areas in our research are speech perception in noise, consonant recognition, and multimodal speech perception. Perception of speech in four-speaker babble was evaluated with a multiple channel cochlear implant presenting first- and second-formant information. In quiet, scores were 80% for vowels, 48% for consonants, and 51% for sentences. The signal-to-noise ratios required to reduce scores to 75% of the value in quiet were 8 dB for vowels, 12 dB for consonants, and 14 dB for sentences. Detailed consonant studies showed that amplitude information contributed directly to the recognition of voicing and manner of articulation and influenced the perception of other cues. In addition to auditory cues, lipreading is commonly used with implants and other aids. A probabilistic model of feature recognition describes the combination of auditory plus visual information well. The combined effect of tactile information is overestimated by the model for minimally trained subjects. The model may clarify issues in the training of hearing-impaired children, as well as the design of speech processing aids.
Within-subject comparisons of analog and pulsatile speech processors for cochlear implants
Authors: Soli Sigfrid D. , Wilson Blake S.
Published: October 2005

Analog and pulsatile speech processors for cochlear implants represent widely different strategies for coding perceptual information in speech. Between-subject comparisons of their efficacy, however, have been confounded by large and unquantified subject differences. Two patients implanted with multiple electrode arrays have been tested both with four-channel analog and pulsatile coding strategies. The analog processor performed bandpass filtering with syllabic compression/limiting and stimulated four bipolar electrode pairs. The pulsatile processor delivered volleys of biphasic pulses at the rate of F0 during voiced intervals of speech to four of six possible bipolar electrode pairs selected on the basis of spectral peak frequencies. Tests of auditory capabilities showed that the two coding strategies provided similar overall benefit. Analyses of consonant and vowel confusions revealed that the two strategies were not equally effective for some types of speech information. The results of these analyses can be used to identify the strengths and weaknesses of analog and pulsatile speech coding strategies for cochlear implants.
Speech processing to aid the profoundly deaf
Authors: Vilaclara Georges
Published: October 2005

This work is part of an extended project to transmit speech information to the profoundly deaf through an alternate sensory modality. A tool was developed to select and define codes to complement speechreading. The learning procedure operates on a set of speech waveforms (known database), preprocessed to produce a large collection of parameters at regular time intervals, which is phonetically segmented according to a stability function. The experimentalist identifies segments in terms of units (e.g., linguistic, acoustic, or articulatory units). For each unit, the variability of its parameters is automatically represented by statistical patterns. In addition, probabilistic rules describing the unit's characteristics and properties can then be defined. All these specifications are stored in a knowledge base. Two techniques are used to recognize the units in unknown speech. A pattern matcher determines the most suitable statistical pattern to represent the current frame. An expert system supervises the pattern marcher, analyses results, and/or infers decisions. Presented in a suitable way (user defined) via a prosthesis, the code can be tested to determine its utility to cross-modal speech perception. [Work supported by the Swiss NSF.]